From 597603d615d2b19a9e451d8cfac24372856a522d Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 9 Aug 2010 14:21:11 +0200 Subject: ALSA: introduce the snd-aloop module for the PCM loopback The snd-aloop module allows redirecting of the PCM playback in the kernel back to the user space using the standard ALSA PCM capture API. The module also allows time synchronization with another timing source and notifications of playback stream parameter changes. Signed-off-by: Jaroslav Kysela --- sound/drivers/Kconfig | 19 + sound/drivers/Makefile | 2 + sound/drivers/aloop.c | 1055 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1076 insertions(+) create mode 100644 sound/drivers/aloop.c (limited to 'sound') diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 84714a65e5c8..b6ae76285255 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -74,6 +74,25 @@ config SND_DUMMY To compile this driver as a module, choose M here: the module will be called snd-dummy. +config SND_ALOOP + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. + This module returns played samples back to the user space using + the standard ALSA PCM device. The devices are routed 0->1 and + 1->0, where first number is the playback PCM device and second + number is the capture device. Module creates two PCM devices and + configured number of substreams (see the pcm_substreams module + parameter). + + The looback device allow time sychronization with an external + timing source using the time shift universal control (+-20% + of system time). + + To compile this driver as a module, choose M here: the module + will be called snd-aloop. + config SND_VIRMIDI tristate "Virtual MIDI soundcard" depends on SND_SEQUENCER diff --git a/sound/drivers/Makefile b/sound/drivers/Makefile index d4a07f9ff2c7..1a8440c8b138 100644 --- a/sound/drivers/Makefile +++ b/sound/drivers/Makefile @@ -4,6 +4,7 @@ # snd-dummy-objs := dummy.o +snd-aloop-objs := aloop.o snd-mtpav-objs := mtpav.o snd-mts64-objs := mts64.o snd-portman2x4-objs := portman2x4.o @@ -13,6 +14,7 @@ snd-ml403-ac97cr-objs := ml403-ac97cr.o pcm-indirect2.o # Toplevel Module Dependency obj-$(CONFIG_SND_DUMMY) += snd-dummy.o +obj-$(CONFIG_SND_ALOOP) += snd-aloop.o obj-$(CONFIG_SND_VIRMIDI) += snd-virmidi.o obj-$(CONFIG_SND_SERIAL_U16550) += snd-serial-u16550.o obj-$(CONFIG_SND_MTPAV) += snd-mtpav.o diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c new file mode 100644 index 000000000000..3123a15d23f6 --- /dev/null +++ b/sound/drivers/aloop.c @@ -0,0 +1,1055 @@ +/* + * Loopback soundcard + * + * Original code: + * Copyright (c) by Jaroslav Kysela + * + * More accurate positioning and full-duplex support: + * Copyright (c) Ahmet İnan + * + * Major (almost complete) rewrite: + * Copyright (c) by Takashi Iwai + * + * A next major update in 2010 (separate timers for playback and capture): + * Copyright (c) Jaroslav Kysela + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("Jaroslav Kysela "); +MODULE_DESCRIPTION("A loopback soundcard"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{ALSA,Loopback soundcard}}"); + +#define MAX_PCM_SUBSTREAMS 8 + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; +static int pcm_notify[SNDRV_CARDS]; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for loopback soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for loopback soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable this loopback soundcard."); +module_param_array(pcm_substreams, int, NULL, 0444); +MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-8) for loopback driver."); +module_param_array(pcm_notify, int, NULL, 0444); +MODULE_PARM_DESC(pcm_notify, "Break capture when PCM format/rate/channels changes."); + +#define NO_PITCH 100000 + +struct loopback_pcm; + +struct loopback_cable { + spinlock_t lock; + struct loopback_pcm *streams[2]; + struct snd_pcm_hardware hw; + /* flags */ + unsigned int valid; + unsigned int running; +}; + +struct loopback_setup { + unsigned int notify: 1; + unsigned int rate_shift; + unsigned int format; + unsigned int rate; + unsigned int channels; + struct snd_ctl_elem_id active_id; + struct snd_ctl_elem_id format_id; + struct snd_ctl_elem_id rate_id; + struct snd_ctl_elem_id channels_id; +}; + +struct loopback { + struct snd_card *card; + struct mutex cable_lock; + struct loopback_cable *cables[MAX_PCM_SUBSTREAMS][2]; + struct snd_pcm *pcm[2]; + struct loopback_setup setup[MAX_PCM_SUBSTREAMS][2]; +}; + +struct loopback_pcm { + struct loopback *loopback; + struct snd_pcm_substream *substream; + struct loopback_cable *cable; + unsigned int pcm_buffer_size; + unsigned int buf_pos; /* position in buffer */ + unsigned int silent_size; + /* PCM parameters */ + unsigned int pcm_period_size; + unsigned int pcm_bps; /* bytes per second */ + unsigned int pcm_salign; /* bytes per sample * channels */ + unsigned int pcm_rate_shift; /* rate shift value */ + /* flags */ + unsigned int period_update_pending :1; + /* timer stuff */ + unsigned int irq_pos; /* fractional IRQ position */ + unsigned int period_size_frac; + unsigned long last_jiffies; + struct timer_list timer; +}; + +static struct platform_device *devices[SNDRV_CARDS]; + +static inline unsigned int byte_pos(struct loopback_pcm *dpcm, unsigned int x) +{ + if (dpcm->pcm_rate_shift == NO_PITCH) { + x /= HZ; + } else { + x = div_u64(NO_PITCH * (unsigned long long)x, + HZ * (unsigned long long)dpcm->pcm_rate_shift); + } + return x - (x % dpcm->pcm_salign); +} + +static inline unsigned int frac_pos(struct loopback_pcm *dpcm, unsigned int x) +{ + if (dpcm->pcm_rate_shift == NO_PITCH) { /* no pitch */ + return x * HZ; + } else { + x = div_u64(dpcm->pcm_rate_shift * (unsigned long long)x * HZ, + NO_PITCH); + } + return x; +} + +static inline struct loopback_setup *get_setup(struct loopback_pcm *dpcm) +{ + int device = dpcm->substream->pstr->pcm->device; + + if (dpcm->substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + device ^= 1; + return &dpcm->loopback->setup[dpcm->substream->number][device]; +} + +static inline unsigned int get_notify(struct loopback_pcm *dpcm) +{ + return get_setup(dpcm)->notify; +} + +static inline unsigned int get_rate_shift(struct loopback_pcm *dpcm) +{ + return get_setup(dpcm)->rate_shift; +} + +static void loopback_timer_start(struct loopback_pcm *dpcm) +{ + unsigned long tick; + unsigned int rate_shift = get_rate_shift(dpcm); + + if (rate_shift != dpcm->pcm_rate_shift) { + dpcm->pcm_rate_shift = rate_shift; + dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); + } + tick = dpcm->period_size_frac - dpcm->irq_pos; + tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; + dpcm->timer.expires = jiffies + tick; + add_timer(&dpcm->timer); +} + +static inline void loopback_timer_stop(struct loopback_pcm *dpcm) +{ + del_timer(&dpcm->timer); +} + +#define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) +#define CABLE_VALID_CAPTURE (1 << SNDRV_PCM_STREAM_CAPTURE) +#define CABLE_VALID_BOTH (CABLE_VALID_PLAYBACK|CABLE_VALID_CAPTURE) + +static int loopback_check_format(struct loopback_cable *cable, int stream) +{ + struct snd_pcm_runtime *runtime; + struct loopback_setup *setup; + struct snd_card *card; + int check; + + if (cable->valid != CABLE_VALID_BOTH) { + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + goto __notify; + return 0; + } + runtime = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]-> + substream->runtime; + check = cable->hw.formats != (1ULL << runtime->format) || + cable->hw.rate_min != runtime->rate || + cable->hw.rate_max != runtime->rate || + cable->hw.channels_min != runtime->channels || + cable->hw.channels_max != runtime->channels; + if (!check) + return 0; + if (stream == SNDRV_PCM_STREAM_CAPTURE) { + return -EIO; + } else { + snd_pcm_stop(cable->streams[SNDRV_PCM_STREAM_CAPTURE]-> + substream, SNDRV_PCM_STATE_DRAINING); + __notify: + runtime = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]-> + substream->runtime; + setup = get_setup(cable->streams[SNDRV_PCM_STREAM_PLAYBACK]); + card = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]->loopback->card; + if (setup->format != runtime->format) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->format_id); + setup->format = runtime->format; + } + if (setup->rate != runtime->rate) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->rate_id); + setup->rate = runtime->rate; + } + if (setup->channels != runtime->channels) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &setup->channels_id); + setup->channels = runtime->channels; + } + } + return 0; +} + +static void loopback_active_notify(struct loopback_pcm *dpcm) +{ + snd_ctl_notify(dpcm->loopback->card, + SNDRV_CTL_EVENT_MASK_VALUE, + &get_setup(dpcm)->active_id); +} + +static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + int err; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + err = loopback_check_format(cable, substream->stream); + if (err < 0) + return err; + dpcm->last_jiffies = jiffies; + dpcm->pcm_rate_shift = 0; + loopback_timer_start(dpcm); + cable->running |= (1 << substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); + break; + case SNDRV_PCM_TRIGGER_STOP: + cable->running &= ~(1 << substream->stream); + loopback_timer_stop(dpcm); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); + break; + default: + return -EINVAL; + } + return 0; +} + +static int loopback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + unsigned int bps, salign; + + salign = (snd_pcm_format_width(runtime->format) * + runtime->channels) / 8; + bps = salign * runtime->rate; + if (bps <= 0 || salign <= 0) + return -EINVAL; + + dpcm->buf_pos = 0; + dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + /* clear capture buffer */ + dpcm->silent_size = dpcm->pcm_buffer_size; + snd_pcm_format_set_silence(runtime->format, runtime->dma_area, + runtime->buffer_size * runtime->channels); + } + + dpcm->irq_pos = 0; + dpcm->period_update_pending = 0; + dpcm->pcm_bps = bps; + dpcm->pcm_salign = salign; + dpcm->pcm_period_size = frames_to_bytes(runtime, runtime->period_size); + + mutex_lock(&dpcm->loopback->cable_lock); + if (!(cable->valid & ~(1 << substream->stream))) { + cable->hw.formats = (1ULL << runtime->format); + cable->hw.rate_min = runtime->rate; + cable->hw.rate_max = runtime->rate; + cable->hw.channels_min = runtime->channels; + cable->hw.channels_max = runtime->channels; + } + cable->valid |= 1 << substream->stream; + mutex_unlock(&dpcm->loopback->cable_lock); + + return 0; +} + +static void clear_capture_buf(struct loopback_pcm *dpcm, unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = dpcm->substream->runtime; + char *dst = runtime->dma_area; + unsigned int dst_off = dpcm->buf_pos; + + if (dpcm->silent_size >= dpcm->pcm_buffer_size) + return; + if (dpcm->silent_size + bytes > dpcm->pcm_buffer_size) + bytes = dpcm->pcm_buffer_size - dpcm->silent_size; + + for (;;) { + unsigned int size = bytes; + if (dst_off + size > dpcm->pcm_buffer_size) + size = dpcm->pcm_buffer_size - dst_off; + snd_pcm_format_set_silence(runtime->format, dst + dst_off, + bytes_to_frames(runtime, size) * + runtime->channels); + dpcm->silent_size += size; + bytes -= size; + if (!bytes) + break; + dst_off = 0; + } +} + +static void copy_play_buf(struct loopback_pcm *play, + struct loopback_pcm *capt, + unsigned int bytes) +{ + struct snd_pcm_runtime *runtime = play->substream->runtime; + char *src = play->substream->runtime->dma_area; + char *dst = capt->substream->runtime->dma_area; + unsigned int src_off = play->buf_pos; + unsigned int dst_off = capt->buf_pos; + unsigned int clear_bytes = 0; + + /* check if playback is draining, trim the capture copy size + * when our pointer is at the end of playback ring buffer */ + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + snd_pcm_playback_hw_avail(runtime) < runtime->buffer_size) { + snd_pcm_uframes_t appl_ptr, appl_ptr1, diff; + appl_ptr = appl_ptr1 = runtime->control->appl_ptr; + appl_ptr1 -= appl_ptr1 % runtime->buffer_size; + appl_ptr1 += play->buf_pos / play->pcm_salign; + if (appl_ptr < appl_ptr1) + appl_ptr1 -= runtime->buffer_size; + diff = (appl_ptr - appl_ptr1) * play->pcm_salign; + if (diff < bytes) { + clear_bytes = bytes - diff; + bytes = diff; + } + } + + for (;;) { + unsigned int size = bytes; + if (src_off + size > play->pcm_buffer_size) + size = play->pcm_buffer_size - src_off; + if (dst_off + size > capt->pcm_buffer_size) + size = capt->pcm_buffer_size - dst_off; + memcpy(dst + dst_off, src + src_off, size); + capt->silent_size = 0; + bytes -= size; + if (!bytes) + break; + src_off = (src_off + size) % play->pcm_buffer_size; + dst_off = (dst_off + size) % capt->pcm_buffer_size; + } + + if (clear_bytes > 0) + clear_capture_buf(capt, clear_bytes); +} + +#define BYTEPOS_UPDATE_POSONLY 0 +#define BYTEPOS_UPDATE_CLEAR 1 +#define BYTEPOS_UPDATE_COPY 2 + +static void loopback_bytepos_update(struct loopback_pcm *dpcm, + unsigned int delta, + unsigned int cmd) +{ + unsigned int count; + unsigned long last_pos; + + last_pos = byte_pos(dpcm, dpcm->irq_pos); + dpcm->irq_pos += delta * dpcm->pcm_bps; + count = byte_pos(dpcm, dpcm->irq_pos) - last_pos; + if (!count) + return; + if (cmd == BYTEPOS_UPDATE_CLEAR) + clear_capture_buf(dpcm, count); + else if (cmd == BYTEPOS_UPDATE_COPY) + copy_play_buf(dpcm->cable->streams[SNDRV_PCM_STREAM_PLAYBACK], + dpcm->cable->streams[SNDRV_PCM_STREAM_CAPTURE], + count); + dpcm->buf_pos += count; + dpcm->buf_pos %= dpcm->pcm_buffer_size; + if (dpcm->irq_pos >= dpcm->period_size_frac) { + dpcm->irq_pos %= dpcm->period_size_frac; + dpcm->period_update_pending = 1; + } +} + +static void loopback_pos_update(struct loopback_cable *cable) +{ + struct loopback_pcm *dpcm_play = + cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; + struct loopback_pcm *dpcm_capt = + cable->streams[SNDRV_PCM_STREAM_CAPTURE]; + unsigned long delta_play = 0, delta_capt = 0; + + spin_lock(&cable->lock); + if (cable->running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { + delta_play = jiffies - dpcm_play->last_jiffies; + dpcm_play->last_jiffies += delta_play; + } + + if (cable->running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { + delta_capt = jiffies - dpcm_capt->last_jiffies; + dpcm_capt->last_jiffies += delta_capt; + } + + if (delta_play == 0 && delta_capt == 0) { + spin_unlock(&cable->lock); + return; + } + + if (delta_play > delta_capt) { + loopback_bytepos_update(dpcm_play, delta_play - delta_capt, + BYTEPOS_UPDATE_POSONLY); + delta_play = delta_capt; + } else if (delta_play < delta_capt) { + loopback_bytepos_update(dpcm_capt, delta_capt - delta_play, + BYTEPOS_UPDATE_CLEAR); + delta_capt = delta_play; + } + + if (delta_play == 0 && delta_capt == 0) { + spin_unlock(&cable->lock); + return; + } + /* note delta_capt == delta_play at this moment */ + loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY); + loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY); + spin_unlock(&cable->lock); +} + +static void loopback_timer_function(unsigned long data) +{ + struct loopback_pcm *dpcm = (struct loopback_pcm *)data; + int stream; + + loopback_pos_update(dpcm->cable); + stream = dpcm->substream->stream; + if (dpcm->cable->running & (1 << stream)) + loopback_timer_start(dpcm); + if (dpcm->period_update_pending) { + dpcm->period_update_pending = 0; + if (dpcm->cable->running & (1 << stream)) + snd_pcm_period_elapsed(dpcm->substream); + } +} + +static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + + loopback_pos_update(dpcm->cable); + return bytes_to_frames(runtime, dpcm->buf_pos); +} + +static struct snd_pcm_hardware loopback_pcm_hardware = +{ + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | + SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), + .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 32, + .buffer_bytes_max = 2 * 1024 * 1024, + .period_bytes_min = 64, + .period_bytes_max = 2 * 1024 * 1024, + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +static void loopback_runtime_free(struct snd_pcm_runtime *runtime) +{ + struct loopback_pcm *dpcm = runtime->private_data; + kfree(dpcm); +} + +static int loopback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); +} + +static int loopback_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + + mutex_lock(&dpcm->loopback->cable_lock); + cable->valid &= ~(1 << substream->stream); + mutex_unlock(&dpcm->loopback->cable_lock); + return snd_pcm_lib_free_pages(substream); +} + +static unsigned int get_cable_index(struct snd_pcm_substream *substream) +{ + if (!substream->pcm->device) + return substream->stream; + else + return !substream->stream; +} + +static int loopback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback *loopback = substream->private_data; + struct loopback_pcm *dpcm; + struct loopback_cable *cable; + int err = 0; + int dev = get_cable_index(substream); + + mutex_lock(&loopback->cable_lock); + dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); + if (!dpcm) { + err = -ENOMEM; + goto unlock; + } + dpcm->loopback = loopback; + dpcm->substream = substream; + setup_timer(&dpcm->timer, loopback_timer_function, + (unsigned long)dpcm); + + cable = loopback->cables[substream->number][dev]; + if (!cable) { + cable = kzalloc(sizeof(*cable), GFP_KERNEL); + if (!cable) { + kfree(dpcm); + err = -ENOMEM; + goto unlock; + } + spin_lock_init(&cable->lock); + cable->hw = loopback_pcm_hardware; + loopback->cables[substream->number][dev] = cable; + } + dpcm->cable = cable; + cable->streams[substream->stream] = dpcm; + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + + runtime->private_data = dpcm; + runtime->private_free = loopback_runtime_free; + if (get_notify(dpcm) && + substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + runtime->hw = loopback_pcm_hardware; + } else { + runtime->hw = cable->hw; + } + unlock: + mutex_unlock(&loopback->cable_lock); + return err; +} + +static int loopback_close(struct snd_pcm_substream *substream) +{ + struct loopback *loopback = substream->private_data; + struct loopback_pcm *dpcm = substream->runtime->private_data; + struct loopback_cable *cable; + int dev = get_cable_index(substream); + + loopback_timer_stop(dpcm); + mutex_lock(&loopback->cable_lock); + cable = loopback->cables[substream->number][dev]; + if (cable->streams[!substream->stream]) { + /* other stream is still alive */ + cable->streams[substream->stream] = NULL; + } else { + /* free the cable */ + loopback->cables[substream->number][dev] = NULL; + kfree(cable); + } + mutex_unlock(&loopback->cable_lock); + return 0; +} + +static struct snd_pcm_ops loopback_playback_ops = { + .open = loopback_open, + .close = loopback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = loopback_hw_params, + .hw_free = loopback_hw_free, + .prepare = loopback_prepare, + .trigger = loopback_trigger, + .pointer = loopback_pointer, +}; + +static struct snd_pcm_ops loopback_capture_ops = { + .open = loopback_open, + .close = loopback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = loopback_hw_params, + .hw_free = loopback_hw_free, + .prepare = loopback_prepare, + .trigger = loopback_trigger, + .pointer = loopback_pointer, +}; + +static int __devinit loopback_pcm_new(struct loopback *loopback, + int device, int substreams) +{ + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(loopback->card, "Loopback PCM", device, + substreams, substreams, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_capture_ops); + + pcm->private_data = loopback; + pcm->info_flags = 0; + strcpy(pcm->name, "Loopback PCM"); + + loopback->pcm[device] = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 2 * 1024 * 1024); + return 0; +} + +static int loopback_rate_shift_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 80000; + uinfo->value.integer.max = 120000; + uinfo->value.integer.step = 1; + return 0; +} + +static int loopback_rate_shift_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].rate_shift; + return 0; +} + +static int loopback_rate_shift_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + + val = ucontrol->value.integer.value[0]; + if (val < 80000) + val = 80000; + if (val > 120000) + val = 120000; + mutex_lock(&loopback->cable_lock); + if (val != loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].rate_shift) { + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].rate_shift = val; + change = 1; + } + mutex_unlock(&loopback->cable_lock); + return change; +} + +static int loopback_notify_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].notify; + return 0; +} + +static int loopback_notify_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + + val = ucontrol->value.integer.value[0] ? 1 : 0; + if (val != loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].notify) { + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].notify = val; + change = 1; + } + return change; +} + +static int loopback_active_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + struct loopback_cable *cable = loopback->cables + [kcontrol->id.subdevice][kcontrol->id.device]; + unsigned int val = 0; + + if (cable != NULL) + val = (cable->running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? + 1 : 0; + ucontrol->value.integer.value[0] = val; + return 0; +} + +static int loopback_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SNDRV_PCM_FORMAT_LAST; + uinfo->value.integer.step = 1; + return 0; +} + +static int loopback_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].format; + return 0; +} + +static int loopback_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 192000; + uinfo->value.integer.step = 1; + return 0; +} + +static int loopback_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].rate; + return 0; +} + +static int loopback_channels_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 1; + uinfo->value.integer.max = 1024; + uinfo->value.integer.step = 1; + return 0; +} + +static int loopback_channels_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct loopback *loopback = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + loopback->setup[kcontrol->id.subdevice] + [kcontrol->id.device].rate; + return 0; +} + +static struct snd_kcontrol_new loopback_controls[] __devinitdata = { +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Rate Shift 100000", + .info = loopback_rate_shift_info, + .get = loopback_rate_shift_get, + .put = loopback_rate_shift_put, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Notify", + .info = snd_ctl_boolean_mono_info, + .get = loopback_notify_get, + .put = loopback_notify_put, +}, +#define ACTIVE_IDX 2 +{ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Slave Active", + .info = snd_ctl_boolean_mono_info, + .get = loopback_active_get, +}, +#define FORMAT_IDX 3 +{ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Slave Format", + .info = loopback_format_info, + .get = loopback_format_get +}, +#define RATE_IDX 4 +{ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Slave Rate", + .info = loopback_rate_info, + .get = loopback_rate_get +}, +#define CHANNELS_IDX 5 +{ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Slave Channels", + .info = loopback_channels_info, + .get = loopback_channels_get +} +}; + +static int __devinit loopback_mixer_new(struct loopback *loopback, int notify) +{ + struct snd_card *card = loopback->card; + struct snd_pcm *pcm; + struct snd_kcontrol *kctl; + struct loopback_setup *setup; + int err, dev, substr, substr_count, idx; + + strcpy(card->mixername, "Loopback Mixer"); + for (dev = 0; dev < 2; dev++) { + pcm = loopback->pcm[dev]; + substr_count = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream_count; + for (substr = 0; substr < substr_count; substr++) { + setup = &loopback->setup[substr][dev]; + setup->notify = notify; + setup->rate_shift = NO_PITCH; + setup->format = SNDRV_PCM_FORMAT_S16_LE; + setup->rate = 48000; + setup->channels = 2; + for (idx = 0; idx < ARRAY_SIZE(loopback_controls); + idx++) { + kctl = snd_ctl_new1(&loopback_controls[idx], + loopback); + if (!kctl) + return -ENOMEM; + kctl->id.device = dev; + kctl->id.subdevice = substr; + switch (idx) { + case ACTIVE_IDX: + setup->active_id = kctl->id; + break; + case FORMAT_IDX: + setup->format_id = kctl->id; + break; + case RATE_IDX: + setup->rate_id = kctl->id; + break; + case CHANNELS_IDX: + setup->channels_id = kctl->id; + break; + default: + break; + } + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + } + } + return 0; +} + +static int __devinit loopback_probe(struct platform_device *devptr) +{ + struct snd_card *card; + struct loopback *loopback; + int dev = devptr->id; + int err; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct loopback), &card); + if (err < 0) + return err; + loopback = card->private_data; + + if (pcm_substreams[dev] < 1) + pcm_substreams[dev] = 1; + if (pcm_substreams[dev] > MAX_PCM_SUBSTREAMS) + pcm_substreams[dev] = MAX_PCM_SUBSTREAMS; + + loopback->card = card; + mutex_init(&loopback->cable_lock); + + err = loopback_pcm_new(loopback, 0, pcm_substreams[dev]); + if (err < 0) + goto __nodev; + err = loopback_pcm_new(loopback, 1, pcm_substreams[dev]); + if (err < 0) + goto __nodev; + err = loopback_mixer_new(loopback, pcm_notify[dev] ? 1 : 0); + if (err < 0) + goto __nodev; + strcpy(card->driver, "Loopback"); + strcpy(card->shortname, "Loopback"); + sprintf(card->longname, "Loopback %i", dev + 1); + err = snd_card_register(card); + if (!err) { + platform_set_drvdata(devptr, card); + return 0; + } + __nodev: + snd_card_free(card); + return err; +} + +static int __devexit loopback_remove(struct platform_device *devptr) +{ + snd_card_free(platform_get_drvdata(devptr)); + platform_set_drvdata(devptr, NULL); + return 0; +} + +#ifdef CONFIG_PM +static int loopback_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct loopback *loopback = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(loopback->pcm[0]); + snd_pcm_suspend_all(loopback->pcm[1]); + return 0; +} + +static int loopback_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + +#define SND_LOOPBACK_DRIVER "snd_aloop" + +static struct platform_driver loopback_driver = { + .probe = loopback_probe, + .remove = __devexit_p(loopback_remove), +#ifdef CONFIG_PM + .suspend = loopback_suspend, + .resume = loopback_resume, +#endif + .driver = { + .name = SND_LOOPBACK_DRIVER + }, +}; + +static void loopback_unregister_all(void) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(devices); ++i) + platform_device_unregister(devices[i]); + platform_driver_unregister(&loopback_driver); +} + +static int __init alsa_card_loopback_init(void) +{ + int i, err, cards; + + err = platform_driver_register(&loopback_driver); + if (err < 0) + return err; + + + cards = 0; + for (i = 0; i < SNDRV_CARDS; i++) { + struct platform_device *device; + if (!enable[i]) + continue; + device = platform_device_register_simple(SND_LOOPBACK_DRIVER, + i, NULL, 0); + if (IS_ERR(device)) + continue; + if (!platform_get_drvdata(device)) { + platform_device_unregister(device); + continue; + } + devices[i] = device; + cards++; + } + if (!cards) { +#ifdef MODULE + printk(KERN_ERR "aloop: No loopback enabled\n"); +#endif + loopback_unregister_all(); + return -ENODEV; + } + return 0; +} + +static void __exit alsa_card_loopback_exit(void) +{ + loopback_unregister_all(); +} + +module_init(alsa_card_loopback_init) +module_exit(alsa_card_loopback_exit) -- cgit v1.2.3 From 495311927ffbe3604e915aeafdf03325e9925b9d Mon Sep 17 00:00:00 2001 From: René Herman Date: Wed, 11 Aug 2010 13:08:06 +0200 Subject: ALSA: ISA: New Aztech Sound Galaxy driver This is a new driver for Aztech Sound Galaxy ISA soundcards based on the AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers for either chipset generated from the same source file, with (very) minimal ifdeffery. The drivers check the SB DSP version to decide if they are being loaded for the right chip. AZT1605 returns 2.1 by default and AZT2316 3.1. This isn't full-proof as the DSP version can actually be set through software but it's close enough -- as far as I've been able to see, the DSP version can not be stored in the EEPROM and the cards will therefore startup with the defaults. This distinction could (with the same success rate) also be used to decide which chip we're looking at at runtime meaning a single, merged driver is also an option but I feel it's actually nicer this way. A merged driver would have to postpone translating the passed in resource values to the card configuration until it knew which one it was looking at and would need to postpone erring out on mpu_irq=10 for azt1605 and mpu_irq=3 for azt2316. The drivers have been tested on various cards. For snd-azt1605: FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II and for snd-azt2316: FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201) FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202) FCC-ID I38-MMSN837: Packard Bell ISA Soundcard 030069 FCC-ID I38-MMSN846: Trust Sound Expert DeLuxe 16-3D (06300) FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301) FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D 826 and 846 were also marketed directly by Aztech and then known as: FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+ FCC-ID I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D Together, these cover the AZT1605 and AT2316A, AZT2316R and AZT2316-S chipsets. All cards work fully -- full-duplex PCM, MIDI and FM. Full duplex is a little flaky on some. I38-MSN811 tends to not work in full-duplex but sometimes does with the highest success rate being achieved when you first start the capture and then a playback instead of the other way around (it's a CS4231-KL codec). The cards with an AD1845XP codec (my I38-MMSN826 and one of my I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex works, sometimes not and this varies from try to try. This seems likely to be a timing problem somewhere inside wss-lib. I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth onboard that isn't supported yet. The wavetable synths on I38-MMSN847 and I38-MMSN852 are wired directly to the standard MPU-401 UART and the AUX1 input on the codec and work without problem. CD-ROM audio on the cards is routed to the codec "Line" input, Line-In to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename the controls due to the capture source enumeration: I see that capture-source overrides are hardcoded in wss-lib and this is just too ugly to live. Versus the old snd-sgalaxy driver these drivers add support for the models without a configuration EEPROM (which are common), full-duplex, MPU-401 UART and OPL3. In the future they might grow support for that ICS2115 WaveFront synth on 826 and an hwdep interface to write to the EEPROM on the models that have one. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 68 +++ sound/isa/Kconfig | 26 + sound/isa/Makefile | 2 +- sound/isa/galaxy/Makefile | 10 + sound/isa/galaxy/azt1605.c | 91 ++++ sound/isa/galaxy/azt2316.c | 111 ++++ sound/isa/galaxy/galaxy.c | 652 ++++++++++++++++++++++++ 7 files changed, 959 insertions(+), 1 deletion(-) create mode 100644 sound/isa/galaxy/Makefile create mode 100644 sound/isa/galaxy/azt1605.c create mode 100644 sound/isa/galaxy/azt2316.c create mode 100644 sound/isa/galaxy/galaxy.c (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7f4dcebda9c6..e25555c02166 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -300,6 +300,74 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + Module snd-azt1605 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-azt2316 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + Module snd-aw2 -------------- diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index c6990c680796..f9aa13d8dacc 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -77,6 +77,32 @@ config SND_ALS100 To compile this driver as a module, choose M here: the module will be called snd-als100. +config SND_AZT1605 + tristate "Aztech AZT1605 Driver" + depends on SND + select SND_WSS_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT1605 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt1605. + +config SND_AZT2316 + tristate "Aztech AZT2316 Driver" + depends on SND + select SND_WSS_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT2316 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt2316. + config SND_AZT2320 tristate "Aztech Systems AZT2320" depends on PNP diff --git a/sound/isa/Makefile b/sound/isa/Makefile index c73d30c4f462..d2bd8f536dd8 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -24,5 +24,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ opti9xx/ \ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/galaxy/Makefile b/sound/isa/galaxy/Makefile new file mode 100644 index 000000000000..e307066d4315 --- /dev/null +++ b/sound/isa/galaxy/Makefile @@ -0,0 +1,10 @@ +# +# Makefile for ALSA +# Copyright (c) 2001 by Jaroslav Kysela +# + +snd-azt1605-objs := azt1605.o +snd-azt2316-objs := azt2316.o + +obj-$(CONFIG_SND_AZT1605) += snd-azt1605.o +obj-$(CONFIG_SND_AZT2316) += snd-azt2316.o diff --git a/sound/isa/galaxy/azt1605.c b/sound/isa/galaxy/azt1605.c new file mode 100644 index 000000000000..9a97643cb713 --- /dev/null +++ b/sound/isa/galaxy/azt1605.c @@ -0,0 +1,91 @@ +/* + * Aztech AZT1605 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + */ + +#define AZT1605 + +#define CRD_NAME "Aztech AZT1605" +#define DRV_NAME "AZT1605" +#define DEV_NAME "azt1605" + +#define GALAXY_DSP_MAJOR 2 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 3 + +/* + * 24-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_MPUA_300 (0 << 2) +#define GALAXY_CONFIG_MPUA_330 (1 << 2) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 3) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 4) + +#define GALAXY_CONFIG_CD_PANASONIC (1 << 5) +#define GALAXY_CONFIG_CD_MITSUMI (1 << 6) +#define GALAXY_CONFIG_CD_MASK (\ + GALAXY_CONFIG_CD_PANASONIC | GALAXY_CONFIG_CD_MITSUMI) + +#define GALAXY_CONFIG_UNUSED (1 << 7) +#define GALAXY_CONFIG_UNUSED_MASK GALAXY_CONFIG_UNUSED + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 8) +#define GALAXY_CONFIG_SBIRQ_3 (1 << 9) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 10) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 11) + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 12) +#define GALAXY_CONFIG_MPUIRQ_3 (1 << 13) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 14) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 15) + +#define GALAXY_CONFIG_WSSA_530 (0 << 16) +#define GALAXY_CONFIG_WSSA_604 (1 << 16) +#define GALAXY_CONFIG_WSSA_E80 (2 << 16) +#define GALAXY_CONFIG_WSSA_F40 (3 << 16) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 18) + +#define GALAXY_CONFIG_CDIRQ_11 (1 << 19) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 20) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 21) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_11 | GALAXY_CONFIG_CDIRQ_12 |\ + GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_CDDMA_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA_0 (1 << 22) +#define GALAXY_CONFIG_CDDMA_1 (2 << 22) +#define GALAXY_CONFIG_CDDMA_3 (3 << 22) +#define GALAXY_CONFIG_CDDMA_MASK GALAXY_CONFIG_CDDMA_3 + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CD_MASK |\ + GALAXY_CONFIG_UNUSED_MASK | GALAXY_CONFIG_CDIRQ_MASK |\ + GALAXY_CONFIG_CDDMA_MASK) + +#include "galaxy.c" diff --git a/sound/isa/galaxy/azt2316.c b/sound/isa/galaxy/azt2316.c new file mode 100644 index 000000000000..189441141df6 --- /dev/null +++ b/sound/isa/galaxy/azt2316.c @@ -0,0 +1,111 @@ +/* + * Aztech AZT2316 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + */ + +#define AZT2316 + +#define CRD_NAME "Aztech AZT2316" +#define DRV_NAME "AZT2316" +#define DEV_NAME "azt2316" + +#define GALAXY_DSP_MAJOR 3 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 4 + +/* + * 32-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 2) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 3) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 4) +#define GALAXY_CONFIG_SBIRQ_10 (1 << 5) + +#define GALAXY_CONFIG_SBDMA_DISABLE (0 << 6) +#define GALAXY_CONFIG_SBDMA_0 (1 << 6) +#define GALAXY_CONFIG_SBDMA_1 (2 << 6) +#define GALAXY_CONFIG_SBDMA_3 (3 << 6) + +#define GALAXY_CONFIG_WSSA_530 (0 << 8) +#define GALAXY_CONFIG_WSSA_604 (1 << 8) +#define GALAXY_CONFIG_WSSA_E80 (2 << 8) +#define GALAXY_CONFIG_WSSA_F40 (3 << 8) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 10) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 11) + +#define GALAXY_CONFIG_MPUA_300 (0 << 12) +#define GALAXY_CONFIG_MPUA_330 (1 << 12) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 13) + +#define GALAXY_CONFIG_CDA_310 (0 << 14) +#define GALAXY_CONFIG_CDA_320 (1 << 14) +#define GALAXY_CONFIG_CDA_340 (2 << 14) +#define GALAXY_CONFIG_CDA_350 (3 << 14) +#define GALAXY_CONFIG_CDA_MASK GALAXY_CONFIG_CDA_350 + +#define GALAXY_CONFIG_CD_DISABLE (0 << 16) +#define GALAXY_CONFIG_CD_PANASONIC (1 << 16) +#define GALAXY_CONFIG_CD_SONY (2 << 16) +#define GALAXY_CONFIG_CD_MITSUMI (3 << 16) +#define GALAXY_CONFIG_CD_AZTECH (4 << 16) +#define GALAXY_CONFIG_CD_UNUSED_5 (5 << 16) +#define GALAXY_CONFIG_CD_UNUSED_6 (6 << 16) +#define GALAXY_CONFIG_CD_UNUSED_7 (7 << 16) +#define GALAXY_CONFIG_CD_MASK GALAXY_CONFIG_CD_UNUSED_7 + +#define GALAXY_CONFIG_CDDMA8_DISABLE (0 << 20) +#define GALAXY_CONFIG_CDDMA8_0 (1 << 20) +#define GALAXY_CONFIG_CDDMA8_1 (2 << 20) +#define GALAXY_CONFIG_CDDMA8_3 (3 << 20) +#define GALAXY_CONFIG_CDDMA8_MASK GALAXY_CONFIG_CDDMA8_3 + +#define GALAXY_CONFIG_CDDMA16_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA16_5 (1 << 22) +#define GALAXY_CONFIG_CDDMA16_6 (2 << 22) +#define GALAXY_CONFIG_CDDMA16_7 (3 << 22) +#define GALAXY_CONFIG_CDDMA16_MASK GALAXY_CONFIG_CDDMA16_7 + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 24) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 25) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 26) +#define GALAXY_CONFIG_MPUIRQ_10 (1 << 27) + +#define GALAXY_CONFIG_CDIRQ_5 (1 << 28) +#define GALAXY_CONFIG_CDIRQ_11 (1 << 29) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 30) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 31) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_5 | GALAXY_CONFIG_CDIRQ_11 |\ + GALAXY_CONFIG_CDIRQ_12 | GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CDA_MASK |\ + GALAXY_CONFIG_CD_MASK | GALAXY_CONFIG_CDDMA16_MASK |\ + GALAXY_CONFIG_CDDMA8_MASK | GALAXY_CONFIG_CDIRQ_MASK) + +#include "galaxy.c" diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c new file mode 100644 index 000000000000..ee54df082b9c --- /dev/null +++ b/sound/isa/galaxy/galaxy.c @@ -0,0 +1,652 @@ +/* + * Aztech AZT1605/AZT2316 Driver + * Copyright (C) 2007,2010 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see . + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rene Herman"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); + +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param_array(wss_port, long, NULL, 0444); +MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for " CRD_NAME " driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); + +/* + * Generic SB DSP support routines + */ + +#define DSP_PORT_RESET 0x6 +#define DSP_PORT_READ 0xa +#define DSP_PORT_COMMAND 0xc +#define DSP_PORT_STATUS 0xc +#define DSP_PORT_DATA_AVAIL 0xe + +#define DSP_SIGNATURE 0xaa + +#define DSP_COMMAND_GET_VERSION 0xe1 + +static int __devinit dsp_get_byte(void __iomem *port, u8 *val) +{ + int loops = 1000; + + while (!(ioread8(port + DSP_PORT_DATA_AVAIL) & 0x80)) { + if (!loops--) + return -EIO; + cpu_relax(); + } + *val = ioread8(port + DSP_PORT_READ); + return 0; +} + +static int __devinit dsp_reset(void __iomem *port) +{ + u8 val; + + iowrite8(1, port + DSP_PORT_RESET); + udelay(10); + iowrite8(0, port + DSP_PORT_RESET); + + if (dsp_get_byte(port, &val) < 0 || val != DSP_SIGNATURE) + return -ENODEV; + + return 0; +} + +static int __devinit dsp_command(void __iomem *port, u8 cmd) +{ + int loops = 1000; + + while (ioread8(port + DSP_PORT_STATUS) & 0x80) { + if (!loops--) + return -EIO; + cpu_relax(); + } + iowrite8(cmd, port + DSP_PORT_COMMAND); + return 0; +} + +static int __devinit dsp_get_version(void __iomem *port, u8 *major, u8 *minor) +{ + int err; + + err = dsp_command(port, DSP_COMMAND_GET_VERSION); + if (err < 0) + return err; + + err = dsp_get_byte(port, major); + if (err < 0) + return err; + + err = dsp_get_byte(port, minor); + if (err < 0) + return err; + + return 0; +} + +/* + * Generic WSS support routines + */ + +#define WSS_CONFIG_DMA_0 (1 << 0) +#define WSS_CONFIG_DMA_1 (2 << 0) +#define WSS_CONFIG_DMA_3 (3 << 0) +#define WSS_CONFIG_DUPLEX (1 << 2) +#define WSS_CONFIG_IRQ_7 (1 << 3) +#define WSS_CONFIG_IRQ_9 (2 << 3) +#define WSS_CONFIG_IRQ_10 (3 << 3) +#define WSS_CONFIG_IRQ_11 (4 << 3) + +#define WSS_PORT_CONFIG 0 +#define WSS_PORT_SIGNATURE 3 + +#define WSS_SIGNATURE 4 + +static int __devinit wss_detect(void __iomem *wss_port) +{ + if ((ioread8(wss_port + WSS_PORT_SIGNATURE) & 0x3f) != WSS_SIGNATURE) + return -ENODEV; + + return 0; +} + +static void wss_set_config(void __iomem *wss_port, u8 wss_config) +{ + iowrite8(wss_config, wss_port + WSS_PORT_CONFIG); +} + +/* + * Aztech Sound Galaxy specifics + */ + +#define GALAXY_PORT_CONFIG 1024 +#define CONFIG_PORT_SET 4 + +#define DSP_COMMAND_GALAXY_8 8 +#define GALAXY_COMMAND_GET_TYPE 5 + +#define DSP_COMMAND_GALAXY_9 9 +#define GALAXY_COMMAND_WSSMODE 0 +#define GALAXY_COMMAND_SB8MODE 1 + +#define GALAXY_MODE_WSS GALAXY_COMMAND_WSSMODE +#define GALAXY_MODE_SB8 GALAXY_COMMAND_SB8MODE + +struct snd_galaxy { + void __iomem *port; + void __iomem *config_port; + void __iomem *wss_port; + u32 config; + struct resource *res_port; + struct resource *res_config_port; + struct resource *res_wss_port; +}; + +static u32 config[SNDRV_CARDS]; +static u8 wss_config[SNDRV_CARDS]; + +static int __devinit snd_galaxy_match(struct device *dev, unsigned int n) +{ + if (!enable[n]) + return 0; + + switch (port[n]) { + case SNDRV_AUTO_PORT: + dev_err(dev, "please specify port\n"); + return 0; + case 0x220: + config[n] |= GALAXY_CONFIG_SBA_220; + break; + case 0x240: + config[n] |= GALAXY_CONFIG_SBA_240; + break; + case 0x260: + config[n] |= GALAXY_CONFIG_SBA_260; + break; + case 0x280: + config[n] |= GALAXY_CONFIG_SBA_280; + break; + default: + dev_err(dev, "invalid port %#lx\n", port[n]); + return 0; + } + + switch (wss_port[n]) { + case SNDRV_AUTO_PORT: + dev_err(dev, "please specify wss_port\n"); + return 0; + case 0x530: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_530; + break; + case 0x604: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_604; + break; + case 0xe80: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_E80; + break; + case 0xf40: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_F40; + break; + default: + dev_err(dev, "invalid WSS port %#lx\n", wss_port[n]); + return 0; + } + + switch (irq[n]) { + case SNDRV_AUTO_IRQ: + dev_err(dev, "please specify irq\n"); + return 0; + case 7: + wss_config[n] |= WSS_CONFIG_IRQ_7; + break; + case 2: + irq[n] = 9; + case 9: + wss_config[n] |= WSS_CONFIG_IRQ_9; + break; + case 10: + wss_config[n] |= WSS_CONFIG_IRQ_10; + break; + case 11: + wss_config[n] |= WSS_CONFIG_IRQ_11; + break; + default: + dev_err(dev, "invalid IRQ %d\n", irq[n]); + return 0; + } + + switch (dma1[n]) { + case SNDRV_AUTO_DMA: + dev_err(dev, "please specify dma1\n"); + return 0; + case 0: + wss_config[n] |= WSS_CONFIG_DMA_0; + break; + case 1: + wss_config[n] |= WSS_CONFIG_DMA_1; + break; + case 3: + wss_config[n] |= WSS_CONFIG_DMA_3; + break; + default: + dev_err(dev, "invalid playback DMA %d\n", dma1[n]); + return 0; + } + + if (dma2[n] == SNDRV_AUTO_DMA || dma2[n] == dma1[n]) { + dma2[n] = -1; + goto mpu; + } + + wss_config[n] |= WSS_CONFIG_DUPLEX; + switch (dma2[n]) { + case 0: + break; + case 1: + if (dma1[n] == 0) + break; + default: + dev_err(dev, "invalid capture DMA %d\n", dma2[n]); + return 0; + } + +mpu: + switch (mpu_port[n]) { + case SNDRV_AUTO_PORT: + dev_warn(dev, "mpu_port not specified; not using MPU-401\n"); + mpu_port[n] = -1; + goto fm; + case 0x300: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_300; + break; + case 0x330: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_330; + break; + default: + dev_err(dev, "invalid MPU port %#lx\n", mpu_port[n]); + return 0; + } + + switch (mpu_irq[n]) { + case SNDRV_AUTO_IRQ: + dev_warn(dev, "mpu_irq not specified: using polling mode\n"); + mpu_irq[n] = -1; + break; + case 2: + mpu_irq[n] = 9; + case 9: + config[n] |= GALAXY_CONFIG_MPUIRQ_2; + break; +#ifdef AZT1605 + case 3: + config[n] |= GALAXY_CONFIG_MPUIRQ_3; + break; +#endif + case 5: + config[n] |= GALAXY_CONFIG_MPUIRQ_5; + break; + case 7: + config[n] |= GALAXY_CONFIG_MPUIRQ_7; + break; +#ifdef AZT2316 + case 10: + config[n] |= GALAXY_CONFIG_MPUIRQ_10; + break; +#endif + default: + dev_err(dev, "invalid MPU IRQ %d\n", mpu_irq[n]); + return 0; + } + + if (mpu_irq[n] == irq[n]) { + dev_err(dev, "cannot share IRQ between WSS and MPU-401\n"); + return 0; + } + +fm: + switch (fm_port[n]) { + case SNDRV_AUTO_PORT: + dev_warn(dev, "fm_port not specified: not using OPL3\n"); + fm_port[n] = -1; + break; + case 0x388: + break; + default: + dev_err(dev, "illegal FM port %#lx\n", fm_port[n]); + return 0; + } + + config[n] |= GALAXY_CONFIG_GAME_ENABLE; + return 1; +} + +static int __devinit galaxy_init(struct snd_galaxy *galaxy, u8 *type) +{ + u8 major; + u8 minor; + int err; + + err = dsp_reset(galaxy->port); + if (err < 0) + return err; + + err = dsp_get_version(galaxy->port, &major, &minor); + if (err < 0) + return err; + + if (major != GALAXY_DSP_MAJOR || minor != GALAXY_DSP_MINOR) + return -ENODEV; + + err = dsp_command(galaxy->port, DSP_COMMAND_GALAXY_8); + if (err < 0) + return err; + + err = dsp_command(galaxy->port, GALAXY_COMMAND_GET_TYPE); + if (err < 0) + return err; + + err = dsp_get_byte(galaxy->port, type); + if (err < 0) + return err; + + return 0; +} + +static int __devinit galaxy_set_mode(struct snd_galaxy *galaxy, u8 mode) +{ + int err; + + err = dsp_command(galaxy->port, DSP_COMMAND_GALAXY_9); + if (err < 0) + return err; + + err = dsp_command(galaxy->port, mode); + if (err < 0) + return err; + +#ifdef AZT1605 + /* + * Needed for MPU IRQ on AZT1605, but AZT2316 loses WSS again + */ + err = dsp_reset(galaxy->port); + if (err < 0) + return err; +#endif + + return 0; +} + +static void galaxy_set_config(struct snd_galaxy *galaxy, u32 config) +{ + u8 tmp = ioread8(galaxy->config_port + CONFIG_PORT_SET); + int i; + + iowrite8(tmp | 0x80, galaxy->config_port + CONFIG_PORT_SET); + for (i = 0; i < GALAXY_CONFIG_SIZE; i++) { + iowrite8(config, galaxy->config_port + i); + config >>= 8; + } + iowrite8(tmp & 0x7f, galaxy->config_port + CONFIG_PORT_SET); + msleep(10); +} + +static void __devinit galaxy_config(struct snd_galaxy *galaxy, u32 config) +{ + int i; + + for (i = GALAXY_CONFIG_SIZE; i; i--) { + u8 tmp = ioread8(galaxy->config_port + i - 1); + galaxy->config = (galaxy->config << 8) | tmp; + } + config |= galaxy->config & GALAXY_CONFIG_MASK; + galaxy_set_config(galaxy, config); +} + +static int __devinit galaxy_wss_config(struct snd_galaxy *galaxy, u8 wss_config) +{ + int err; + + err = wss_detect(galaxy->wss_port); + if (err < 0) + return err; + + wss_set_config(galaxy->wss_port, wss_config); + + err = galaxy_set_mode(galaxy, GALAXY_MODE_WSS); + if (err < 0) + return err; + + return 0; +} + +static void snd_galaxy_free(struct snd_card *card) +{ + struct snd_galaxy *galaxy = card->private_data; + + if (galaxy->wss_port) { + wss_set_config(galaxy->wss_port, 0); + ioport_unmap(galaxy->wss_port); + release_and_free_resource(galaxy->res_wss_port); + } + if (galaxy->config_port) { + galaxy_set_config(galaxy, galaxy->config); + ioport_unmap(galaxy->config_port); + release_and_free_resource(galaxy->res_config_port); + } + if (galaxy->port) { + ioport_unmap(galaxy->port); + release_and_free_resource(galaxy->res_port); + } +} + +static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) +{ + struct snd_galaxy *galaxy; + struct snd_wss *chip; + struct snd_card *card; + u8 type; + int err; + + err = snd_card_create(index[n], id[n], THIS_MODULE, sizeof *galaxy, + &card); + if (err < 0) + return err; + + snd_card_set_dev(card, dev); + + card->private_free = snd_galaxy_free; + galaxy = card->private_data; + + galaxy->res_port = request_region(port[n], 16, DRV_NAME); + if (!galaxy->res_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", port[n], + port[n] + 15); + err = -EBUSY; + goto error; + } + galaxy->port = ioport_map(port[n], 16); + + err = galaxy_init(galaxy, &type); + if (err < 0) { + dev_err(dev, "did not find a Sound Galaxy at %#lx\n", port[n]); + goto error; + } + dev_info(dev, "Sound Galaxy (type %d) found at %#lx\n", type, port[n]); + + galaxy->res_config_port = request_region(port[n] + GALAXY_PORT_CONFIG, + 16, DRV_NAME); + if (!galaxy->res_config_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", + port[n] + GALAXY_PORT_CONFIG, + port[n] + GALAXY_PORT_CONFIG + 15); + err = -EBUSY; + goto error; + } + galaxy->config_port = ioport_map(port[n] + GALAXY_PORT_CONFIG, 16); + + galaxy_config(galaxy, config[n]); + + galaxy->res_wss_port = request_region(wss_port[n], 4, DRV_NAME); + if (!galaxy->res_wss_port) { + dev_err(dev, "could not grab ports %#lx-%#lx\n", wss_port[n], + wss_port[n] + 3); + err = -EBUSY; + goto error; + } + galaxy->wss_port = ioport_map(wss_port[n], 4); + + err = galaxy_wss_config(galaxy, wss_config[n]); + if (err < 0) { + dev_err(dev, "could not configure WSS\n"); + goto error; + } + + strcpy(card->driver, DRV_NAME); + strcpy(card->shortname, DRV_NAME); + sprintf(card->longname, "%s at %#lx/%#lx, irq %d, dma %d/%d", + card->shortname, port[n], wss_port[n], irq[n], dma1[n], + dma2[n]); + + err = snd_wss_create(card, wss_port[n] + 4, -1, irq[n], dma1[n], + dma2[n], WSS_HW_DETECT, 0, &chip); + if (err < 0) + goto error; + + err = snd_wss_pcm(chip, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(chip); + if (err < 0) + goto error; + + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) + goto error; + + if (mpu_port[n] >= 0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port[n], 0, mpu_irq[n], + IRQF_DISABLED, NULL); + if (err < 0) + goto error; + } + + if (fm_port[n] >= 0) { + struct snd_opl3 *opl3; + + err = snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_AUTO, 0, &opl3); + if (err < 0) { + dev_err(dev, "no OPL device at %#lx\n", fm_port[n]); + goto error; + } + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto error; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto error; + } + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(dev, card); + return 0; + +error: + snd_card_free(card); + return err; +} + +static int __devexit snd_galaxy_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_galaxy_driver = { + .match = snd_galaxy_match, + .probe = snd_galaxy_probe, + .remove = __devexit_p(snd_galaxy_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __init alsa_card_galaxy_init(void) +{ + return isa_register_driver(&snd_galaxy_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_galaxy_exit(void) +{ + isa_unregister_driver(&snd_galaxy_driver); +} + +module_init(alsa_card_galaxy_init); +module_exit(alsa_card_galaxy_exit); -- cgit v1.2.3 From cbaa9f60d5d5c3af10f94e0d49789d5b82341a4a Mon Sep 17 00:00:00 2001 From: René Herman Date: Fri, 13 Aug 2010 10:43:48 +0200 Subject: ALSA: ISA: Remove snd-sgalaxy Its hardware is handled more fully by the new azt1605/azt2316 drivers. Signed-off-by: Rene Herman Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 14 - sound/isa/Kconfig | 10 - sound/isa/Makefile | 2 - sound/isa/sgalaxy.c | 369 ------------------------ 4 files changed, 395 deletions(-) delete mode 100644 sound/isa/sgalaxy.c (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index e25555c02166..d0eb696d32e8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1709,20 +1709,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - Module snd-sgalaxy - ------------------ - - Module for Aztech Sound Galaxy sound card. - - sbport - Port # for SB16 interface (0x220,0x240) - wssport - Port # for WSS interface (0x530,0xe80,0xf40,0x604) - irq - IRQ # (7,9,10,11) - dma1 - DMA # - - This module supports multiple cards. - - The power-management is supported. - Module snd-sscape ----------------- diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index f9aa13d8dacc..52064cfa91f3 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,16 +377,6 @@ config SND_SB16_CSP coprocessor can do variable tasks like various compression and decompression algorithms. -config SND_SGALAXY - tristate "Aztech Sound Galaxy" - select SND_WSS_LIB - help - Say Y here to include support for Aztech Sound Galaxy - soundcards. - - To compile this driver as a module, choose M here: the module - will be called snd-sgalaxy. - config SND_SSCAPE tristate "Ensoniq SoundScape driver" select SND_MPU401_UART diff --git a/sound/isa/Makefile b/sound/isa/Makefile index d2bd8f536dd8..8d781e419e2e 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -10,7 +10,6 @@ snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sc6000-objs := sc6000.o -snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o # Toplevel Module Dependency @@ -21,7 +20,6 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SC6000) += snd-sc6000.o -obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ msnd/ opti9xx/ \ diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c deleted file mode 100644 index 6fe27b9d9440..000000000000 --- a/sound/isa/sgalaxy.c +++ /dev/null @@ -1,369 +0,0 @@ -/* - * Driver for Aztech Sound Galaxy cards - * Copyright (c) by Christopher Butler -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#define SNDRV_LEGACY_FIND_FREE_IRQ -#define SNDRV_LEGACY_FIND_FREE_DMA -#include - -MODULE_AUTHOR("Christopher Butler "); -MODULE_DESCRIPTION("Aztech Sound Galaxy"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Aztech Systems,Sound Galaxy}}"); - -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ -static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240 */ -static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530,0xe80,0xf40,0x604 */ -static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 7,9,10,11 */ -static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for Sound Galaxy soundcard."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for Sound Galaxy soundcard."); -module_param_array(sbport, long, NULL, 0444); -MODULE_PARM_DESC(sbport, "Port # for Sound Galaxy SB driver."); -module_param_array(wssport, long, NULL, 0444); -MODULE_PARM_DESC(wssport, "Port # for Sound Galaxy WSS driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for Sound Galaxy driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "DMA1 # for Sound Galaxy driver."); - -#define SGALAXY_AUXC_LEFT 18 -#define SGALAXY_AUXC_RIGHT 19 - -#define PFX "sgalaxy: " - -/* - - */ - -#define AD1848P1( port, x ) ( port + c_d_c_AD1848##x ) - -/* from lowlevel/sb/sb.c - to avoid having to allocate a struct snd_sb for the */ -/* short time we actually need it.. */ - -static int snd_sgalaxy_sbdsp_reset(unsigned long port) -{ - int i; - - outb(1, SBP1(port, RESET)); - udelay(10); - outb(0, SBP1(port, RESET)); - udelay(30); - for (i = 0; i < 1000 && !(inb(SBP1(port, DATA_AVAIL)) & 0x80); i++); - if (inb(SBP1(port, READ)) != 0xaa) { - snd_printd("sb_reset: failed at 0x%lx!!!\n", port); - return -ENODEV; - } - return 0; -} - -static int __devinit snd_sgalaxy_sbdsp_command(unsigned long port, - unsigned char val) -{ - int i; - - for (i = 10000; i; i--) - if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { - outb(val, SBP1(port, COMMAND)); - return 1; - } - - return 0; -} - -static irqreturn_t snd_sgalaxy_dummy_interrupt(int irq, void *dev_id) -{ - return IRQ_NONE; -} - -static int __devinit snd_sgalaxy_setup_wss(unsigned long port, int irq, int dma) -{ - static int interrupt_bits[] = {-1, -1, -1, -1, -1, -1, -1, 0x08, -1, - 0x10, 0x18, 0x20, -1, -1, -1, -1}; - static int dma_bits[] = {1, 2, 0, 3}; - int tmp, tmp1; - - if ((tmp = inb(port + 3)) == 0xff) - { - snd_printdd("I/O address dead (0x%lx)\n", port); - return 0; - } -#if 0 - snd_printdd("WSS signature = 0x%x\n", tmp); -#endif - - if ((tmp & 0x3f) != 0x04 && - (tmp & 0x3f) != 0x0f && - (tmp & 0x3f) != 0x00) { - snd_printdd("No WSS signature detected on port 0x%lx\n", - port + 3); - return 0; - } - -#if 0 - snd_printdd(PFX "setting up IRQ/DMA for WSS\n"); -#endif - - /* initialize IRQ for WSS codec */ - tmp = interrupt_bits[irq % 16]; - if (tmp < 0) - return -EINVAL; - - if (request_irq(irq, snd_sgalaxy_dummy_interrupt, IRQF_DISABLED, "sgalaxy", NULL)) { - snd_printk(KERN_ERR "sgalaxy: can't grab irq %d\n", irq); - return -EIO; - } - - outb(tmp | 0x40, port); - tmp1 = dma_bits[dma % 4]; - outb(tmp | tmp1, port); - - free_irq(irq, NULL); - - return 0; -} - -static int __devinit snd_sgalaxy_detect(int dev, int irq, int dma) -{ -#if 0 - snd_printdd(PFX "switching to WSS mode\n"); -#endif - - /* switch to WSS mode */ - snd_sgalaxy_sbdsp_reset(sbport[dev]); - - snd_sgalaxy_sbdsp_command(sbport[dev], 9); - snd_sgalaxy_sbdsp_command(sbport[dev], 0); - - udelay(400); - return snd_sgalaxy_setup_wss(wssport[dev], irq, dma); -} - -static struct snd_kcontrol_new snd_sgalaxy_controls[] = { -WSS_DOUBLE("Aux Playback Switch", 0, - SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) -}; - -static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) -{ - struct snd_card *card = chip->card; - struct snd_ctl_elem_id id1, id2; - unsigned int idx; - int err; - - memset(&id1, 0, sizeof(id1)); - memset(&id2, 0, sizeof(id2)); - id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - /* reassign AUX0 to LINE */ - strcpy(id1.name, "Aux Playback Switch"); - strcpy(id2.name, "Line Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - strcpy(id1.name, "Aux Playback Volume"); - strcpy(id2.name, "Line Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - /* reassign AUX1 to FM */ - strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; - strcpy(id2.name, "FM Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - strcpy(id1.name, "Aux Playback Volume"); - strcpy(id2.name, "FM Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) - return err; - /* build AUX2 input */ - for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_sgalaxy_controls[idx], chip)); - if (err < 0) - return err; - } - return 0; -} - -static int __devinit snd_sgalaxy_match(struct device *devptr, unsigned int dev) -{ - if (!enable[dev]) - return 0; - if (sbport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify SB port\n"); - return 0; - } - if (wssport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify WSS port\n"); - return 0; - } - return 1; -} - -static int __devinit snd_sgalaxy_probe(struct device *devptr, unsigned int dev) -{ - static int possible_irqs[] = {7, 9, 10, 11, -1}; - static int possible_dmas[] = {1, 3, 0, -1}; - int err, xirq, xdma1; - struct snd_card *card; - struct snd_wss *chip; - - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (err < 0) - return err; - - xirq = irq[dev]; - if (xirq == SNDRV_AUTO_IRQ) { - if ((xirq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); - err = -EBUSY; - goto _err; - } - } - xdma1 = dma1[dev]; - if (xdma1 == SNDRV_AUTO_DMA) { - if ((xdma1 = snd_legacy_find_free_dma(possible_dmas)) < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); - err = -EBUSY; - goto _err; - } - } - - if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0) - goto _err; - - err = snd_wss_create(card, wssport[dev] + 4, -1, - xirq, xdma1, -1, - WSS_HW_DETECT, 0, &chip); - if (err < 0) - goto _err; - card->private_data = chip; - - err = snd_wss_pcm(chip, 0, NULL); - if (err < 0) { - snd_printdd(PFX "error creating new WSS PCM device\n"); - goto _err; - } - err = snd_wss_mixer(chip); - if (err < 0) { - snd_printdd(PFX "error creating new WSS mixer\n"); - goto _err; - } - if ((err = snd_sgalaxy_mixer(chip)) < 0) { - snd_printdd(PFX "the mixer rewrite failed\n"); - goto _err; - } - - strcpy(card->driver, "Sound Galaxy"); - strcpy(card->shortname, "Sound Galaxy"); - sprintf(card->longname, "Sound Galaxy at 0x%lx, irq %d, dma %d", - wssport[dev], xirq, xdma1); - - snd_card_set_dev(card, devptr); - - if ((err = snd_card_register(card)) < 0) - goto _err; - - dev_set_drvdata(devptr, card); - return 0; - - _err: - snd_card_free(card); - return err; -} - -static int __devexit snd_sgalaxy_remove(struct device *devptr, unsigned int dev) -{ - snd_card_free(dev_get_drvdata(devptr)); - dev_set_drvdata(devptr, NULL); - return 0; -} - -#ifdef CONFIG_PM -static int snd_sgalaxy_suspend(struct device *pdev, unsigned int n, - pm_message_t state) -{ - struct snd_card *card = dev_get_drvdata(pdev); - struct snd_wss *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->suspend(chip); - return 0; -} - -static int snd_sgalaxy_resume(struct device *pdev, unsigned int n) -{ - struct snd_card *card = dev_get_drvdata(pdev); - struct snd_wss *chip = card->private_data; - - chip->resume(chip); - snd_wss_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); - snd_wss_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); - - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif - -#define DEV_NAME "sgalaxy" - -static struct isa_driver snd_sgalaxy_driver = { - .match = snd_sgalaxy_match, - .probe = snd_sgalaxy_probe, - .remove = __devexit_p(snd_sgalaxy_remove), -#ifdef CONFIG_PM - .suspend = snd_sgalaxy_suspend, - .resume = snd_sgalaxy_resume, -#endif - .driver = { - .name = DEV_NAME - }, -}; - -static int __init alsa_card_sgalaxy_init(void) -{ - return isa_register_driver(&snd_sgalaxy_driver, SNDRV_CARDS); -} - -static void __exit alsa_card_sgalaxy_exit(void) -{ - isa_unregister_driver(&snd_sgalaxy_driver); -} - -module_init(alsa_card_sgalaxy_init) -module_exit(alsa_card_sgalaxy_exit) -- cgit v1.2.3 From 4f4e8f69895c8696a4bcc751817d4b186023ac44 Mon Sep 17 00:00:00 2001 From: Paul Zimmerman Date: Fri, 13 Aug 2010 12:42:07 -0700 Subject: ALSA: usb: USB3 SuperSpeed sound support This is V2 of the patch, after feedback from Clemens and Daniel. This patch adds SuperSpeed support to the USB drivers under sound/. It adds tests for USB_SPEED_SUPER to the appropriate places that check for the USB speed. This patch has been tested with our SS USB3 device emulating a set of Yamaha speakers and a Logitech microphone, but with the descriptors modified to add USB3 support. It has also been tested with the real speakers and microphone, to make sure that USB2 devices still work. Signed-off-by: Paul Zimmerman Cc: Clemens Ladisch Cc: Daniel Mack Cc: Greg Kroah-Hartman Signed-off-by: Takashi Iwai --- sound/usb/card.c | 31 +++++++++++++++++++++++-------- sound/usb/helper.c | 17 +++++++++++------ sound/usb/midi.c | 9 ++++++++- sound/usb/pcm.c | 4 ++-- sound/usb/proc.c | 2 +- sound/usb/urb.c | 2 +- 6 files changed, 46 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 9feb00c831a0..498a2d8fa4bb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -299,9 +299,13 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, *rchip = NULL; - if (snd_usb_get_speed(dev) != USB_SPEED_LOW && - snd_usb_get_speed(dev) != USB_SPEED_FULL && - snd_usb_get_speed(dev) != USB_SPEED_HIGH) { + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + case USB_SPEED_FULL: + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + break; + default: snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev)); return -ENXIO; } @@ -377,11 +381,22 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, if (len < sizeof(card->longname)) usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); - strlcat(card->longname, - snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" : - snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" : - ", high speed", - sizeof(card->longname)); + switch (snd_usb_get_speed(dev)) { + case USB_SPEED_LOW: + strlcat(card->longname, ", low speed", sizeof(card->longname)); + break; + case USB_SPEED_FULL: + strlcat(card->longname, ", full speed", sizeof(card->longname)); + break; + case USB_SPEED_HIGH: + strlcat(card->longname, ", high speed", sizeof(card->longname)); + break; + case USB_SPEED_SUPER: + strlcat(card->longname, ", super speed", sizeof(card->longname)); + break; + default: + break; + } snd_usb_audio_create_proc(chip); diff --git a/sound/usb/helper.c b/sound/usb/helper.c index d48d6f8f6ac9..f280c1903c25 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -103,11 +103,16 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface *alts) { - if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - return get_endpoint(alts, 0)->bInterval - 1; - else - return 0; + switch (snd_usb_get_speed(chip->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + if (get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + break; + default: + break; + } + return 0; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index b9c2bc65f51a..156cd0716c42 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -834,7 +834,14 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; + switch (snd_usb_get_speed(ep->umidi->dev)) { + case USB_SPEED_HIGH: + case USB_SPEED_SUPER: + count = 1; + break; + default: + count = 2; + } count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 456829882f40..ebd09acd186e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -467,7 +467,7 @@ static int hw_check_valid_format(struct snd_usb_substream *subs, return 0; } /* check whether the period time is >= the data packet interval */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) { ptime = 125 * (1 << fp->datainterval); if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); @@ -735,7 +735,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre } param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) /* full speed devices have fixed data packet interval */ ptmin = 1000; if (ptmin == 1000) diff --git a/sound/usb/proc.c b/sound/usb/proc.c index f5e3f356b95f..3c650ab3c91d 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -107,7 +107,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) snd_iprintf(buffer, " Data packet interval: %d us\n", 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); diff --git a/sound/usb/urb.c b/sound/usb/urb.c index de607d4411ac..8deeaad10f10 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -244,7 +244,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->curpacksize = maxsize; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + if (snd_usb_get_speed(subs->dev) != USB_SPEED_FULL) packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; -- cgit v1.2.3 From a5ba6beb839cfa288960c92cd2668a2601c24dda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Aug 2010 08:08:48 +0200 Subject: ALSA: riptide - Fix detection / load of firmware files The detection and loading of firmeware on riptide driver has been broken due to rewrite of some codes, checking the presense wrongly. This patch fixes the logic again. Reference: kernel bug 16596 https://bugzilla.kernel.org/show_bug.cgi?id=16596 Cc: Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index f64fb7d988cb..ad5202efd7a9 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) firmware.firmware.ASIC, firmware.firmware.CODEC, firmware.firmware.AUXDSP, firmware.firmware.PROG); + if (!chip) + return 1; + for (i = 0; i < FIRMWARE_VERSIONS; i++) { if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware))) - break; - } - if (i >= FIRMWARE_VERSIONS) - return 0; /* no match */ + return 1; /* OK */ - if (!chip) - return 1; /* OK */ + } snd_printdd("Writing Firmware\n"); if (!chip->fw_entry) { -- cgit v1.2.3 From c3e68fad88143fd1fe8fe640207fb19c0f087dbc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Aug 2010 10:15:57 +0200 Subject: ALSA: hda - Add quirk for Dell Vostro 1220 model=dell-vostro is needed for Dell Vostro 1220 with Coexnat 5067. Reference: Novell bnc#631066 https://bugzilla.novell.com/show_bug.cgi?id=631066 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 31b5d9eeba68..c424952a734e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), -- cgit v1.2.3 From c69aefabe004d24e6eedf83b6f253647f77dfc43 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 17 Aug 2010 10:39:22 +0200 Subject: ALSA: hda - Fix ALC680 base model capture - Fix capture mixer elements for ALC680 base model - Support auto change ADC for recording from MIC - Cancel capture source assigned in auto mode. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 176 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 144 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2cd1ae809e46..a4dd04524e43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19030,6 +19030,7 @@ static int patch_alc888(struct hda_codec *codec) /* * ALC680 support */ +#define ALC680_DIGIN_NID ALC880_DIGIN_NID #define ALC680_DIGOUT_NID ALC880_DIGOUT_NID #define alc680_modes alc260_modes @@ -19044,23 +19045,93 @@ static hda_nid_t alc680_adc_nids[3] = { 0x07, 0x08, 0x09 }; +/* + * Analog capture ADC cgange + */ +static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int pre_mic, pre_line; + + pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]); + + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + + if (pre_mic || pre_line) { + if (pre_mic) + snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0, + format); + else + snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0, + format); + } else + snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format); + return 0; +} + +static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, 0x07); + snd_hda_codec_cleanup_stream(codec, 0x08); + snd_hda_codec_cleanup_stream(codec, 0x09); + return 0; +} + +static struct hda_pcm_stream alc680_pcm_analog_auto_capture = { + .substreams = 1, /* can be overridden */ + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ + .ops = { + .prepare = alc680_capture_pcm_prepare, + .cleanup = alc680_capture_pcm_cleanup + }, +}; + static struct snd_kcontrol_new alc680_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT), { } }; -static struct snd_kcontrol_new alc680_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), +static struct hda_bind_ctls alc680_bind_cap_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc680_bind_cap_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc680_master_capture_mixer[] = { + HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol), + HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch), { } /* end */ }; @@ -19068,25 +19139,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = { * generic initialization of ADC, input mixers and output mixers */ static struct hda_verb alc680_init_verbs[] = { - /* Unmute DAC0-1 and set vol = 0 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc680_base_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x16; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18; + spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19; +} + +static void alc680_rec_autoswitch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + + new_adc = present ? 0x8 : 0x7; + __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1); + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + +} + +static void alc680_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc_automute_amp(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc680_rec_autoswitch(codec); +} + +static void alc680_inithook(struct hda_codec *codec) +{ + alc_automute_amp(codec); + alc680_rec_autoswitch(codec); +} + /* create input playback/capture controls for the given pin */ static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) @@ -19197,13 +19316,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec) #define alc680_pcm_analog_capture alc880_pcm_analog_capture #define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc680_pcm_digital_playback alc880_pcm_digital_playback - -static struct hda_input_mux alc680_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x0 }, - }, -}; +#define alc680_pcm_digital_capture alc880_pcm_digital_capture /* * BIOS auto configuration @@ -19218,6 +19331,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec) alc680_ignore); if (err < 0) return err; + if (!spec->autocfg.line_outs) { if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { spec->multiout.max_channels = 2; @@ -19239,8 +19353,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); add_verb(spec, alc680_init_verbs); - spec->num_mux_defs = 1; - spec->input_mux = &alc680_capture_source; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -19279,17 +19391,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = { static struct alc_config_preset alc680_presets[] = { [ALC680_BASE] = { .mixers = { alc680_base_mixer }, - .cap_mixer = alc680_capture_mixer, + .cap_mixer = alc680_master_capture_mixer, .init_verbs = { alc680_init_verbs }, .num_dacs = ARRAY_SIZE(alc680_dac_nids), .dac_nids = alc680_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc680_adc_nids), - .adc_nids = alc680_adc_nids, - .hp_nid = 0x04, .dig_out_nid = ALC680_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc680_modes), .channel_mode = alc680_modes, - .input_mux = &alc680_capture_source, + .unsol_event = alc680_unsol_event, + .setup = alc680_base_setup, + .init_hook = alc680_inithook, + }, }; @@ -19333,9 +19445,9 @@ static int patch_alc680(struct hda_codec *codec) setup_preset(codec, &alc680_presets[board_config]); spec->stream_analog_playback = &alc680_pcm_analog_playback; - spec->stream_analog_capture = &alc680_pcm_analog_capture; - spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture; + spec->stream_analog_capture = &alc680_pcm_analog_auto_capture; spec->stream_digital_playback = &alc680_pcm_digital_playback; + spec->stream_digital_capture = &alc680_pcm_digital_capture; if (!spec->adc_nids) { spec->adc_nids = alc680_adc_nids; -- cgit v1.2.3 From 56385a12d9bb9e173751f74b6c430742018cafc0 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Aug 2010 14:08:17 +0200 Subject: ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter) With some hardware combinations, the PCM interrupts are acknowledged before the period boundary from the emu10k1 chip. The midlevel PCM code gets confused and the playback stream is interrupted. It seems that the interrupt processing shift by 2 samples is enough to fix this issue. This default value does not harm other, non-affected hardware. More information: Kernel bugzilla bug#16300 [A copmile warning fixed by tiwai] Signed-off-by: Jaroslav Kysela Cc: Signed-off-by: Takashi Iwai --- include/sound/emu10k1.h | 1 + sound/core/pcm_native.c | 4 ++++ sound/pci/emu10k1/emu10k1.c | 4 ++++ sound/pci/emu10k1/emupcm.c | 30 ++++++++++++++++++++++++++---- sound/pci/emu10k1/memory.c | 4 +++- 5 files changed, 38 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 6a664c3f7c1e..7dc97d12253c 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -1707,6 +1707,7 @@ struct snd_emu10k1 { unsigned int card_type; /* EMU10K1_CARD_* */ unsigned int ecard_ctrl; /* ecard control bits */ unsigned long dma_mask; /* PCI DMA mask */ + unsigned int delay_pcm_irq; /* in samples */ int max_cache_pages; /* max memory size / PAGE_SIZE */ struct snd_dma_buffer silent_page; /* silent page */ struct snd_dma_buffer ptb_pages; /* page table pages */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a3b2a6479246..134fc6c2e08d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) { if (substream->runtime->trigger_master != substream) return 0; + /* some drivers might use hw_ptr to recover from the pause - + update the hw_ptr now */ + if (push) + snd_pcm_update_hw_ptr(substream); /* The jiffies check in snd_pcm_update_hw_ptr*() is done by * a delta betwen the current jiffies, this gives a large enough * delta, effectively to skip the check once. diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 4203782d7cb7..aff8387c45cf 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; static int enable_ir[SNDRV_CARDS]; static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ +static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); @@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444); MODULE_PARM_DESC(enable_ir, "Enable IR."); module_param_array(subsystem, uint, NULL, 0444); MODULE_PARM_DESC(subsystem, "Force card subsystem model."); +module_param_array(delay_pcm_irq, uint, NULL, 0444); +MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0)."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ @@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, &emu)) < 0) goto error; card->private_data = emu; + emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f; if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0) goto error; if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 55b83ef73c63..622bace148e3 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, evoice->epcm->ccca_start_addr = start_addr + ccis; if (extra) { start_addr += ccis; - end_addr += ccis; + end_addr += ccis + emu->delay_pcm_irq; } if (stereo && !extra) { snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK); @@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* Assumption that PT is already 0 so no harm overwriting */ snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); - snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); + snd_emu10k1_ptr_write(emu, PSST, voice, + (start_addr + (extra ? emu->delay_pcm_irq : 0)) | + (send_amount[2] << 24)); if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ else @@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_ snd_emu10k1_ptr_write(emu, IP, voice, 0); } +static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu, + struct snd_emu10k1_pcm *epcm, + struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) +{ + unsigned int ptr, period_pos; + + /* try to sychronize the current position for the interrupt + source voice */ + period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt; + period_pos %= runtime->period_size; + ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number); + ptr &= ~0x00ffffff; + ptr |= epcm->ccca_start_addr + period_pos; + snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr); +} + static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, /* follow thru */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: + if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) + snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime); mix = &emu->pcm_mixer[substream->number]; snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix); snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix); @@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * #endif /* printk(KERN_DEBUG - "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", - ptr, runtime->buffer_size, runtime->period_size); + "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n", + (long)ptr, (long)runtime->buffer_size, + (long)runtime->period_size); */ return ptr; } diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index ffb1ddb8dc28..957a311514c8 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst if (snd_BUG_ON(!hdr)) return NULL; + idx = runtime->period_size >= runtime->buffer_size ? + (emu->delay_pcm_irq * 2) : 0; mutex_lock(&hdr->block_mutex); - blk = search_empty(emu, runtime->dma_bytes); + blk = search_empty(emu, runtime->dma_bytes + idx); if (blk == NULL) { mutex_unlock(&hdr->block_mutex); return NULL; -- cgit v1.2.3 From bd76af0f87f7a1815b311bde269a3f18305b3169 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Aug 2010 14:16:54 +0200 Subject: ALSA: pcm midlevel code - add time check for double interrupt acknowledge The current code in pcm_lib.c do all checks using only the position in the ring buffer. Unfortunately, where the interrupts gets delayed or merged into one, we need another timing source to check when the buffer size boundary overlaps to avoid the wrong updating of the ring buffer pointers. This code uses jiffies to check the right time window without any performance impact. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 14 +++++++++----- sound/core/pcm_native.c | 2 ++ 3 files changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 85f1c6bf8566..dfd9b76b1853 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -278,6 +278,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e23e0e7ab26f..a1707cca9c66 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -334,11 +334,15 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - goto __delta; + /* check for double acknowledged interrupts */ + hdelta = jiffies - runtime->hw_ptr_jiffies; + if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + goto __delta; + } } } /* new_hw_ptr might be lower than old_hw_ptr in case when */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 134fc6c2e08d..e2e73895db12 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -864,6 +864,8 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; + runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / + runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) -- cgit v1.2.3 From 4d8ec5f3b65dd64fa785192dc7ab2807916a05b2 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 19 Aug 2010 08:06:16 +0200 Subject: ALSA: hda - Add support for IDT 92HD89XX codecs Just added new codec ids. These are almost compatible with existing ones. Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3f861bd1bf8..95148e58026c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6303,6 +6303,21 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, + { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, + { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, {} /* terminator */ }; -- cgit v1.2.3 From 274714f55c023c683a6b2deedfb2209a9457f4ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Aug 2010 08:11:53 +0200 Subject: ALSA: hda - Fix build error with CONFIG_PROC_FS=n hdmi_eld_update_pcm_info() must be always compiled in. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 803b298f7411..26c3ade73583 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -596,6 +596,8 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) } EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); +#endif /* CONFIG_PROC_FS */ + /* update PCM info based on ELD */ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, struct hda_pcm_stream *codec_pars) @@ -644,5 +646,3 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info); - -#endif /* CONFIG_PROC_FS */ -- cgit v1.2.3 From 9c77b846ec8b4e0c7107dd7f820172462dc84a61 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Wed, 18 Aug 2010 19:33:43 -0400 Subject: ALSA: intel8x0: Mute External Amplifier by default for ThinkPad X31 BugLink: https://bugs.launchpad.net/bugs/619439 This ThinkPad model needs External Amplifier muted for audible playback, so set the inv_eapd quirk for it. Reported-and-tested-by: Dennis Bell Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6433e65c9507..467749249576 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1774,6 +1774,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "HP/Compaq nx7010", .type = AC97_TUNE_MUTE_LED }, + { + .subvendor = 0x1014, + .subdevice = 0x0534, + .name = "ThinkPad X31", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x1014, .subdevice = 0x1f00, -- cgit v1.2.3 From d7d28bc29f4ea7c2d23ed002a9973c64a92bcdb8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Aug 2010 14:16:54 +0200 Subject: ALSA: pcm midlevel code - add time check for double interrupt acknowledge The current code in pcm_lib.c do all checks using only the position in the ring buffer. Unfortunately, where the interrupts gets delayed or merged into one, we need another timing source to check when the buffer size boundary overlaps to avoid the wrong updating of the ring buffer pointers. This code uses jiffies to check the right time window without any performance impact. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 14 +++++++++----- sound/core/pcm_native.c | 2 ++ 3 files changed, 12 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cdede64d..54c4ccf6fec2 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -274,6 +274,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e9d98be190c5..d6ecca27bb68 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -329,11 +329,15 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - goto __delta; + /* check for double acknowledged interrupts */ + hdelta = jiffies - runtime->hw_ptr_jiffies; + if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + goto __delta; + } } } /* new_hw_ptr might be lower than old_hw_ptr in case when */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 303ac04ff6e4..2d2e1b65ee9a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -867,6 +867,8 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; + runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / + runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) -- cgit v1.2.3 From 4f34760787c3751a3146f0eecdc79c3e97b94962 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Aug 2010 09:41:59 +0200 Subject: ALSA: hda - Fix conflict of sticky PCM parameter in HDMI codecs Intel and Nvidia HDMI codec drivers have own implementations of sticky PCM parameters. Now HD-audio core part already has it, thus both setups conflict. The fix is simply remove the part in patch_intelhdmi.c and patch_nvhdmi.c and simply call snd_hda_codec_setup_stream() as usual. Reported-and-tested-by: Stephen Warren Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 21 +-------------------- sound/pci/hda/patch_intelhdmi.c | 8 -------- sound/pci/hda/patch_nvhdmi.c | 8 -------- 3 files changed, 1 insertion(+), 36 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2bc0f07cf33f..afd6022a96a7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -707,8 +707,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int format) { struct hdmi_spec *spec = codec->spec; - int tag; - int fmt; int pinctl; int new_pinctl = 0; int i; @@ -745,24 +743,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, return -EINVAL; } - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index d382d3c81c0f..36a9b83a6174 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -69,20 +69,12 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); } -static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .ops = { .open = hdmi_pcm_open, .prepare = intel_hdmi_playback_pcm_prepare, - .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index f636870dc718..69b950d527c3 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -326,13 +326,6 @@ static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, return 0; } -static int nvhdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - return 0; -} - static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -350,7 +343,6 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { .ops = { .open = hdmi_pcm_open, .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, - .cleanup = nvhdmi_playback_pcm_cleanup, }, }; -- cgit v1.2.3 From 3f50ac6a0ec80a83a1a033fe5004fb319ad72db7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Aug 2010 09:44:36 +0200 Subject: ALSA: hda - Fix stream and channel-ids codec-bus wide The new sticky PCM parameter introduced the delayed clean-ups of stream- and channel-id tags. In the current implementation, this check (adding dirty flag) and actual clean-ups are done only for the codec chip. However, with HD-audio architecture, multiple codecs can be on a single bus, and the controller assign stream- and channel-ids in the bus-wide. In this patch, the stream-id and channel-id are checked over all codecs connected to the corresponding bus. Together with it, the mutex is moved to struct hda_bus, as this becomes also bus-wide. Reported-and-tested-by: Stephen Warren Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++++++------------- sound/pci/hda/hda_codec.h | 2 +- 2 files changed, 21 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index dd8fb86c842b..3827092cc1d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -589,6 +589,7 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); + mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); snprintf(bus->workq_name, sizeof(bus->workq_name), @@ -1068,7 +1069,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); - mutex_init(&codec->prepare_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); @@ -1213,6 +1213,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag, int channel_id, int format) { + struct hda_codec *c; struct hda_cvt_setup *p; unsigned int oldval, newval; int i; @@ -1253,10 +1254,12 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p->dirty = 0; /* make other inactive cvts with the same stream-tag dirty */ - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); - if (!p->active && p->stream_tag == stream_tag) - p->dirty = 1; + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + p = snd_array_elem(&c->cvt_setups, i); + if (!p->active && p->stream_tag == stream_tag) + p->dirty = 1; + } } } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); @@ -1306,12 +1309,16 @@ static void really_cleanup_stream(struct hda_codec *codec, /* clean up the all conflicting obsolete streams */ static void purify_inactive_streams(struct hda_codec *codec) { + struct hda_codec *c; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); - if (p->dirty) - really_cleanup_stream(codec, p); + list_for_each_entry(c, &codec->bus->codec_list, list) { + for (i = 0; i < c->cvt_setups.used; i++) { + struct hda_cvt_setup *p; + p = snd_array_elem(&c->cvt_setups, i); + if (p->dirty) + really_cleanup_stream(c, p); + } } } @@ -3502,11 +3509,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, struct snd_pcm_substream *substream) { int ret; - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); if (ret >= 0) purify_inactive_streams(codec); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); return ret; } EXPORT_SYMBOL_HDA(snd_hda_codec_prepare); @@ -3515,9 +3522,9 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { - mutex_lock(&codec->prepare_mutex); + mutex_lock(&codec->bus->prepare_mutex); hinfo->ops.cleanup(hinfo, codec, substream); - mutex_unlock(&codec->prepare_mutex); + mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4303353feda9..62c702240108 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -648,6 +648,7 @@ struct hda_bus { struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; struct mutex cmd_mutex; + struct mutex prepare_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -826,7 +827,6 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; - struct mutex prepare_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ -- cgit v1.2.3 From 23b224d9d42a111ce451e4300304415a0ba5da75 Mon Sep 17 00:00:00 2001 From: Garnet MacPhee Date: Sat, 21 Aug 2010 14:37:34 -0600 Subject: ALSA: ice1712: Add support for Edirol DA-2496 This device is similar to the M-Audio Delta 1010LT in that it uses the AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF. The SPDIF appears to be set up correctly, but I am not able to test it as I do not have any devices that use it. This patch makes the ADC/DAC's and the hardware mixer visible to apps such as alsamixer and envy24control. Signed-off-by: Garnet MacPhee Signed-off-by: Takashi Iwai --- sound/pci/ice1712/delta.c | 10 ++++++++++ sound/pci/ice1712/delta.h | 4 +++- 2 files changed, 13 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index d216362626d0..712c1710f9a2 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -563,6 +563,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA1010E: case ICE1712_SUBDEVICE_DELTA1010LT: case ICE1712_SUBDEVICE_MEDIASTATION: + case ICE1712_SUBDEVICE_EDIROLDA2496: ice->num_total_dacs = 8; ice->num_total_adcs = 8; break; @@ -635,6 +636,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) err = snd_ice1712_akm4xxx_init(ak, &akm_delta410, &akm_delta410_priv, ice); break; case ICE1712_SUBDEVICE_DELTA1010LT: + case ICE1712_SUBDEVICE_EDIROLDA2496: err = snd_ice1712_akm4xxx_init(ak, &akm_delta1010lt, &akm_delta1010lt_priv, ice); break; case ICE1712_SUBDEVICE_DELTA66: @@ -734,6 +736,7 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) case ICE1712_SUBDEVICE_DELTA66: case ICE1712_SUBDEVICE_VX442: case ICE1712_SUBDEVICE_DELTA66E: + case ICE1712_SUBDEVICE_EDIROLDA2496: err = snd_ice1712_akm4xxx_build_controls(ice); if (err < 0) return err; @@ -813,5 +816,12 @@ struct snd_ice1712_card_info snd_ice1712_delta_cards[] __devinitdata = { .chip_init = snd_ice1712_delta_init, .build_controls = snd_ice1712_delta_add_controls, }, + { + .subvendor = ICE1712_SUBDEVICE_EDIROLDA2496, + .name = "Edirol DA2496", + .model = "da2496", + .chip_init = snd_ice1712_delta_init, + .build_controls = snd_ice1712_delta_add_controls, + }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/delta.h b/sound/pci/ice1712/delta.h index f7f14df81f26..1a0ac6cd6501 100644 --- a/sound/pci/ice1712/delta.h +++ b/sound/pci/ice1712/delta.h @@ -34,7 +34,8 @@ "{MidiMan M Audio,Delta 410},"\ "{MidiMan M Audio,Audiophile 24/96},"\ "{Digigram,VX442},"\ - "{Lionstracs,Mediastation}," + "{Lionstracs,Mediastation},"\ + "{Edirol,DA2496}," #define ICE1712_SUBDEVICE_DELTA1010 0x121430d6 #define ICE1712_SUBDEVICE_DELTA1010E 0xff1430d6 @@ -47,6 +48,7 @@ #define ICE1712_SUBDEVICE_DELTA1010LT 0x12143bd6 #define ICE1712_SUBDEVICE_VX442 0x12143cd6 #define ICE1712_SUBDEVICE_MEDIASTATION 0x694c0100 +#define ICE1712_SUBDEVICE_EDIROLDA2496 0xce164010 /* entry point */ extern struct snd_ice1712_card_info snd_ice1712_delta_cards[]; -- cgit v1.2.3 From 6f0ef6ea1d11ef242de584e345355b0de756fcb2 Mon Sep 17 00:00:00 2001 From: Jerone Young Date: Mon, 23 Aug 2010 08:34:36 +0200 Subject: ALSA: hda - Add support for Lenovo S10-3t This patch adds quirk for the Lenovo S10-3t so the headphone & microphone jacks will now work. Signed-off-by: Jerone Young Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c424952a734e..5cdb80edbd7f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3059,6 +3059,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), {} -- cgit v1.2.3 From dbbcbc073ad3132bfbc410b11546b2fb4bdf2568 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 23 Aug 2010 08:14:35 +0200 Subject: ALSA: hda - Add Sony VAIO quirk for ALC269 The attached patch enables playback on a Sony VAIO machine. BugLink: http://launchpad.net/bugs/618271 Signed-off-by: David Henningsson Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4dd04524e43..627bf9963368 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14467,6 +14467,7 @@ static const struct alc_fixup alc269_fixups[] = { static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), {} }; -- cgit v1.2.3 From 60f1deb595c08687a96157a6a3ce08ef34142362 Mon Sep 17 00:00:00 2001 From: Eliot Blennerhassett Date: Sat, 28 Aug 2010 19:52:24 +1200 Subject: ALSA: asihpi - Return hw error directly from oustream_write. If hw error is ignored, status is updated with invalid info. Signed-off-by: Eliot Blennerhassett Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpi6205.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 3b4413448226..22c5fc625533 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,8 +941,7 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static u32 outstream_get_space_available(struct hpi_hostbuffer_status - *status) +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { return status->size_in_bytes - (status->host_index - status->dSP_index); @@ -987,6 +986,10 @@ static void outstream_write(struct hpi_adapter_obj *pao, /* write it */ phm->function = HPI_OSTREAM_WRITE; hw_message(pao, phm, phr); + + if (phr->error) + return; + /* update status information that the DSP would typically * update (and will update next time the DSP * buffer update task reads data from the host BBM buffer) -- cgit v1.2.3 From 3182c8a72b31414e043184a97b0d5d3c0d590168 Mon Sep 17 00:00:00 2001 From: Akinobu Mita Date: Sat, 28 Aug 2010 13:25:33 +0900 Subject: sound: oss: fix uninitialized spinlock The spinlock lock in sound_timer.c is used without initialization. Signed-off-by: Akinobu Mita Signed-off-by: Takashi Iwai --- sound/oss/sound_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index f0f0c19fbff7..48cda6c4c257 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -26,7 +26,7 @@ static unsigned long prev_event_time; static volatile unsigned long usecs_per_tmr; /* Length of the current interval */ static struct sound_lowlev_timer *tmr; -static spinlock_t lock; +static DEFINE_SPINLOCK(lock); static unsigned long tmr2ticks(int tmr_value) { -- cgit v1.2.3 From 7a28826ac73d31a379d93785d8fbd24ab492b0bd Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 27 Aug 2010 22:02:15 +0200 Subject: ALSA: pcm: add more format names There were some new formats added in commit 15c0cee6c809 "ALSA: pcm: Define G723 3-bit and 5-bit formats". That commit increased SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple places which do: for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } I haven't tested these but it looks like if "i" were equal to SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of the array. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cbe815dfbdc8..204af48c5cc1 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -203,10 +203,16 @@ static char *snd_pcm_format_names[] = { FORMAT(S18_3BE), FORMAT(U18_3LE), FORMAT(U18_3BE), + FORMAT(G723_24), + FORMAT(G723_24_1B), + FORMAT(G723_40), + FORMAT(G723_40_1B), }; const char *snd_pcm_format_name(snd_pcm_format_t format) { + if (format >= ARRAY_SIZE(snd_pcm_format_names)) + return "Unknown"; return snd_pcm_format_names[format]; } EXPORT_SYMBOL_GPL(snd_pcm_format_name); -- cgit v1.2.3 From 73413b120d5d6eb6c98451bbc19acf43e0e300ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 09:39:57 +0200 Subject: ALSA: hda - embed alc_fixup contents into struct definitions Instead of defining each content as a separate struct, put all into the definition of struct alc_fixup arrays so that reader doesn't go back to see the definition again. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 70 ++++++++++++++++++------------------------- 1 file changed, 29 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 627bf9963368..50e0c82fd994 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6799,14 +6799,12 @@ enum { PINFIX_HP_DC5750, }; -static struct alc_pincfg alc260_hp_dc5750_pinfix[] = { - { 0x11, 0x90130110 }, /* speaker */ - { } -}; - static const struct alc_fixup alc260_fixups[] = { [PINFIX_HP_DC5750] = { - .pins = alc260_hp_dc5750_pinfix + .pins = (const struct alc_pincfg[]) { + { 0x11, 0x90130110 }, /* speaker */ + { } + } }, }; @@ -10452,24 +10450,20 @@ enum { PINFIX_PB_M5210, }; -static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { - { 0x15, 0x01080104 }, /* side */ - { 0x16, 0x01011012 }, /* rear */ - { 0x17, 0x01016011 }, /* clfe */ - { } -}; - -static const struct hda_verb pb_m5210_verbs[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, - {} -}; - static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { - .pins = alc882_abit_aw9d_pinfix + .pins = (const struct alc_pincfg[]) { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } + } }, [PINFIX_PB_M5210] = { - .verbs = pb_m5210_verbs + .verbs = (const struct hda_verb[]) { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + {} + } }, }; @@ -14454,14 +14448,12 @@ enum { ALC269_FIXUP_SONY_VAIO, }; -static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, - {} -}; - static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { - .verbs = alc269_sony_vaio_fixup_verbs + .verbs = (const struct hda_verb[]) { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + {} + } }, }; @@ -15819,15 +15811,13 @@ enum { PINFIX_FSC_AMILO_PI1505, }; -static struct alc_pincfg alc861_fsc_amilo_pi1505_pinfix[] = { - { 0x0b, 0x0221101f }, /* HP */ - { 0x0f, 0x90170310 }, /* speaker */ - { } -}; - static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { - .pins = alc861_fsc_amilo_pi1505_pinfix + .pins = (const struct alc_pincfg[]) { + { 0x0b, 0x0221101f }, /* HP */ + { 0x0f, 0x90170310 }, /* speaker */ + { } + } }, }; @@ -16794,16 +16784,14 @@ enum { }; /* reset GPIO1 */ -static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = { - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - { } -}; - static const struct alc_fixup alc861vd_fixups[] = { [ALC660VD_FIX_ASUS_GPIO1] = { - .verbs = alc660vd_fix_asus_gpio1_verbs, + .verbs = (const struct hda_verb[]) { + {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + { } + } }, }; -- cgit v1.2.3 From f3268512c3a5dea587cfe875b8bca98d9e164cd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 11:00:19 +0200 Subject: ALSA: hda - Refactor input-pin parser for VIA codecs patch_via.c has redundant codes for parsing the input-pins. Although they are pretty similar, but all implemented in different functions just because of hard-coded ids and slight incompatibilities. This patch refactors the codes to use the common helper function, resulting in the reduction of many lines. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 390 +++++++--------------------------------------- 1 file changed, 60 insertions(+), 330 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ae3acb2b42d1..41861388f43a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2413,51 +2413,53 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, - const struct auto_pin_cfg *cfg) +static int vt_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg, + hda_nid_t cap_nid, + hda_nid_t pin_idxs[], int num_idxs) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; + int i, err, idx; /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = idx; - imux->num_items++; + for (idx = 0; idx < num_idxs; idx++) { + if (pin_idxs[idx] == 0xff) { + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = idx; + imux->num_items++; + break; + } + } for (i = 0; i < AUTO_PIN_LAST; i++) { if (!cfg->input_pins[i]) continue; - switch (cfg->input_pins[i]) { - case 0x1d: /* Mic */ - idx = 2; - break; - - case 0x1e: /* Line In */ - idx = 3; - break; - - case 0x21: /* Front Mic */ - idx = 4; - break; - - case 0x24: /* CD */ - idx = 1; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x17); + for (idx = 0; idx < num_idxs; idx++) + if (pin_idxs[idx] == cfg->input_pins[i]) + break; + if (idx >= num_idxs) + continue; + err = via_new_analog_input(spec, auto_pin_cfg_labels[i], + idx, cap_nid); if (err < 0) return err; - imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx; imux->num_items++; } return 0; } +/* create playback/capture controls for input pins */ +static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x17, pin_idxs, + ARRAY_SIZE(pin_idxs)); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE static struct hda_amp_list vt1708_loopbacks[] = { { 0x17, HDA_INPUT, 1 }, @@ -3024,46 +3026,9 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = idx; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x1d: /* Mic */ - idx = 2; - break; - - case 0x1e: /* Line In */ - idx = 3; - break; - - case 0x21: /* Front Mic */ - idx = 4; - break; - - case 0x23: /* CD */ - idx = 1; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x18); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x18, pin_idxs, + ARRAY_SIZE(pin_idxs)); } static int vt1709_parse_auto_config(struct hda_codec *codec) @@ -3591,46 +3556,9 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = idx; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x1a: /* Mic */ - idx = 2; - break; - - case 0x1b: /* Line In */ - idx = 3; - break; - - case 0x1e: /* Front Mic */ - idx = 4; - break; - - case 0x1f: /* CD */ - idx = 1; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x16); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + ARRAY_SIZE(pin_idxs)); } static int vt1708B_parse_auto_config(struct hda_codec *codec) @@ -4064,46 +3992,9 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 5; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x1a: /* Mic */ - idx = 2; - break; - - case 0x1b: /* Line In */ - idx = 3; - break; - - case 0x1e: /* Front Mic */ - idx = 4; - break; - - case 0x1f: /* CD */ - idx = 1; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x16); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx-1; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + ARRAY_SIZE(pin_idxs)); } /* fill out digital output widgets; one for master and one for slave outputs */ @@ -4457,42 +4348,9 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 3; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x14: /* Mic */ - idx = 1; - break; - - case 0x15: /* Line In */ - idx = 2; - break; - - case 0x18: /* Front Mic */ - idx = 3; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x1A); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx-1; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x1a, pin_idxs, + ARRAY_SIZE(pin_idxs)); } static int vt1702_parse_auto_config(struct hda_codec *codec) @@ -4875,46 +4733,9 @@ static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 5; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x2b: /* Mic */ - idx = 1; - break; - - case 0x2a: /* Line In */ - idx = 2; - break; - - case 0x29: /* Front Mic */ - idx = 3; - break; - - case 0x2c: /* CD */ - idx = 0; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x21); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + ARRAY_SIZE(pin_idxs)); } static int vt1718S_parse_auto_config(struct hda_codec *codec) @@ -5374,46 +5195,9 @@ static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 5; - imux->num_items++; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x1a: /* Mic */ - idx = 2; - break; - - case 0x1b: /* Line In */ - idx = 3; - break; - - case 0x1e: /* Front Mic */ - idx = 4; - break; - - case 0x1f: /* CD */ - idx = 1; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x16); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx-1; - imux->num_items++; - } - return 0; + static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; + return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + ARRAY_SIZE(pin_idxs)); } static int vt1716S_parse_auto_config(struct hda_codec *codec) @@ -5720,47 +5504,19 @@ static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x2b: /* Mic */ - idx = 0; - break; - - case 0x2a: /* Line In */ - idx = 1; - break; - - case 0x29: /* Front Mic */ - idx = 2; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x21); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } + static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff }; + int err; + err = vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + ARRAY_SIZE(pin_idxs)); + if (err < 0) + return err; /* build volume/mute control of loopback */ err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); if (err < 0) return err; - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 3; - imux->num_items++; - /* for digital mic select */ imux->items[imux->num_items].label = "Digital Mic"; imux->items[imux->num_items].index = 4; @@ -6070,46 +5826,20 @@ static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, const struct auto_pin_cfg *cfg) { - static char *labels[] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL - }; struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx = 0; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - - switch (cfg->input_pins[i]) { - case 0x2b: /* Mic */ - idx = 0; - break; + static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff }; + int err; - case 0x2a: /* Line In */ - idx = 1; - break; + err = vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + ARRAY_SIZE(pin_idxs)); + if (err < 0) + return err; - case 0x29: /* Front Mic */ - idx = 2; - break; - } - err = via_new_analog_input(spec, labels[i], idx, 0x21); - if (err < 0) - return err; - imux->items[imux->num_items].label = labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } /* build volume/mute control of loopback */ err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); if (err < 0) return err; - /* for internal loopback recording select */ - imux->items[imux->num_items].label = "Stereo Mixer"; - imux->items[imux->num_items].index = 5; - imux->num_items++; - /* for digital mic select */ imux->items[imux->num_items].label = "Digital Mic"; imux->items[imux->num_items].index = 6; -- cgit v1.2.3 From 75e0eb24ee3ec3549c2e53707dcc87e5f7a2c791 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 12:56:55 +0200 Subject: ALSA: hda - Add inputs[] to auto_pin_cfg struct Added the new fields to contain all input-pins to struct auto_pin_cfg. Unlike the existing input_pins[], this array contains all input pins even if the multiple pins are assigned for a single role (i.e. two front mics). The former input_pins[] still remains for a while, but will be removed in near future. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 40 +++++++++++++++++++++++++++++----------- sound/pci/hda/hda_local.h | 12 +++++++++++- 2 files changed, 40 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3827092cc1d2..280a739c2a99 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4372,6 +4372,17 @@ static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, } +/* add the found input-pin to the cfg->inputs[] table */ +static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, + int type) +{ + if (cfg->num_inputs < AUTO_CFG_MAX_INS) { + cfg->inputs[cfg->num_inputs].pin = nid; + cfg->inputs[cfg->num_inputs].type = type; + cfg->num_inputs++; + } +} + /* * Parse all pin widgets and store the useful pin nids to cfg * @@ -4398,6 +4409,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, short sequences_line_out[ARRAY_SIZE(cfg->line_out_pins)]; short sequences_speaker[ARRAY_SIZE(cfg->speaker_pins)]; short sequences_hp[ARRAY_SIZE(cfg->hp_pins)]; + int i; memset(cfg, 0, sizeof(*cfg)); @@ -4482,19 +4494,26 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[preferred] = nid; else if (!cfg->input_pins[alt]) cfg->input_pins[alt] = nid; + add_auto_cfg_input_pin(cfg, nid, preferred); break; } - case AC_JACK_LINE_IN: + case AC_JACK_LINE_IN: { + int type; if (loc == AC_JACK_LOC_FRONT) - cfg->input_pins[AUTO_PIN_FRONT_LINE] = nid; + type = AUTO_PIN_FRONT_LINE; else - cfg->input_pins[AUTO_PIN_LINE] = nid; + type = AUTO_PIN_LINE; + cfg->input_pins[type] = nid; + add_auto_cfg_input_pin(cfg, nid, type); break; + } case AC_JACK_CD: cfg->input_pins[AUTO_PIN_CD] = nid; + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); break; case AC_JACK_AUX: cfg->input_pins[AUTO_PIN_AUX] = nid; + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); break; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: @@ -4621,14 +4640,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, if (cfg->dig_outs) snd_printd(" dig-out=0x%x/0x%x\n", cfg->dig_out_pins[0], cfg->dig_out_pins[1]); - snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," - " cd=0x%x, aux=0x%x\n", - cfg->input_pins[AUTO_PIN_MIC], - cfg->input_pins[AUTO_PIN_FRONT_MIC], - cfg->input_pins[AUTO_PIN_LINE], - cfg->input_pins[AUTO_PIN_FRONT_LINE], - cfg->input_pins[AUTO_PIN_CD], - cfg->input_pins[AUTO_PIN_AUX]); + snd_printd(" inputs:"); + for (i = 0; i < cfg->num_inputs; i++) { + snd_printdd(" %s=0x%x", + auto_pin_cfg_labels[cfg->inputs[i].type], + cfg->inputs[i].pin); + } + snd_printd("\n"); if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 28ab4aead48f..44c909445ba2 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -383,6 +383,14 @@ enum { extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST]; #define AUTO_CFG_MAX_OUTS 5 +#define AUTO_CFG_MAX_INS 8 + +struct auto_pin_cfg_item { + hda_nid_t pin; + int type; +}; + +struct auto_pin_cfg; struct auto_pin_cfg { int line_outs; @@ -393,7 +401,9 @@ struct auto_pin_cfg { int hp_outs; int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; - hda_nid_t input_pins[AUTO_PIN_LAST]; + hda_nid_t input_pins[AUTO_PIN_LAST]; /* old config; to be deprecated */ + int num_inputs; + struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; int dig_outs; hda_nid_t dig_out_pins[2]; hda_nid_t dig_in_pin; -- cgit v1.2.3 From d7b1ae9d8851bd247590cf7ab53248a2dac0419f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:00:16 +0200 Subject: ALSA: hda - Add snd_hda_get_input_pin_label() helper function Added snd_hda_get_input_pin_label() helper function to return the string that can be used for control or capture-source ids. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 25 ++++++++++++++++++++++++- sound/pci/hda/hda_local.h | 2 ++ 2 files changed, 26 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 280a739c2a99..72334b7f60e5 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4654,12 +4654,35 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); -/* labels for input pins */ +/* labels for input pins - for obsoleted config stuff */ const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" }; EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); +static const char *input_labels[AUTO_PIN_LAST][4] = { + { "Mic", "Mic 2", "Mic 3", "Mic 4" }, + { "Front Mic", "Front Mic 2", "Front Mic 3", "Front Mic 4" }, + { "Line", "Line 2", "Line 3", "Line 4" }, + { "Front Line", "Front Line 2", "Front Line 3", "Front Line 4" }, + { "CD", "CD 2", "CD 3", "CD 4" }, + { "Aux", "Aux 2", "Aux 3", "Aux 4" }, +}; + +const char *snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, + int input) +{ + int type = cfg->inputs[input].type; + int idx; + + for (idx = 0; idx < 3 && --input >= 0; idx++) { + if (type != cfg->inputs[input].type) + break; + } + return input_labels[type][idx]; +} +EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_label); + #ifdef CONFIG_PM /* diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 44c909445ba2..fb561748adb8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -391,6 +391,8 @@ struct auto_pin_cfg_item { }; struct auto_pin_cfg; +const char *snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, + int input); struct auto_pin_cfg { int line_outs; -- cgit v1.2.3 From 9e042e71325eeda03636aedfde6f2d27d6332188 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:04:44 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for AD codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b697fd2a6f8b..3409d315f507 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2880,7 +2880,7 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, /* create input playback/capture controls for the given pin */ static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin, - const char *ctlname, int boost) + const char *ctlname, int ctlidx, int boost) { char name[32]; int err, idx; @@ -2913,16 +2913,23 @@ static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, const struct auto_pin_cfg *cfg) { struct hda_input_mux *imux = &spec->private_imux; - int i, err; + int i, err, type, type_idx = 0; - for (i = 0; i < AUTO_PIN_LAST; i++) { - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - i <= AUTO_PIN_FRONT_MIC); + for (i = 0; i < cfg->num_inputs; i++) { + type = cfg->inputs[i].type; + if (i > 0 && type != cfg->inputs[i - 1].type) + type_idx++; + else + type_idx = 0; + err = new_analog_input(spec, cfg->inputs[i].pin, + auto_pin_cfg_labels[type], type_idx, + type <= AUTO_PIN_FRONT_MIC); if (err < 0) return err; - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = ad1988_pin_to_adc_idx(cfg->input_pins[i]); + imux->items[imux->num_items].label = + snd_hda_get_input_pin_label(cfg, i); + imux->items[imux->num_items].index = + ad1988_pin_to_adc_idx(cfg->inputs[i].pin); imux->num_items++; } imux->items[imux->num_items].label = "Mix"; @@ -2994,12 +3001,11 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) static void ad1988_auto_init_analog_input(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; int i, idx; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (! nid) - continue; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; switch (nid) { case 0x15: /* port-C */ snd_hda_codec_write(codec, 0x33, 0, AC_VERB_SET_CONNECT_SEL, 0x0); -- cgit v1.2.3 From fa4968a8b231816d161583e604a9f972e5713f17 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:05:08 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for CA-IBG codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index af478019088e..42b3fb4cafc4 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -468,13 +468,14 @@ static void parse_input(struct hda_codec *codec) spec->dig_in = nid; continue; } - for (j = 0; j < AUTO_PIN_LAST; j++) - if (cfg->input_pins[j] == pin) + for (j = 0; j < cfg->num_inputs; j++) + if (cfg->inputs[j].pin == pin) break; - if (j >= AUTO_PIN_LAST) + if (j >= cfg->num_inputs) continue; spec->input_pins[n] = pin; - spec->input_labels[n] = auto_pin_cfg_labels[j]; + spec->input_labels[n] = + auto_pin_cfg_labels[cfg->inputs[j].type]; spec->adcs[n] = nid; n++; } -- cgit v1.2.3 From c1e0bb92174dd16ffba5be0e4e5fbd366f61ff7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:05:30 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for CirrusLogic codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 38 +++++++++++++++++--------------------- 1 file changed, 17 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4ef5efaaaef1..ee1aea7296eb 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -329,7 +329,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t pin = cfg->input_pins[idx]; + hda_nid_t pin = cfg->inputs[idx].pin; unsigned int val = snd_hda_query_pin_caps(codec, pin); if (!(val & AC_PINCAP_PRES_DETECT)) return 0; @@ -424,10 +424,8 @@ static int parse_input(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t pin = cfg->input_pins[i]; - if (!pin) - continue; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; spec->input_idx[spec->num_inputs] = i; spec->capsrc_idx[i] = spec->num_inputs++; spec->cur_input = i; @@ -438,16 +436,17 @@ static int parse_input(struct hda_codec *codec) /* check whether the automatic mic switch is available */ if (spec->num_inputs == 2 && - spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) { - if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) { - if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) { + cfg->inputs[0].type <= AUTO_PIN_FRONT_MIC && + cfg->inputs[1].type == AUTO_PIN_FRONT_MIC) { + if (is_ext_mic(codec, cfg->inputs[0].pin)) { + if (!is_ext_mic(codec, cfg->inputs[1].pin)) { spec->mic_detect = 1; - spec->automic_idx = AUTO_PIN_FRONT_MIC; + spec->automic_idx = 0; } } else { - if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) { + if (is_ext_mic(codec, cfg->inputs[1].pin)) { spec->mic_detect = 1; - spec->automic_idx = AUTO_PIN_MIC; + spec->automic_idx = 1; } } } @@ -853,15 +852,12 @@ static void cs_automic(struct hda_codec *codec) hda_nid_t nid; unsigned int present; - nid = cfg->input_pins[spec->automic_idx]; + nid = cfg->inputs[spec->automic_idx].pin; present = snd_hda_jack_detect(codec, nid); if (present) change_cur_input(codec, spec->automic_idx, 0); - else { - unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ? - AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC; - change_cur_input(codec, imic, 0); - } + else + change_cur_input(codec, !spec->automic_idx, 0); } /* @@ -918,14 +914,14 @@ static void init_input(struct hda_codec *codec) unsigned int coef; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { + for (i = 0; i < cfg->num_inputs; i++) { unsigned int ctl; - hda_nid_t pin = cfg->input_pins[i]; - if (!pin || !spec->adc_nid[i]) + hda_nid_t pin = cfg->inputs[i].pin; + if (!spec->adc_nid[i]) continue; /* set appropriate pin control and mute first */ ctl = PIN_IN; - if (i <= AUTO_PIN_FRONT_MIC) { + if (cfg->inputs[i].type <= AUTO_PIN_FRONT_MIC) { unsigned int caps = snd_hda_query_pin_caps(codec, pin); caps >>= AC_PINCAP_VREF_SHIFT; if (caps & AC_PINCAP_VREF_80) -- cgit v1.2.3 From 66ceeb6bc2809bef0cfa18b1e22ddad5fc9b58b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:05:52 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for Realtek codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 208 ++++++++++++++++++++++-------------------- 1 file changed, 110 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50e0c82fd994..3e0f4816aed7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1265,16 +1265,14 @@ static void alc_init_auto_mic(struct hda_codec *codec) int i; /* there must be only two mic inputs exclusively */ - for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) - if (cfg->input_pins[i]) + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].type >= AUTO_PIN_LINE) return; fixed = ext = 0; - for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) { - hda_nid_t nid = cfg->input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; unsigned int defcfg; - if (!nid) - return; defcfg = snd_hda_codec_get_pincfg(codec, nid); switch (get_defcfg_connect(defcfg)) { case AC_JACK_PORT_FIXED: @@ -4719,7 +4717,7 @@ static struct snd_kcontrol_new alc880_control_templates[] = { /* add dynamic controls */ static int add_control(struct alc_spec *spec, int type, const char *name, - unsigned long val) + int cidx, unsigned long val) { struct snd_kcontrol_new *knew; @@ -4731,6 +4729,7 @@ static int add_control(struct alc_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; + knew->index = cidx; if (get_amp_nid_(val)) knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; @@ -4739,17 +4738,21 @@ static int add_control(struct alc_spec *spec, int type, const char *name, static int add_control_with_pfx(struct alc_spec *spec, int type, const char *pfx, const char *dir, - const char *sfx, unsigned long val) + const char *sfx, int cidx, unsigned long val) { char name[32]; snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); - return add_control(spec, type, name, val); + return add_control(spec, type, name, cidx, val); } -#define add_pb_vol_ctrl(spec, type, pfx, val) \ - add_control_with_pfx(spec, type, pfx, "Playback", "Volume", val) -#define add_pb_sw_ctrl(spec, type, pfx, val) \ - add_control_with_pfx(spec, type, pfx, "Playback", "Switch", val) +#define add_pb_vol_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Volume", 0, val) +#define add_pb_sw_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Switch", 0, val) +#define __add_pb_vol_ctrl(spec, type, pfx, cidx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Volume", cidx, val) +#define __add_pb_sw_ctrl(spec, type, pfx, cidx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Switch", cidx, val) #define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17) #define alc880_fixed_pin_idx(nid) ((nid) - 0x14) @@ -4902,16 +4905,16 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, /* create input playback/capture controls for the given pin */ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, - const char *ctlname, + const char *ctlname, int ctlidx, int idx, hda_nid_t mix_nid) { int err; - err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, + err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, ctlidx, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; - err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, + err = __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, ctlidx, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -4932,21 +4935,26 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx; + int i, err, idx, type, type_idx = 0; - for (i = 0; i < AUTO_PIN_LAST; i++) { + for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t pin; - pin = cfg->input_pins[i]; + pin = cfg->inputs[i].pin; if (!alc_is_input_pin(codec, pin)) continue; + type = cfg->inputs[i].type; + if (i > 0 && type == cfg->inputs[i - 1].type) + type_idx++; + else + type_idx = 0; if (mixer) { idx = get_connection_index(codec, mixer, pin); if (idx >= 0) { err = new_analog_input(spec, pin, - auto_pin_cfg_labels[i], - idx, mixer); + auto_pin_cfg_labels[type], + type_idx, idx, mixer); if (err < 0) return err; } @@ -4959,7 +4967,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, idx = get_connection_index(codec, cap2, pin); if (idx >= 0) { imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; + snd_hda_get_input_pin_label(cfg, i); imux->items[imux->num_items].index = idx; imux->num_items++; } @@ -5034,10 +5042,11 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) static void alc880_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC880_PIN_CD_NID && @@ -5204,19 +5213,13 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) static void fixup_single_adc(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin = 0; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; /* search for the input pin; there must be only one */ - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (spec->autocfg.input_pins[i]) { - pin = spec->autocfg.input_pins[i]; - break; - } - } - if (!pin) + if (cfg->num_inputs != 1) return; - i = init_capsrc_for_pin(codec, pin); + i = init_capsrc_for_pin(codec, cfg->inputs[0].pin); if (i >= 0) { /* use only this ADC */ if (spec->capsrc_nids) @@ -5269,6 +5272,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, int num_nids) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int n; hda_nid_t fallback_adc = 0, fallback_cap = 0; @@ -5294,10 +5298,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, fallback_adc = adc; fallback_cap = cap; } - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (!nid) - continue; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; for (j = 0; j < nconns; j++) { if (conn[j] == nid) break; @@ -5305,7 +5307,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, if (j >= nconns) break; } - if (i >= AUTO_PIN_LAST) { + if (i >= cfg->num_inputs) { int num_adcs = spec->num_adc_nids; spec->private_adc_nids[num_adcs] = adc; spec->private_capsrc_nids[num_adcs] = cap; @@ -6672,10 +6674,11 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) static void alc260_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (nid >= 0x12) { alc_set_input_pin(codec, nid, i); if (nid != ALC260_PIN_CD_NID && @@ -10538,12 +10541,11 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) static void alc882_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (!nid) - continue; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; alc_set_input_pin(codec, nid, i); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, @@ -10606,24 +10608,23 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) static int alc_auto_add_mic_boost(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, err; hda_nid_t nid; - nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Mic Boost", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; - } - nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Front Mic Boost", + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type > AUTO_PIN_FRONT_MIC) + break; + nid = cfg->inputs[i].pin; + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { + char label[32]; + snprintf(label, sizeof(label), "%s Boost", + snd_hda_get_input_pin_label(cfg, i)); + err = add_control(spec, ALC_CTL_WIDGET_VOL, label, 0, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; + if (err < 0) + return err; + } } return 0; } @@ -15577,10 +15578,11 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) static void alc861_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (nid >= 0x0c && nid <= 0x11) alc_set_input_pin(codec, nid, i); } @@ -16569,10 +16571,11 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) static void alc861vd_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC861VD_PIN_CD_NID && @@ -18805,10 +18808,11 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) static void alc662_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID && @@ -19037,6 +19041,39 @@ static hda_nid_t alc680_adc_nids[3] = { /* * Analog capture ADC cgange */ +static void alc680_rec_autoswitch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int pin_found = 0; + int type_found = AUTO_PIN_LAST; + hda_nid_t nid; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + nid = cfg->inputs[i].pin; + if (!(snd_hda_query_pin_caps(codec, nid) & + AC_PINCAP_PRES_DETECT)) + continue; + if (snd_hda_jack_detect(codec, nid)) { + if (cfg->inputs[i].type < type_found) { + type_found = cfg->inputs[i].type; + pin_found = nid; + } + } + } + + nid = 0x07; + if (pin_found) + snd_hda_get_connections(codec, pin_found, &nid, 1); + + if (nid != spec->cur_adc) + __snd_hda_codec_cleanup_stream(codec, spec->cur_adc, 1); + spec->cur_adc = nid; + snd_hda_codec_setup_stream(codec, nid, spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); +} + static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -19044,24 +19081,12 @@ static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int pre_mic, pre_line; - - pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); - pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]); + spec->cur_adc = 0x07; spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; - if (pre_mic || pre_line) { - if (pre_mic) - snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0, - format); - else - snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0, - format); - } else - snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format); + alc680_rec_autoswitch(codec); return 0; } @@ -19147,6 +19172,7 @@ static struct hda_verb alc680_init_verbs[] = { {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } }; @@ -19159,25 +19185,11 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x16; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x15; - spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18; - spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19; -} - -static void alc680_rec_autoswitch(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int present; - hda_nid_t new_adc; - - present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); - - new_adc = present ? 0x8 : 0x7; - __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1); - snd_hda_codec_setup_stream(codec, new_adc, - spec->cur_adc_stream_tag, 0, - spec->cur_adc_format); - + spec->autocfg.num_inputs = 2; + spec->autocfg.inputs[0].pin = 0x18; + spec->autocfg.inputs[0].type = AUTO_PIN_MIC; + spec->autocfg.inputs[1].pin = 0x19; + spec->autocfg.inputs[1].type = AUTO_PIN_LINE; } static void alc680_unsol_event(struct hda_codec *codec, -- cgit v1.2.3 From eea7dc932bfa802ad0377755ea821f416f4f8623 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:06:15 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for STAC/IDT codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 183 +++++++++++++++++++++-------------------- 1 file changed, 96 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 95148e58026c..d226edd1e143 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1180,14 +1180,11 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (err < 0) return err; } - for (i = 0; i < AUTO_PIN_LAST; i++) { - nid = cfg->input_pins[i]; - if (nid) { - err = stac92xx_add_jack(codec, nid, - SND_JACK_MICROPHONE); - if (err < 0) - return err; - } + for (i = 0; i < cfg->num_inputs; i++) { + nid = cfg->inputs[i].pin; + err = stac92xx_add_jack(codec, nid, SND_JACK_MICROPHONE); + if (err < 0) + return err; } return 0; @@ -2821,41 +2818,55 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; unsigned int pincap; + int i; if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; - nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) - return nid; + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type == AUTO_PIN_LINE) { + nid = cfg->inputs[i].pin; + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_OUT) + return nid; + } + } return 0; } +static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid); + /* check whether the mic-input can be used as line-out */ -static hda_nid_t check_mic_out_switch(struct hda_codec *codec) +static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int def_conf, pincap; - unsigned int mic_pin; + int i, mic_type; + *dac = 0; if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; - mic_pin = AUTO_PIN_MIC; - for (;;) { - hda_nid_t nid = cfg->input_pins[mic_pin]; + mic_type = AUTO_PIN_MIC; + again: + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + if (cfg->inputs[i].type != mic_type) + continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_OUT) - return nid; + if (pincap & AC_PINCAP_OUT) { + *dac = get_unassigned_dac(codec, nid); + if (*dac) + return nid; + } } - if (mic_pin == AUTO_PIN_MIC) - mic_pin = AUTO_PIN_FRONT_MIC; - else - break; + } + if (mic_type == AUTO_PIN_MIC) { + mic_type = AUTO_PIN_FRONT_MIC; + goto again; } return 0; } @@ -3002,17 +3013,14 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } } /* add mic as output */ - nid = check_mic_out_switch(codec); - if (nid) { - dac = get_unassigned_dac(codec, nid); - if (dac) { - snd_printdd("STAC: Add mic-in 0x%x as output %d\n", - nid, cfg->line_outs); - cfg->line_out_pins[cfg->line_outs] = nid; - cfg->line_outs++; - spec->mic_switch = nid; - add_spec_dacs(spec, dac); - } + nid = check_mic_out_switch(codec, &dac); + if (nid && dac) { + snd_printdd("STAC: Add mic-in 0x%x as output %d\n", + nid, cfg->line_outs); + cfg->line_out_pins[cfg->line_outs] = nid; + cfg->line_outs++; + spec->mic_switch = nid; + add_spec_dacs(spec, dac); } snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", @@ -3202,13 +3210,13 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, return err; } - for (idx = AUTO_PIN_MIC; idx <= AUTO_PIN_FRONT_LINE; idx++) { - nid = cfg->input_pins[idx]; - if (nid) { - err = stac92xx_add_jack_mode_control(codec, nid, idx); - if (err < 0) - return err; - } + for (idx = 0; idx < cfg->num_inputs; idx++) { + if (cfg->inputs[idx].type > AUTO_PIN_FRONT_LINE) + break; + nid = cfg->inputs[idx].pin; + err = stac92xx_add_jack_mode_control(codec, nid, idx); + if (err < 0) + return err; } return 0; @@ -3415,7 +3423,7 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, /* create a volume assigned to the given pin (only if supported) */ /* return 1 if the volume control is created */ static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, - const char *label, int direction) + const char *label, int idx, int direction) { unsigned int caps, nums; char name[32]; @@ -3432,8 +3440,8 @@ static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, if (!nums) return 0; snprintf(name, sizeof(name), "%s Capture Volume", label); - err = stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction)); + err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction)); if (err < 0) return err; return 1; @@ -3485,11 +3493,11 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, else label = stac92xx_dmic_labels[dimux->num_items]; - err = create_elem_capture_vol(codec, nid, label, HDA_INPUT); + err = create_elem_capture_vol(codec, nid, label, 0, HDA_INPUT); if (err < 0) return err; if (!err) { - err = create_elem_capture_vol(codec, nid, label, + err = create_elem_capture_vol(codec, nid, label, 0, HDA_OUTPUT); if (err < 0) return err; @@ -3540,10 +3548,11 @@ static int set_mic_route(struct hda_codec *codec, int i; mic->pin = pin; - for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) - if (pin == cfg->input_pins[i]) + for (i = 0; i < cfg->num_inputs; i++) { + if (pin == cfg->inputs[i].pin) break; - if (i <= AUTO_PIN_FRONT_MIC) { + } + if (i < cfg->num_inputs && cfg->inputs[i].type <= AUTO_PIN_FRONT_MIC) { /* analog pin */ i = get_connection_index(codec, spec->mux_nids[0], pin); if (i < 0) @@ -3577,13 +3586,13 @@ static int stac_check_auto_mic(struct hda_codec *codec) hda_nid_t fixed, ext; int i; - for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) { - if (cfg->input_pins[i]) + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type >= AUTO_PIN_LINE) return 0; /* must be exclusively mics */ } fixed = ext = 0; - for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) - if (check_mic_pin(codec, cfg->input_pins[i], &fixed, &ext)) + for (i = 0; i < cfg->num_inputs; i++) + if (check_mic_pin(codec, cfg->inputs[i].pin, &fixed, &ext)) return 0; for (i = 0; i < spec->num_dmics; i++) if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext)) @@ -3603,14 +3612,12 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; - int i, j; + int i, j, type_idx = 0; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = cfg->input_pins[i]; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; int index, err; - if (!nid) - continue; index = -1; for (j = 0; j < spec->num_muxes; j++) { index = get_connection_index(codec, spec->mux_nids[j], @@ -3621,13 +3628,18 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const if (index < 0) continue; + if (i > 0 && cfg->inputs[i].type == cfg->inputs[i - 1].type) + type_idx++; + else + type_idx = 0; err = create_elem_capture_vol(codec, nid, - auto_pin_cfg_labels[i], + auto_pin_cfg_labels[i], type_idx, HDA_INPUT); if (err < 0) return err; - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].label = + snd_hda_get_input_pin_label(cfg, i); imux->items[imux->num_items].index = index; imux->num_items++; } @@ -4304,37 +4316,34 @@ static int stac92xx_init(struct hda_codec *codec) if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT)) stac_issue_unsol_event(codec, spec->ext_mic.pin); } - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = cfg->input_pins[i]; - if (nid) { - unsigned int pinctl, conf; - if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) { - /* for mic pins, force to initialize */ - pinctl = stac92xx_get_default_vref(codec, nid); + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + int type = cfg->inputs[i].type; + unsigned int pinctl, conf; + if (type == AUTO_PIN_MIC || type == AUTO_PIN_FRONT_MIC) { + /* for mic pins, force to initialize */ + pinctl = stac92xx_get_default_vref(codec, nid); + pinctl |= AC_PINCTL_IN_EN; + stac92xx_auto_set_pinctl(codec, nid, pinctl); + } else { + pinctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + /* if PINCTL already set then skip */ + /* Also, if both INPUT and OUTPUT are set, + * it must be a BIOS bug; need to override, too + */ + if (!(pinctl & AC_PINCTL_IN_EN) || + (pinctl & AC_PINCTL_OUT_EN)) { + pinctl &= ~AC_PINCTL_OUT_EN; pinctl |= AC_PINCTL_IN_EN; stac92xx_auto_set_pinctl(codec, nid, pinctl); - } else { - pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - /* if PINCTL already set then skip */ - /* Also, if both INPUT and OUTPUT are set, - * it must be a BIOS bug; need to override, too - */ - if (!(pinctl & AC_PINCTL_IN_EN) || - (pinctl & AC_PINCTL_OUT_EN)) { - pinctl &= ~AC_PINCTL_OUT_EN; - pinctl |= AC_PINCTL_IN_EN; - stac92xx_auto_set_pinctl(codec, nid, - pinctl); - } - } - conf = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { - if (enable_pin_detect(codec, nid, - STAC_INSERT_EVENT)) - stac_issue_unsol_event(codec, nid); } } + conf = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { + if (enable_pin_detect(codec, nid, STAC_INSERT_EVENT)) + stac_issue_unsol_event(codec, nid); + } } for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], -- cgit v1.2.3 From 7b315bb4980448250c80a7464c256b54d546cb26 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Aug 2010 13:06:30 +0200 Subject: ALSA: hda - Use new inputs[] field to parse input-pins for VIA codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 144 +++++++++++++++++++++++----------------------- 1 file changed, 73 insertions(+), 71 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 41861388f43a..93b86adbce63 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -444,8 +444,8 @@ static hda_nid_t vt1812_adc_nids[2] = { /* add dynamic controls */ -static int via_add_control(struct via_spec *spec, int type, const char *name, - unsigned long val) +static int __via_add_control(struct via_spec *spec, int type, const char *name, + int idx, unsigned long val) { struct snd_kcontrol_new *knew; @@ -463,6 +463,9 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, return 0; } +#define via_add_control(spec, type, name, val) \ + __via_add_control(spec, type, name, 0, val) + static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, struct snd_kcontrol_new *tmpl) { @@ -494,18 +497,18 @@ static void via_free_kctls(struct hda_codec *codec) /* create input playback/capture controls for the given pin */ static int via_new_analog_input(struct via_spec *spec, const char *ctlname, - int idx, int mix_nid) + int type_idx, int idx, int mix_nid) { char name[32]; int err; sprintf(name, "%s Playback Volume", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, + err = __via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, type_idx, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -557,14 +560,12 @@ static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); static void via_auto_init_analog_input(struct hda_codec *codec) { struct via_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int ctl; int i; - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (!nid) - continue; - + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(spec, nid)) ctl = PIN_OUT; else if (i <= AUTO_PIN_FRONT_MIC) @@ -1322,15 +1323,14 @@ static void mute_aa_path(struct hda_codec *codec, int mute) } static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) { - int res = 0; - int index; - for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { - if (pin == spec->autocfg.input_pins[index]) { - res = 1; - break; - } + const struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + if (pin == cfg->inputs[i].pin) + return cfg->inputs[i].type < AUTO_PIN_FRONT_LINE; } - return res; + return 0; } static int via_smart51_info(struct snd_kcontrol *kcontrol, @@ -1348,25 +1348,21 @@ static int via_smart51_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + const struct auto_pin_cfg *cfg = &spec->autocfg; int on = 1; int i; - for (i = 0; i < ARRAY_SIZE(index); i++) { - hda_nid_t nid = spec->autocfg.input_pins[index[i]]; - if (nid) { - int ctl = - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0); - if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode - && spec->codec_type != VT1718S) - continue; /* ignore FMic for independent HP */ - if (ctl & AC_PINCTL_IN_EN - && !(ctl & AC_PINCTL_OUT_EN)) - on = 0; - } + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + int ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (cfg->inputs[i].type >= AUTO_PIN_FRONT_LINE) + continue; + if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC && + spec->hp_independent_mode && spec->codec_type != VT1718S) + continue; /* ignore FMic for independent HP */ + if ((ctl & AC_PINCTL_IN_EN) && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; } *ucontrol->value.integer.value = on; return 0; @@ -1377,36 +1373,38 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; int out_in = *ucontrol->value.integer.value ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; - int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; int i; - for (i = 0; i < ARRAY_SIZE(index); i++) { - hda_nid_t nid = spec->autocfg.input_pins[index[i]]; - if (i == AUTO_PIN_FRONT_MIC - && spec->hp_independent_mode - && spec->codec_type != VT1718S) + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + unsigned int parm; + + if (cfg->inputs[i].type >= AUTO_PIN_FRONT_LINE) + continue; + if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC && + spec->hp_independent_mode && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ - if (nid) { - unsigned int parm = snd_hda_codec_read( - codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - parm |= out_in; - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - parm); - if (out_in == AC_PINCTL_OUT_EN) { - mute_aa_path(codec, 1); - notify_aa_path_ctls(codec); - } - if (spec->codec_type == VT1718S) - snd_hda_codec_amp_stereo( + + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + if (spec->codec_type == VT1718S) { + snd_hda_codec_amp_stereo( codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); } - if (i == AUTO_PIN_FRONT_MIC) { + if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC) { if (spec->codec_type == VT1708S || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ @@ -1442,7 +1440,7 @@ static struct snd_kcontrol_new via_smart51_mixer[2] = { static int via_smart51_build(struct via_spec *spec) { struct snd_kcontrol_new *knew; - int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + const struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; int i; @@ -1450,13 +1448,14 @@ static int via_smart51_build(struct via_spec *spec) if (knew == NULL) return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(index); i++) { - nid = spec->autocfg.input_pins[index[i]]; - if (nid) { + for (i = 0; i < cfg->num_inputs; i++) { + nid = cfg->inputs[i].pin; + if (cfg->inputs[i].type < AUTO_PIN_FRONT_LINE) { knew = via_clone_control(spec, &via_smart51_mixer[1]); if (knew == NULL) return -ENOMEM; knew->subdevice = nid; + break; } } @@ -2419,7 +2418,7 @@ static int vt_auto_create_analog_input_ctls(struct via_spec *spec, hda_nid_t pin_idxs[], int num_idxs) { struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx; + int i, err, idx, type, type_idx = 0; /* for internal loopback recording select */ for (idx = 0; idx < num_idxs; idx++) { @@ -2431,20 +2430,23 @@ static int vt_auto_create_analog_input_ctls(struct via_spec *spec, } } - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (!cfg->input_pins[i]) - continue; - + for (i = 0; i < cfg->num_inputs; i++) { + type = cfg->inputs[i].type; for (idx = 0; idx < num_idxs; idx++) - if (pin_idxs[idx] == cfg->input_pins[i]) + if (pin_idxs[idx] == cfg->inputs[i].pin) break; if (idx >= num_idxs) continue; - err = via_new_analog_input(spec, auto_pin_cfg_labels[i], - idx, cap_nid); + if (i > 0 && type == cfg->inputs[i - 1].type) + type_idx++; + else + type_idx = 0; + err = via_new_analog_input(spec, auto_pin_cfg_labels[type], + type_idx, idx, cap_nid); if (err < 0) return err; - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].label = + snd_hda_get_input_pin_label(cfg, i); imux->items[imux->num_items].index = idx; imux->num_items++; } @@ -5513,7 +5515,7 @@ static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, if (err < 0) return err; /* build volume/mute control of loopback */ - err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + err = via_new_analog_input(spec, "Stereo Mixer", 0, 3, 0x21); if (err < 0) return err; @@ -5836,7 +5838,7 @@ static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, return err; /* build volume/mute control of loopback */ - err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + err = via_new_analog_input(spec, "Stereo Mixer", 0, 5, 0x21); if (err < 0) return err; -- cgit v1.2.3 From 048e78a5bc22c27410cb5ca9680c3c7ac400607f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 2 Sep 2010 08:35:47 +0200 Subject: ALSA: hda - Add a new hp-laptop model for Conexant 5066, tested on HP G60 This new model adds the following functionality to HP G60: - Automute of internal speakers - Autoswitch of internal/external mics - Remove SPDIF not physically present BugLink: http://launchpad.net/bugs/587388 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ce46fa1e643e..37c6aad5e590 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5051 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5cdb80edbd7f..4f0619908a30 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2407,6 +2420,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + /* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { @@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3031,6 +3068,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3052,6 +3091,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), @@ -3116,6 +3156,23 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- cgit v1.2.3 From 65f04443c96dbda11b8fff21d6390e082846aa3c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Sep 2010 12:58:25 +0200 Subject: ALSA: usb-audio: fix Fast Track Ultra (8R) 44.1 sample rates The M-Audio Fast Track Ultra series devices did not play sound correctly at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive fixes this. Signed-off-by: Felix Homann Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 2 -- sound/usb/quirks-table.h | 90 +++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 85 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1a701f1e8f50..bb9f938558fd 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -404,8 +404,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 2e8003f98fca..4818fbdc02fb 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1830,7 +1830,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE(0x0763, 0x2080), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "M-Audio", */ - /* .product_name = "Fast Track Ultra 8", */ + /* .product_name = "Fast Track Ultra", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = & (const struct snd_usb_audio_quirk[]) { @@ -1840,11 +1840,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { @@ -1867,11 +1907,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x01, + .ep_attr = 0x09, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, { .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x81, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, + .rate_table = (unsigned int[]) { + 44100, 48000, 88200, 96000 + } + } }, /* interface 3 (MIDI) is standard compliant */ { -- cgit v1.2.3 From 7b6717e144de6592e614fd7fc3b914b6bf686a9d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 2 Sep 2010 17:13:15 +0800 Subject: ALSA: usb-audio: Assume first control interface is for audio For devices with more than one control interface, let's assume the first one contains the audio controls. Unfortunately, there is no field in any of the descriptors to tell us whether a control interface is for audio or MIDI controls, so a better check is not easy to implement. On a composite device with audio and MIDI functions, for example, the code currently overwrites chip->ctrl_intf, causing operations on the control interface to fail if they are issued after the device probe. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 9feb00c831a0..b443a33d31c9 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -465,7 +465,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ -- cgit v1.2.3 From aa70201fdc374f245cfa1874e11df145ace3ffaf Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 30 Aug 2010 16:32:43 +0200 Subject: ALSA: usb-audio: add Edirol PCR-1 PCM support Add a quirk for the other logical device of the PCR-1 so that not only the MIDI interface but also the audio interface works. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 27 +++++++++++++++++++++++++-- 1 file changed, 25 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4818fbdc02fb..838b81b74fd7 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1135,12 +1135,35 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_STANDARD_INTERFACE } }, +{ + /* has ID 0x0066 when not in "Advanced Driver" mode */ + USB_DEVICE(0x0582, 0x0064), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "EDIROL", */ + /* .product_name = "PCR-1", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, { /* has ID 0x0067 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0065), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "PCR-1", + /* .vendor_name = "EDIROL", */ + /* .product_name = "PCR-1", */ .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { -- cgit v1.2.3 From 9d0c91938e3a42c683e4e55d108b928e89a3bbc4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 30 Aug 2010 16:42:17 +0200 Subject: ALSA: usb-audio: add Roland A-PRO support Add a quirk for the Roland/Cakewalk A-300PRO/A-500PRO/A-800PRO keyboards. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 838b81b74fd7..92f099a804ca 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1548,6 +1548,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* has ID 0x0110 when not in Advanced Driver mode */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x010f), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "A-PRO", */ + .ifnum = 1, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0007 + } + } +}, /* Guillemot devices */ { -- cgit v1.2.3 From 7b28079b3284ccb15ad4f003fb7073890600d0c1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 30 Aug 2010 16:45:38 +0200 Subject: ALSA: usb-audio: add BOSS ME-25 support Add a quirk to make the BOSS ME-25 work. Many thanks to Kees van Veen. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 92f099a804ca..c86c613e0b96 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1562,6 +1562,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x0113), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "BOSS", */ + /* .product_name = "ME-25", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { -- cgit v1.2.3 From a2acad8298a42b7be684a32fafaf83332bba9c2b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Sep 2010 10:53:11 +0200 Subject: ALSA: usb-audio: fix detection of vendor-specific device protocol settings The Audio Class v2 support code in 2.6.35 added checks for the bInterfaceProtocol field. However, there are devices (usually those detected by vendor-specific quirks) that do not have one of the predefined values in this field, which made the driver reject them. To fix this regression, restore the old behaviour, i.e., assume that a device with an unknown bInterfaceProtocol field (other than UAC_VERSION_2) has more or less UAC-v1-compatible descriptors. [compile warning fixes by tiwai] Signed-off-by: Clemens Ladisch Cc: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 9 +++++---- sound/usb/clock.c | 3 +-- sound/usb/endpoint.c | 11 ++++++----- sound/usb/format.c | 22 ++++++++++------------ sound/usb/mixer.c | 10 +++++++++- sound/usb/pcm.c | 3 +-- 6 files changed, 32 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index b443a33d31c9..32e4be8a187c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } switch (protocol) { + default: + snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n", + protocol); + /* fall through */ + case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1 = control_header; @@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; } return 0; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b853f8df794f..7754a1034545 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: return set_sample_rate_v2(chip, iface, alts, fmt, rate); } - - return -EINVAL; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1a701f1e8f50..ef0a07e34844 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + case UAC_VERSION_1: { struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno, as->bTerminalLink); continue; } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; } /* get format type */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 3a1375459c06..69148212aa70 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, u64 pcm_formats; switch (protocol) { - case UAC_VERSION_1: { + case UAC_VERSION_1: + default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; @@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } - - default: - return -EINVAL; } pcm_formats = 0; @@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * audio class v2 uses class specific EP0 range requests for that. */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, /* fp->channels is already set in this case */ ret = parse_audio_format_rates_v2(chip, fp); break; - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } if (fp->channels < 1) { @@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, ret = parse_audio_format_rates_v2(chip, fp); break; } - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } return ret; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c166db0057d3..3ed3901369ce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol; + switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + case UAC_VERSION_1: + default: + mixer->protocol = UAC_VERSION_1; + break; + case UAC_VERSION_2: + mixer->protocol = UAC_VERSION_2; + break; + } if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3634cedf9306..3b5135c93062 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return init_pitch_v1(chip, iface, alts, fmt); case UAC_VERSION_2: return init_pitch_v2(chip, iface, alts, fmt); } - - return -EINVAL; } /* -- cgit v1.2.3 From 9fe856e47e1751204faf3d604c6d20ab24bd3b93 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 4 Sep 2010 18:52:54 -0700 Subject: sound: Remove unnecessary casts of private_data Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/core/oss/mixer_oss.c | 22 +++++++++++----------- sound/core/pcm.c | 3 +-- sound/drivers/virmidi.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 2 +- sound/isa/ad1816a/ad1816a.c | 2 +- sound/isa/azt2320.c | 2 +- sound/isa/gus/gusmax.c | 4 ++-- sound/isa/sb/sb8.c | 2 +- sound/oss/au1550_ac97.c | 18 +++++++++--------- sound/pci/emu10k1/emumpu401.c | 2 +- sound/pci/ice1712/pontis.c | 6 +++--- sound/pci/ice1712/prodigy192.c | 2 +- sound/pci/rme96.c | 8 ++++---- sound/pci/rme9652/hdsp.c | 8 ++++---- 14 files changed, 41 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf20df96..86afb13cd240 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -77,7 +77,7 @@ static int snd_mixer_oss_release(struct inode *inode, struct file *file) struct snd_mixer_oss_file *fmixer; if (file->private_data) { - fmixer = (struct snd_mixer_oss_file *) file->private_data; + fmixer = file->private_data; module_put(fmixer->card->module); snd_card_file_remove(fmixer->card, file); kfree(fmixer); @@ -368,7 +368,7 @@ static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int static long snd_mixer_oss_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - return snd_mixer_oss_ioctl1((struct snd_mixer_oss_file *) file->private_data, cmd, arg); + return snd_mixer_oss_ioctl1(file->private_data, cmd, arg); } int snd_mixer_oss_ioctl_card(struct snd_card *card, unsigned int cmd, unsigned long arg) @@ -582,7 +582,7 @@ static int snd_mixer_oss_get_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *left, int *right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; *left = *right = 100; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { @@ -691,7 +691,7 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int left, int right) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; if (slot->present & SNDRV_MIXER_OSS_PRESENT_PVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); @@ -740,7 +740,7 @@ static int snd_mixer_oss_get_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -753,7 +753,7 @@ static int snd_mixer_oss_get_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int *active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; int left, right; left = right = 1; @@ -766,7 +766,7 @@ static int snd_mixer_oss_put_recsrc1_sw(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], active, active, 0); return 0; @@ -776,7 +776,7 @@ static int snd_mixer_oss_put_recsrc1_route(struct snd_mixer_oss_file *fmixer, struct snd_mixer_oss_slot *pslot, int active) { - struct slot *slot = (struct slot *)pslot->private_data; + struct slot *slot = pslot->private_data; snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], active, active, 1); return 0; @@ -813,7 +813,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -861,7 +861,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned if (!(mixer->mask_recsrc & (1 << idx))) continue; pslot = &mixer->slots[idx]; - slot = (struct slot *)pslot->private_data; + slot = pslot->private_data; if (slot->signature != SNDRV_MIXER_OSS_SIGNATURE) continue; if (!(slot->present & SNDRV_MIXER_OSS_PRESENT_CAPTURE)) @@ -925,7 +925,7 @@ static int snd_mixer_oss_build_test(struct snd_mixer_oss *mixer, struct slot *sl static void snd_mixer_oss_slot_free(struct snd_mixer_oss_slot *chn) { - struct slot *p = (struct slot *)chn->private_data; + struct slot *p = chn->private_data; if (p) { if (p->allocated && p->assigned) { kfree(p->assigned->name); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 204af48c5cc1..88525a958291 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -364,8 +364,7 @@ static void snd_pcm_stream_proc_info_read(struct snd_info_entry *entry, static void snd_pcm_substream_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - snd_pcm_proc_info_read((struct snd_pcm_substream *)entry->private_data, - buffer); + snd_pcm_proc_info_read(entry->private_data, buffer); } static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 0e631c3221e3..f4cd49336f33 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -94,7 +94,7 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) sizeof(struct snd_card_virmidi), &card); if (err < 0) return err; - vmidi = (struct snd_card_virmidi *)card->private_data; + vmidi = card->private_data; vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 1adb8a3c2b62..ebab6c7aaa81 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -878,7 +878,7 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) static void proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_akm4xxx *ak = (struct snd_akm4xxx *)entry->private_data; + struct snd_akm4xxx *ak = entry->private_data; int reg, val, chip; for (chip = 0; chip < ak->num_chips; chip++) { for (reg = 0; reg < ak->total_regs; reg++) { diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index bbcbf92a8ebe..3cb75bc97699 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -162,7 +162,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard sizeof(struct snd_card_ad1816a), &card); if (error < 0) return error; - acard = (struct snd_card_ad1816a *)card->private_data; + acard = card->private_data; if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { snd_card_free(card); diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index f7aa637b0d18..aac8dc15c2fe 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -188,7 +188,7 @@ static int __devinit snd_card_azt2320_probe(int dev, sizeof(struct snd_card_azt2320), &card); if (error < 0) return error; - acard = (struct snd_card_azt2320 *)card->private_data; + acard = card->private_data; if ((error = snd_card_azt2320_pnp(dev, acard, pcard, pid))) { snd_card_free(card); diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index f26eac8d8110..3e4a58b72913 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -191,7 +191,7 @@ static int __devinit snd_gusmax_mixer(struct snd_wss *chip) static void snd_gusmax_free(struct snd_card *card) { - struct snd_gusmax *maxcard = (struct snd_gusmax *)card->private_data; + struct snd_gusmax *maxcard = card->private_data; if (maxcard == NULL) return; @@ -219,7 +219,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) if (err < 0) return err; card->private_free = snd_gusmax_free; - maxcard = (struct snd_gusmax *)card->private_data; + maxcard = card->private_data; maxcard->card = card; maxcard->irq = -1; diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 81284a8fa0ce..2259e3f726a7 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -72,7 +72,7 @@ static irqreturn_t snd_sb8_interrupt(int irq, void *dev_id) static void snd_sb8_free(struct snd_card *card) { - struct snd_sb8 *acard = (struct snd_sb8 *)card->private_data; + struct snd_sb8 *acard = card->private_data; if (acard == NULL) return; diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index c6f2621221ba..8a12621d8b3a 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -171,7 +171,7 @@ au1550_delay(int msec) static u16 rdcodec(struct ac97_codec *codec, u8 addr) { - struct au1550_state *s = (struct au1550_state *)codec->private_data; + struct au1550_state *s = codec->private_data; unsigned long flags; u32 cmd, val; u16 data; @@ -239,7 +239,7 @@ rdcodec(struct ac97_codec *codec, u8 addr) static void wrcodec(struct ac97_codec *codec, u8 addr, u16 data) { - struct au1550_state *s = (struct au1550_state *)codec->private_data; + struct au1550_state *s = codec->private_data; unsigned long flags; u32 cmd, val; int i; @@ -820,7 +820,7 @@ mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, static long au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct ac97_codec *codec = s->codec; int ret; @@ -1031,7 +1031,7 @@ copy_dmabuf_user(struct dmabuf *db, char* userbuf, int count, int to_user) static ssize_t au1550_read(struct file *file, char *buffer, size_t count, loff_t *ppos) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db = &s->dma_adc; DECLARE_WAITQUEUE(wait, current); ssize_t ret; @@ -1111,7 +1111,7 @@ out2: static ssize_t au1550_write(struct file *file, const char *buffer, size_t count, loff_t * ppos) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db = &s->dma_dac; DECLARE_WAITQUEUE(wait, current); ssize_t ret = 0; @@ -1211,7 +1211,7 @@ out2: static unsigned int au1550_poll(struct file *file, struct poll_table_struct *wait) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; unsigned long flags; unsigned int mask = 0; @@ -1250,7 +1250,7 @@ au1550_poll(struct file *file, struct poll_table_struct *wait) static int au1550_mmap(struct file *file, struct vm_area_struct *vma) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; struct dmabuf *db; unsigned long size; int ret = 0; @@ -1342,7 +1342,7 @@ dma_count_done(struct dmabuf *db) static int au1550_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; unsigned long flags; audio_buf_info abinfo; count_info cinfo; @@ -1868,7 +1868,7 @@ out2: static int au1550_release(struct inode *inode, struct file *file) { - struct au1550_state *s = (struct au1550_state *)file->private_data; + struct au1550_state *s = file->private_data; lock_kernel(); diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c index 8578c70c61f2..bab564824efe 100644 --- a/sound/pci/emu10k1/emumpu401.c +++ b/sound/pci/emu10k1/emumpu401.c @@ -321,7 +321,7 @@ static struct snd_rawmidi_ops snd_emu10k1_midi_input = static void snd_emu10k1_midi_free(struct snd_rawmidi *rmidi) { - struct snd_emu10k1_midi *midi = (struct snd_emu10k1_midi *)rmidi->private_data; + struct snd_emu10k1_midi *midi = rmidi->private_data; midi->interrupt = NULL; midi->rmidi = NULL; } diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 6bc3f91b7281..cdb873f5da50 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -638,7 +638,7 @@ static struct snd_kcontrol_new pontis_controls[] __devinitdata = { */ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; char line[64]; unsigned int reg, val; mutex_lock(&ice->gpio_mutex); @@ -653,7 +653,7 @@ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buf static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; mutex_lock(&ice->gpio_mutex); @@ -676,7 +676,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; mutex_lock(&ice->gpio_mutex); diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 2a8e5cd8f2d8..e36ddb94c382 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -654,7 +654,7 @@ static int prodigy192_ak4114_init(struct snd_ice1712 *ice) static void stac9460_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; + struct snd_ice1712 *ice = entry->private_data; int reg, val; /* registers 0x0 - 0x14 */ for (reg = 0; reg <= 0x15; reg++) { diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index d19dc052c391..d5f5b440fc40 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -1527,14 +1527,14 @@ snd_rme96_free(void *private_data) static void snd_rme96_free_spdif_pcm(struct snd_pcm *pcm) { - struct rme96 *rme96 = (struct rme96 *) pcm->private_data; + struct rme96 *rme96 = pcm->private_data; rme96->spdif_pcm = NULL; } static void snd_rme96_free_adat_pcm(struct snd_pcm *pcm) { - struct rme96 *rme96 = (struct rme96 *) pcm->private_data; + struct rme96 *rme96 = pcm->private_data; rme96->adat_pcm = NULL; } @@ -1661,7 +1661,7 @@ static void snd_rme96_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { int n; - struct rme96 *rme96 = (struct rme96 *)entry->private_data; + struct rme96 *rme96 = entry->private_data; rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); @@ -2348,7 +2348,7 @@ snd_rme96_probe(struct pci_dev *pci, if (err < 0) return err; card->private_free = snd_rme96_card_free; - rme96 = (struct rme96 *)card->private_data; + rme96 = card->private_data; rme96->card = card; rme96->pci = pci; snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b92adef8e81e..599e09051663 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3284,7 +3284,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) static void snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct hdsp *hdsp = (struct hdsp *) entry->private_data; + struct hdsp *hdsp = entry->private_data; unsigned int status; unsigned int status2; char *pref_sync_ref; @@ -4566,7 +4566,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -5155,7 +5155,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) static void snd_hdsp_card_free(struct snd_card *card) { - struct hdsp *hdsp = (struct hdsp *) card->private_data; + struct hdsp *hdsp = card->private_data; if (hdsp) snd_hdsp_free(hdsp); @@ -5181,7 +5181,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, if (err < 0) return err; - hdsp = (struct hdsp *) card->private_data; + hdsp = card->private_data; card->private_free = snd_hdsp_card_free; hdsp->dev = dev; hdsp->pci = pci; -- cgit v1.2.3 From add7c0a6a4b8669ebd726f9c08ba6002900ca671 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Jul 2009 18:19:12 +0200 Subject: ALSA: ca0106 - clean up playback pointer callback Clean up the playback pointer callback function a bit, and make the pointer check more strictly to avoid bogus pointers. Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 34 ++++++++++++++++------------------ 1 file changed, 16 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 0a3d3d6e77b4..8e69620da20b 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1002,29 +1002,27 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_ca0106_pcm *epcm = runtime->private_data; - snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0; + unsigned int ptr, prev_ptr; int channel = epcm->channel_id; + int timeout = 10; if (!epcm->running) return 0; - ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); - ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); - ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); - if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); - ptr2 = bytes_to_frames(runtime, ptr1); - ptr2+= (ptr4 >> 3) * runtime->period_size; - ptr=ptr2; - if (ptr >= runtime->buffer_size) - ptr -= runtime->buffer_size; - /* - printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " - "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", - ptr1, ptr2, ptr, (int)runtime->buffer_size, - (int)runtime->period_size, (int)runtime->frame_bits, - (int)runtime->rate); - */ - return ptr; + prev_ptr = -1; + do { + ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + ptr = (ptr >> 3) * runtime->period_size; + ptr += bytes_to_frames(runtime, + snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel)); + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + if (prev_ptr == ptr) + return ptr; + prev_ptr = ptr; + } while (--timeout); + snd_printk(KERN_WARNING "ca0106: unstable DMA pointer!\n"); + return 0; } /* pointer_capture callback */ -- cgit v1.2.3 From 4d155641c81203440da64c4633b4efaab75f63b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Sep 2010 11:58:30 +0200 Subject: ALSA: hda - Add quirk for Lenovo T400s Lenovo T400s requires the quirk to make automatic HP/mic switching working. Reported-by: Frank Becker Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f0619908a30..71f9d6475b09 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3094,6 +3094,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), -- cgit v1.2.3 From 4c25b93223340deff73381cc47f9244fb379a74d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:37:10 +0200 Subject: ALSA: virtuoso: work around missing reset in the Xonar DS Windows driver For the WM8776 chip, this driver uses a different sample format and more features than the Windows driver. When rebooting from Linux into Windows, the latter driver does not reset the chip but assumes all its registers have their default settings, so we get garbled sound or, if the output happened to be muted before rebooting, no sound. To make that driver happy, hook our driver's cleanup function into the shutdown notifier and ensure that the chip gets reset. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_lib.c | 21 ++++++++++++++++++--- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_wm87x6.c | 1 + 4 files changed, 21 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6147216af744..a3409edcfb50 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -155,6 +155,7 @@ void oxygen_pci_remove(struct pci_dev *pci); int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); int oxygen_pci_resume(struct pci_dev *pci); #endif +void oxygen_pci_shutdown(struct pci_dev *pci); /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index fad03d64e3ad..7e93cf884437 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -519,16 +519,21 @@ static void oxygen_init(struct oxygen *chip) } } -static void oxygen_card_free(struct snd_card *card) +static void oxygen_shutdown(struct oxygen *chip) { - struct oxygen *chip = card->private_data; - spin_lock_irq(&chip->reg_lock); chip->interrupt_mask = 0; chip->pcm_running = 0; oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); +} + +static void oxygen_card_free(struct snd_card *card) +{ + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); if (chip->irq >= 0) free_irq(chip->irq, chip); flush_scheduled_work(); @@ -778,3 +783,13 @@ int oxygen_pci_resume(struct pci_dev *pci) } EXPORT_SYMBOL(oxygen_pci_resume); #endif /* CONFIG_PM */ + +void oxygen_pci_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); + chip->model.cleanup(chip); +} +EXPORT_SYMBOL(oxygen_pci_shutdown); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index f03a2f2cffee..06c863e86e3d 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -95,6 +95,7 @@ static struct pci_driver xonar_driver = { .suspend = oxygen_pci_suspend, .resume = oxygen_pci_resume, #endif + .shutdown = oxygen_pci_shutdown, }; static int __init alsa_card_xonar_init(void) diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index dbc4b89d74e4..0b89932fb8c4 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -193,6 +193,7 @@ static void xonar_ds_init(struct oxygen *chip) static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); } static void xonar_ds_suspend(struct oxygen *chip) -- cgit v1.2.3 From fe6ce80ae25953d95ebaf9bce27b585218cda25c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:38:49 +0200 Subject: ALSA: virtuoso: fix setting of Xonar DS line-in/mic-in controls The Line and Mic inputs cannot be used at the same time, so the driver has to automatically disable one of them if both are set. However, it forgot to notify userspace about this change, so the mixer state would be inconsistent. To fix this, check if the other control gets muted, and send a notification event in this case. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 0b89932fb8c4..b82c1cfa96f5 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -53,6 +53,8 @@ struct xonar_wm87x6 { struct xonar_generic generic; u16 wm8776_regs[0x17]; u16 wm8766_regs[0x10]; + struct snd_kcontrol *line_adcmux_control; + struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; }; @@ -604,6 +606,7 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; struct xonar_wm87x6 *data = chip->model_data; + struct snd_kcontrol *other_ctl; unsigned int mux_bit = ctl->private_value; u16 reg; int changed; @@ -611,8 +614,18 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = data->wm8776_regs[WM8776_ADCMUX]; if (value->value.integer.value[0]) { - reg &= ~0x003; reg |= mux_bit; + /* line-in and mic-in are exclusive */ + mux_bit ^= 3; + if (reg & mux_bit) { + reg &= ~mux_bit; + if (mux_bit == 1) + other_ctl = data->line_adcmux_control; + else + other_ctl = data->mic_adcmux_control; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &other_ctl->id); + } } else reg &= ~mux_bit; changed = reg != data->wm8776_regs[WM8776_ADCMUX]; @@ -964,7 +977,13 @@ static int xonar_ds_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; + if (!strcmp(ctl->id.name, "Line Capture Switch")) + data->line_adcmux_control = ctl; + else if (!strcmp(ctl->id.name, "Mic Capture Switch")) + data->mic_adcmux_control = ctl; } + if (!data->line_adcmux_control || !data->mic_adcmux_control) + return -ENXIO; BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { ctl = snd_ctl_new1(&lc_controls[i], chip); -- cgit v1.2.3 From 76195fb096ca6db2f8bbaffb96e3025aaf1649a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:27:02 +0200 Subject: ALSA: usb - Release capture substream URBs properly Due to the wrong "return" in the loop, a capture substream won't be released at disconnection properly if the device is capture only and has no playback substream. This caused Oops occasionally at the device reconnection. Reported-by: Kim Minhyoung Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/card.c b/sound/usb/card.c index 32e4be8a187c..4eabafa5b037 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -126,7 +126,7 @@ static void snd_usb_stream_disconnect(struct list_head *head) for (idx = 0; idx < 2; idx++) { subs = &as->substream[idx]; if (!subs->num_formats) - return; + continue; snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; } -- cgit v1.2.3 From a769cbcf60cee51f4431c0938acd39e7e5b76b8d Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 7 Sep 2010 14:36:22 -0500 Subject: ALSA: hda - Add errata initverb sequence for CS42xx codecs Add init verb sequence for errata ER880C3 http://www.cirrus.com/en/pubs/errata/ER880C3.pdf Signed-off-by: Brian Austin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 50 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4ef5efaaaef1..488fd9ade1ba 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -972,6 +972,53 @@ static struct hda_verb cs_coef_init_verbs[] = { {} /* terminator */ }; +/* Errata: CS4207 rev C0/C1/C2 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf + * + * 6. At high temperature (TA > +85°C), the digital supply current (IVD) + * may be excessive (up to an additional 200 μA), which is most easily + * observed while the part is being held in reset (RESET# active low). + * + * Root Cause: At initial powerup of the device, the logic that drives + * the clock and write enable to the S/PDIF SRC RAMs is not properly + * initialized. + * Certain random patterns will cause a steady leakage current in those + * RAM cells. The issue will resolve once the SRCs are used (turned on). + * + * Workaround: The following verb sequence briefly turns on the S/PDIF SRC + * blocks, which will alleviate the issue. + */ + +static struct hda_verb cs_errata_init_verbs[] = { + {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ + {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x9999}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0xa412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0009}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Rx: D0 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Tx: D0 */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0x2412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0000}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0008}, + {0x11, AC_VERB_SET_PROC_STATE, 0x00}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */ + /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */ + + {} /* terminator */ +}; + /* SPDIF setup */ static void init_digital(struct hda_codec *codec) { @@ -991,6 +1038,9 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); if (spec->gpio_mask) { -- cgit v1.2.3 From 080dc7bc2562615a5be0a705a9d1a8c24eb198d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:38:41 +0200 Subject: ALSA: hda - Enable PC-beep for EeePC with ALC269 codec EeePC 1001HAG has a similar problem like other ASUS machine, which doesn't set the codec SSID properly for indicating the beep capability. To enable PC-beep again, put this to the whitelist. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 627bf9963368..bcbf9160ed81 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5334,6 +5334,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- cgit v1.2.3 From ab5a6ebee38f3ed311f0565ecd3fba5cf111564a Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 8 Sep 2010 09:00:17 +0200 Subject: ALSA: hda - Adding support for new IDT 92HD90BXX and 92HD91BXX codecs Adding support for digital MIC in 92HD83/90/91XXX codecs family. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 79 ++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d226edd1e143..82d1e4378621 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -382,6 +382,11 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x20, 0x40, }; +#define STAC92HD83XXX_NUM_DMICS 2 +static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { + 0x11, 0x20, 0 +}; + #define STAC92HD83XXX_NUM_CAPS 2 static unsigned long stac92hd83xxx_capvols[] = { HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), @@ -4695,6 +4700,36 @@ static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) } } +/* get the pin connection (fixed, none, etc) */ +static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int cfg; + + cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); + return get_defcfg_connect(cfg); +} + +static int stac92xx_connected_ports(struct hda_codec *codec, + hda_nid_t *nids, int num_nids) +{ + struct sigmatel_spec *spec = codec->spec; + int idx, num; + unsigned int def_conf; + + for (num = 0; num < num_nids; num++) { + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == nids[num]) + break; + if (idx >= spec->num_pins) + break; + def_conf = stac_get_defcfg_connect(codec, idx); + if (def_conf == AC_JACK_PORT_NONE) + break; + } + return num; +} + static void stac92xx_mic_detect(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -5325,6 +5360,8 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->linear_tone_beep = 1; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; + spec->dmic_nids = stac92hd83xxx_dmic_nids; + spec->dmux_nids = stac92hd83xxx_mux_nids; spec->mux_nids = stac92hd83xxx_mux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids); spec->adc_nids = stac92hd83xxx_adc_nids; @@ -5370,9 +5407,13 @@ again: case 0x111d76d4: case 0x111d7605: case 0x111d76d5: + case 0x111d76e7: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; + spec->num_dmics = stac92xx_connected_ports(codec, + stac92hd83xxx_dmic_nids, + STAC92HD83XXX_NUM_DMICS); break; } @@ -5431,36 +5472,6 @@ again: return 0; } -/* get the pin connection (fixed, none, etc) */ -static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) -{ - struct sigmatel_spec *spec = codec->spec; - unsigned int cfg; - - cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); - return get_defcfg_connect(cfg); -} - -static int stac92hd71bxx_connected_ports(struct hda_codec *codec, - hda_nid_t *nids, int num_nids) -{ - struct sigmatel_spec *spec = codec->spec; - int idx, num; - unsigned int def_conf; - - for (num = 0; num < num_nids; num++) { - for (idx = 0; idx < spec->num_pins; idx++) - if (spec->pin_nids[idx] == nids[num]) - break; - if (idx >= spec->num_pins) - break; - def_conf = stac_get_defcfg_connect(codec, idx); - if (def_conf == AC_JACK_PORT_NONE) - break; - } - return num; -} - static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, hda_nid_t dig0pin) { @@ -5599,7 +5610,7 @@ again: case 0x111d76b5: spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92hd71bxx_connected_ports(codec, + spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS); break; @@ -5631,7 +5642,7 @@ again: snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; - spec->num_dmics = stac92hd71bxx_connected_ports(codec, + spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS - 1); break; @@ -5645,7 +5656,7 @@ again: default: spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; - spec->num_dmics = stac92hd71bxx_connected_ports(codec, + spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS); break; @@ -6327,6 +6338,8 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx}, {} /* terminator */ }; -- cgit v1.2.3 From 263d0328c46995d8e4fb478005177839104483d2 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 8 Sep 2010 08:56:03 +0200 Subject: ALSA: hda - Improve input control names for IDT/STAC codecs Changing the way the input controls are named using port connection type and jack location info. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 107 +++++++++++++++++++++++++++++++++-------- 1 file changed, 88 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 82d1e4378621..7f09e140953e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -191,6 +191,11 @@ struct sigmatel_mic_route { signed char dmux_idx; }; +struct unique_input_names { + int num; + char uname[HDA_MAX_NUM_INPUTS][32]; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -307,6 +312,7 @@ struct sigmatel_spec { struct hda_input_mux private_imux; struct hda_input_mux private_smux; struct hda_input_mux private_mono_mux; + struct unique_input_names private_u_inp_names; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -3452,6 +3458,76 @@ static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, return 1; } +static const char *get_input_src_label(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf; + + def_conf = snd_hda_codec_get_pincfg(codec, nid); + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_MIC_IN: + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || + ((get_defcfg_location(def_conf) & 0xf0) + == AC_JACK_LOC_INTERNAL)) + return "Internal Mic"; + if ((get_defcfg_location(def_conf) & 0xf0) + == AC_JACK_LOC_SEPARATE) + return "Dock Mic"; + if (get_defcfg_location(def_conf) == AC_JACK_LOC_REAR) + return "Rear Mic"; + return "Mic"; + case AC_JACK_LINE_IN: + if ((get_defcfg_location(def_conf) & 0xf0) + == AC_JACK_LOC_SEPARATE) + return "Dock Line"; + return "Line"; + case AC_JACK_AUX: + return "Aux"; + case AC_JACK_CD: + return "CD"; + case AC_JACK_SPDIF_IN: + return "SPDIF In"; + case AC_JACK_DIG_OTHER_IN: + return "Digital In"; + } + + snd_printd("invalid inp pin %02x device config %08x", nid, def_conf); + return NULL; +} + +static const char *get_unique_inp_src_label(struct hda_codec *codec, + hda_nid_t nid) +{ + int i, n; + const char *label; + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *dimux = &spec->private_dimux; + struct unique_input_names *unames = &spec->private_u_inp_names; + + label = get_input_src_label(codec, nid); + n = 0; + + for (i = 0; i < imux->num_items; i++) { + if (!strncmp(label, imux->items[i].label, strlen(label))) + n++; + } + if (snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { + for (i = 0; i < dimux->num_items; i++) { + if (!strncmp(label, dimux->items[i].label, + strlen(label))) + n++; + } + } + if (n > 0 && unames->num < HDA_MAX_NUM_INPUTS) { + sprintf(&unames->uname[unames->num][0], "%.28s %d", label, n); + label = &unames->uname[unames->num][0]; + unames->num++; + } + + return label; +} + /* create playback/capture controls for input pins on dmic capable codecs */ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -3459,24 +3535,13 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; struct hda_input_mux *dimux = &spec->private_dimux; - int err, i, active_mics; + int err, i; unsigned int def_conf; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; - active_mics = 0; - for (i = 0; i < spec->num_dmics; i++) { - /* check the validity: sometimes it's a dead vendor-spec node */ - if (get_wcaps_type(get_wcaps(codec, spec->dmic_nids[i])) - != AC_WID_PIN) - continue; - def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) - active_mics++; - } - for (i = 0; i < spec->num_dmics; i++) { hda_nid_t nid; int index; @@ -3493,10 +3558,9 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (index < 0) continue; - if (active_mics == 1) - label = "Digital Mic"; - else - label = stac92xx_dmic_labels[dimux->num_items]; + label = get_unique_inp_src_label(codec, nid); + if (label == NULL) + return -EINVAL; err = create_elem_capture_vol(codec, nid, label, 0, HDA_INPUT); if (err < 0) @@ -3618,6 +3682,7 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; int i, j, type_idx = 0; + const char *label; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; @@ -3637,14 +3702,18 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const type_idx++; else type_idx = 0; + + label = get_unique_inp_src_label(codec, nid); + if (label == NULL) + return -EINVAL; + err = create_elem_capture_vol(codec, nid, - auto_pin_cfg_labels[i], type_idx, + label, type_idx, HDA_INPUT); if (err < 0) return err; - imux->items[imux->num_items].label = - snd_hda_get_input_pin_label(cfg, i); + imux->items[imux->num_items].label = label; imux->items[imux->num_items].index = index; imux->num_items++; } -- cgit v1.2.3 From e4ee8dd8afcbcbe502fa8a3d3af6eb09c96dd806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 09:58:12 +0200 Subject: ALSA: msnd-classic: Fix invalid cfg parameter The driver doesn't probe the device properly because of left-over cfg[] that isn't used at all for msnd-classic device. This is only for msnd- pinnacle. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 5f3e68401f90..91d6023a63e5 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -764,9 +764,9 @@ static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +#ifndef MSND_CLASSIC static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; -#ifndef MSND_CLASSIC /* Extra Peripheral Configuration (Default: Disable) */ static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -894,7 +894,11 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) struct snd_card *card; struct snd_msnd *chip; - if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { + if (has_isapnp(idx) +#ifndef MSND_CLASSIC + || cfg[idx] == SNDRV_AUTO_PORT +#endif + ) { printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } -- cgit v1.2.3 From 27f7ad53829f79e799a253285318bff79ece15bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Sep 2010 09:13:45 +0200 Subject: ALSA: seq/oss - Fix double-free at error path of snd_seq_oss_open() The error handling in snd_seq_oss_open() has several bad codes that do dereferecing released pointers and double-free of kmalloc'ed data. The object dp is release in free_devinfo() that is called via private_free callback. The rest shouldn't touch this object any more. The patch changes delete_port() to call kfree() in any case, and gets rid of unnecessary calls of destructors in snd_seq_oss_open(). Fixes CVE-2010-3080. Reported-and-tested-by: Tavis Ormandy Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 685712276ac9..69cd7b3c362d 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -281,13 +281,10 @@ snd_seq_oss_open(struct file *file, int level) return 0; _error: - snd_seq_oss_writeq_delete(dp->writeq); - snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - delete_port(dp); delete_seq_queue(dp->queue); - kfree(dp); + delete_port(dp); return rc; } @@ -350,8 +347,10 @@ create_port(struct seq_oss_devinfo *dp) static int delete_port(struct seq_oss_devinfo *dp) { - if (dp->port < 0) + if (dp->port < 0) { + kfree(dp); return 0; + } debug_printk(("delete_port %i\n", dp->port)); return snd_seq_event_port_detach(dp->cseq, dp->port); -- cgit v1.2.3 From 122661b67899980f1372812d907e73ebcfb3d037 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 14:57:04 +0200 Subject: ALSA: hda - Fix wrong HP pin detection in snd_hda_parse_pin_def_config() snd_hda_parse_pin_def_config() has some workaround for re-assigning some pins declared as headphones to line-outs. This didn't work properly for some cases because it used memmove() stupidly wrongly. Reference: Novell bnc#637263 https://bugzilla.novell.com/show_bug.cgi?id=637263 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3827092cc1d2..14829210ef0b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4536,7 +4536,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_outs--; memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i - 1, sequences_hp + i, + memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } } -- cgit v1.2.3 From 03642c9a444079aa13f0864383a8f9ca04bfd198 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 15:28:19 +0200 Subject: ALSA: hda - Clear left-over hp_pins in snd_hda_parse_pin_def_config() In snd_hda_parse_def_config(), some unused values may remain in hp_pins[] array during the headphone-reassignment workaround. This patch clears the unused array members. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2980c277847a..bfdde7b0bafb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4558,6 +4558,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } + memset(cfg->hp_pins + cfg->hp_outs, 0, + sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - cfg->hp_outs)); } /* sort by sequence */ -- cgit v1.2.3 From bb35febd16fe5ac8c30f9116a25210c4f63a5267 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 15:30:49 +0200 Subject: ALSA: hda - Support multiple headphone auto-mute Currently headphone auto-mute using alc_automute_pin() assumes only the single pin used for the headphone output. Since there are devices with multiple headphone jacks, we need to check all these pins there, too. Also this patch merges the common code between alc_automute_pin() and alc_automute_amp() helper functions. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 92 +++++++++++++++++++++++-------------------- 1 file changed, 50 insertions(+), 42 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 81e4b1d957c5..ee59df7a41f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -990,25 +990,46 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, alc_fix_pll(codec); } -static void alc_automute_pin(struct hda_codec *codec) +static void alc_automute_speaker(struct hda_codec *codec, int pinctl) { struct alc_spec *spec = codec->spec; - unsigned int nid = spec->autocfg.hp_pins[0]; + unsigned int mute; + hda_nid_t nid; int i; - if (!nid) - return; - spec->jack_present = snd_hda_jack_detect(codec, nid); + spec->jack_present = 0; + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + nid = spec->autocfg.hp_pins[i]; + if (!nid) + break; + if (snd_hda_jack_detect(codec, nid)) { + spec->jack_present = 1; + break; + } + } + + mute = spec->jack_present ? HDA_AMP_MUTE : 0; + /* Toggle internal speakers muting */ for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { nid = spec->autocfg.speaker_pins[i]; if (!nid) break; - snd_hda_codec_write(codec, nid, 0, + if (pinctl) { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->jack_present ? 0 : PIN_OUT); + } else { + snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } } } +static void alc_automute_pin(struct hda_codec *codec) +{ + alc_automute_speaker(codec, 1); +} + static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid) { @@ -1236,24 +1257,35 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) static void alc_init_auto_hp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; - if (!spec->autocfg.hp_pins[0]) - return; + if (!cfg->hp_pins[0]) { + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + return; + } - if (!spec->autocfg.speaker_pins[0]) { - if (spec->autocfg.line_out_pins[0] && - spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) - spec->autocfg.speaker_pins[0] = - spec->autocfg.line_out_pins[0]; - else + if (!cfg->speaker_pins[0]) { + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) return; + memcpy(cfg->speaker_pins, cfg->line_out_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = cfg->line_outs; + } + + if (!cfg->hp_pins[0]) { + memcpy(cfg->hp_pins, cfg->line_out_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = cfg->line_outs; } - snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", - spec->autocfg.hp_pins[0]); - snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0, + for (i = 0; i < cfg->hp_outs; i++) { + snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n", + cfg->hp_pins[i]); + snd_hda_codec_write_cache(codec, cfg->hp_pins[i], 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); + } spec->unsol_event = alc_sku_unsol_event; } @@ -1711,31 +1743,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - unsigned int mute; - hda_nid_t nid; - int i; - - spec->jack_present = 0; - for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { - nid = spec->autocfg.hp_pins[i]; - if (!nid) - break; - if (snd_hda_jack_detect(codec, nid)) { - spec->jack_present = 1; - break; - } - } - - mute = spec->jack_present ? HDA_AMP_MUTE : 0; - /* Toggle internal speakers muting */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { - nid = spec->autocfg.speaker_pins[i]; - if (!nid) - break; - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); - } + alc_automute_speaker(codec, 0); } static void alc_automute_amp_unsol_event(struct hda_codec *codec, -- cgit v1.2.3 From 033688a5a80f9d56b2e7d56c4cb8188ae1448919 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 15:47:09 +0200 Subject: ALSA: hda - Add multiple headphone support to ALC262 codec This patch changes the alc262 auto-parser to allow multiple pins assigned for a single purpose (line-out, headphone or speaker). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 71 +++++++++++++++++++++++++------------------ 1 file changed, 42 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ee59df7a41f8..26069e397fc9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11824,7 +11824,7 @@ static int alc262_check_volbit(hda_nid_t nid) } static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, - const char *pfx, int *vbits) + const char *pfx, int *vbits, int idx) { unsigned long val; int vbit; @@ -11839,11 +11839,11 @@ static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT); - return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, val); + return __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, idx, val); } static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid, - const char *pfx) + const char *pfx, int idx) { unsigned long val; @@ -11853,7 +11853,7 @@ static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid, val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val); + return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, idx, val); } /* add playback controls from the parsed DAC table */ @@ -11862,7 +11862,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, { const char *pfx; int vbits; - int err; + int i, err; spec->multiout.num_dacs = 1; /* only use one dac */ spec->multiout.dac_nids = spec->private_dac_nids; @@ -11872,39 +11872,52 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, pfx = "Master"; else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) pfx = "Speaker"; + else if (cfg->line_out_type == AUTO_PIN_HP_OUT) + pfx = "Headphone"; else pfx = "Front"; - err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[0], pfx); - if (err < 0) - return err; - err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[0], "Speaker"); - if (err < 0) - return err; - err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[0], "Headphone"); - if (err < 0) - return err; + for (i = 0; i < 2; i++) { + err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[i], pfx, i); + if (err < 0) + return err; + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[i], + "Speaker", i); + if (err < 0) + return err; + } + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[i], + "Headphone", i); + if (err < 0) + return err; + } + } vbits = alc262_check_volbit(cfg->line_out_pins[0]) | alc262_check_volbit(cfg->speaker_pins[0]) | alc262_check_volbit(cfg->hp_pins[0]); if (vbits == 1 || vbits == 2) pfx = "Master"; /* only one mixer is used */ - else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) - pfx = "Speaker"; - else - pfx = "Front"; vbits = 0; - err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[0], pfx, &vbits); - if (err < 0) - return err; - err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[0], "Speaker", - &vbits); - if (err < 0) - return err; - err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[0], "Headphone", - &vbits); - if (err < 0) - return err; + for (i = 0; i < 2; i++) { + err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[i], pfx, + &vbits, i); + if (err < 0) + return err; + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[i], + "Speaker", &vbits, i); + if (err < 0) + return err; + } + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[i], + "Headphone", &vbits, i); + if (err < 0) + return err; + } + } return 0; } -- cgit v1.2.3 From 18675e4283f575594d55ef1239c14ab5b4de53b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 15:55:44 +0200 Subject: ALSA: hda - Add fixup for FSC Celsius H270 Added a fixup table for ALC262 codec containing the entry for FSC Celsius H270. Now both headphone jacks are detected properly as headphones. Reference: Novell bnc637263 https://bugzilla.novell.com/show_bug.cgi?id=637263 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 26069e397fc9..f11a9ca2c4b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12205,6 +12205,35 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { {} }; +/* + * Pin config fixes + */ +enum { + PINFIX_FSC_H270, +}; + +static const struct alc_fixup alc262_fixups[] = { + [PINFIX_FSC_H270] = { + .pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x15, 0x0221142f }, /* front HP */ + { 0x1b, 0x0121141f }, /* rear HP */ + { } + } + }, + [PINFIX_PB_M5210] = { + .verbs = (const struct hda_verb[]) { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, + {} + } + }, +}; + +static struct snd_pci_quirk alc262_fixup_tbl[] = { + SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), + {} +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc262_loopbacks alc880_loopbacks @@ -12628,6 +12657,9 @@ static int patch_alc262(struct hda_codec *codec) board_config = ALC262_AUTO; } + if (board_config == ALC262_AUTO) + alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 1); + if (board_config == ALC262_AUTO) { /* automatic parse from the BIOS config */ err = alc262_parse_auto_config(codec); @@ -12696,6 +12728,9 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->no_analog && has_cdefine_beep(codec)) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (board_config == ALC262_AUTO) + alc_pick_fixup(codec, alc262_fixup_tbl, alc262_fixups, 0); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; -- cgit v1.2.3 From a7a13d0676335a7dc9dd72264cca02606e43aaba Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 9 Sep 2010 00:11:41 +0200 Subject: ALSA: rawmidi: fix the get next midi device ioctl If we pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then the "next device" should be -1. This function just returns device + 1. But the main thing is that "device + 1" can lead to a (harmless) integer overflow and that annoys static analysis tools. [fix the case for device == SNDRV_RAWMIDI_DEVICE by tiwai] Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326c37d4..a7868ad4d530 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,6 +829,8 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ + device = SNDRV_RAWMIDI_DEVICES - 1; mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_RAWMIDI_DEVICES) { -- cgit v1.2.3 From 6cb3b707f95954ac18f19b4b3919af235738371a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 9 Sep 2010 08:51:44 +0200 Subject: ALSA: HDA: Add fixup pins for Ideapad Y550 By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting. BugLink: https://launchpad.net/bugs/611803 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f11a9ca2c4b2..0c25d22be875 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18963,6 +18963,26 @@ static void alc662_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC662_FIXUP_IDEAPAD, +}; + +static const struct alc_fixup alc662_fixups[] = { + [ALC662_FIXUP_IDEAPAD] = { + .pins = (const struct alc_pincfg[]) { + { 0x17, 0x99130112 }, /* subwoofer */ + { } + } + }, +}; + +static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), + {} +}; + + + static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; @@ -18995,6 +19015,7 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config == ALC662_AUTO) { + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 1); /* automatic parse from the BIOS config */ err = alc662_parse_auto_config(codec); if (err < 0) { @@ -19053,8 +19074,11 @@ static int patch_alc662(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; - if (board_config == ALC662_AUTO) + if (board_config == ALC662_AUTO) { spec->init_hook = alc662_auto_init; + alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 0); + } + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; -- cgit v1.2.3 From da0dab5ecb5001f76e739e71ee199db4c61e7af2 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:18:35 +0200 Subject: ALSA: virtuoso: fix WM8766 register writes with MSB The check for the volume update latch bit was accidentally in the wrong function, where it would prevent the MSB from being written, instead of correctly ignoring it for cached values. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index b82c1cfa96f5..4346006df3ec 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -97,8 +97,12 @@ static void wm8766_write(struct oxygen *chip, (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); - if (reg < ARRAY_SIZE(data->wm8766_regs)) + if (reg < ARRAY_SIZE(data->wm8766_regs)) { + if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || + (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) + value &= ~WM8766_UPDATE; data->wm8766_regs[reg] = value; + } } static void wm8766_write_cached(struct oxygen *chip, @@ -107,12 +111,8 @@ static void wm8766_write_cached(struct oxygen *chip, struct xonar_wm87x6 *data = chip->model_data; if (reg >= ARRAY_SIZE(data->wm8766_regs) || - value != data->wm8766_regs[reg]) { - if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || - (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) - value &= ~WM8766_UPDATE; + value != data->wm8766_regs[reg]) wm8766_write(chip, reg, value); - } } static void wm8776_registers_init(struct oxygen *chip) -- cgit v1.2.3 From 9bac84edf0360ac94a27308778ef98dc9068777c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:19:21 +0200 Subject: ALSA: virtuoso: fix Xonar DS input switches Use the correct number, register bits, and names for the input switches. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 4346006df3ec..fb3f95ccafa1 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -29,6 +29,13 @@ * GPIO 6 -> route input jack to input 1/2 (1/0) * GPIO 7 -> enable output to speakers * GPIO 8 -> enable output to speakers + * + * WM8766: + * + * input 1 <- line + * input 2 <- mic + * input 3 <- front mic + * input 4 <- aux */ #include @@ -896,7 +903,10 @@ static const struct snd_kcontrol_new ds_controls[] = { .put = wm8776_input_mux_put, .private_value = 1 << 1, }, - WM8776_BIT_SWITCH("Aux", WM8776_ADCMUX, 1 << 2, 0, 0), + WM8776_BIT_SWITCH("Front Mic Capture Switch", + WM8776_ADCMUX, 1 << 2, 0, 0), + WM8776_BIT_SWITCH("Aux Capture Switch", + WM8776_ADCMUX, 1 << 3, 0, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "ADC Filter Capture Enum", -- cgit v1.2.3 From 435feac648cab190990aa0bf9355f77d1f082db3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:20:29 +0200 Subject: ALSA: virtuoso: add Xonar DS headphone jack detection Now that the polarity of the headphone detection pin is known, replace the debugging message with a proper jack plug input device. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 1 + sound/pci/oxygen/xonar_wm87x6.c | 25 +++++++++++++++++++------ 2 files changed, 20 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index e7a8cd058efb..b40f2b8df536 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -817,6 +817,7 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 100/200 (Xonar)" select SND_OXYGEN_LIB + select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index fb3f95ccafa1..9d57b5eee3f5 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -25,8 +25,8 @@ * SPI 0 -> WM8766 (surround, center/LFE, back) * SPI 1 -> WM8776 (front, input) * - * GPIO 4 <- headphone detect - * GPIO 6 -> route input jack to input 1/2 (1/0) + * GPIO 4 <- headphone detect, 0 = plugged + * GPIO 6 -> route input jack to mic-in (0) or line-in (1) * GPIO 7 -> enable output to speakers * GPIO 8 -> enable output to speakers * @@ -42,6 +42,7 @@ #include #include #include +#include #include #include #include @@ -63,6 +64,7 @@ struct xonar_wm87x6 { struct snd_kcontrol *line_adcmux_control; struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; + struct snd_jack *hp_jack; }; static void wm8776_write(struct oxygen *chip, @@ -177,6 +179,16 @@ static void wm8776_init(struct oxygen *chip) wm8776_registers_init(chip); } +static void xonar_ds_report_hp_jack(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + u16 bits; + + bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + snd_jack_report(data->hp_jack, + bits & GPIO_DS_HP_DETECT ? 0 : SND_JACK_HEADPHONE); +} + static void xonar_ds_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -195,6 +207,10 @@ static void xonar_ds_init(struct oxygen *chip) xonar_enable_output(chip); + snd_jack_new(chip->card, "Headphone", + SND_JACK_HEADPHONE, &data->hp_jack); + xonar_ds_report_hp_jack(chip); + snd_component_add(chip->card, "WM8776"); snd_component_add(chip->card, "WM8766"); } @@ -332,10 +348,7 @@ static void update_wm87x6_mute(struct oxygen *chip) static void xonar_ds_gpio_changed(struct oxygen *chip) { - u16 bits; - - bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - snd_printk(KERN_INFO "HP detect: %d\n", !!(bits & GPIO_DS_HP_DETECT)); + xonar_ds_report_hp_jack(chip); } static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, -- cgit v1.2.3 From 84cf83a28d4a3cd1fac1384cbaa4ed0ba650d309 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:23:06 +0200 Subject: ALSA: virtuoso: automatically handle Xonar DS headphone routing Automatically mute the speaker outputs as long as a headphone is plugged. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 57 +++++++++++++++++++++++++++++++---------- 1 file changed, 44 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 9d57b5eee3f5..cee07fe3aa36 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -27,8 +27,8 @@ * * GPIO 4 <- headphone detect, 0 = plugged * GPIO 6 -> route input jack to mic-in (0) or line-in (1) - * GPIO 7 -> enable output to speakers - * GPIO 8 -> enable output to speakers + * GPIO 7 -> enable output to front L/R speaker channels + * GPIO 8 -> enable output to other speaker channels and front panel headphone * * WM8766: * @@ -52,7 +52,8 @@ #define GPIO_DS_HP_DETECT 0x0010 #define GPIO_DS_INPUT_ROUTE 0x0040 -#define GPIO_DS_OUTPUT_ENABLE 0x0180 +#define GPIO_DS_OUTPUT_FRONTLR 0x0080 +#define GPIO_DS_OUTPUT_ENABLE 0x0100 #define LC_CONTROL_LIMITER 0x40000000 #define LC_CONTROL_ALC 0x20000000 @@ -150,7 +151,10 @@ static void wm8776_registers_init(struct oxygen *chip) static void wm8766_registers_init(struct oxygen *chip) { + struct xonar_wm87x6 *data = chip->model_data; + wm8766_write(chip, WM8766_RESET, 0); + wm8766_write(chip, WM8766_DAC_CTRL, data->wm8766_regs[WM8766_DAC_CTRL]); wm8766_write(chip, WM8766_INT_CTRL, WM8766_FMT_LJUST | WM8766_IWL_24); wm8766_write(chip, WM8766_DAC_CTRL2, WM8766_ZCD | (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); @@ -179,14 +183,38 @@ static void wm8776_init(struct oxygen *chip) wm8776_registers_init(chip); } -static void xonar_ds_report_hp_jack(struct oxygen *chip) +static void wm8766_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; - u16 bits; - bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - snd_jack_report(data->hp_jack, - bits & GPIO_DS_HP_DETECT ? 0 : SND_JACK_HEADPHONE); + data->wm8766_regs[WM8766_DAC_CTRL] = + WM8766_PL_LEFT_LEFT | WM8766_PL_RIGHT_RIGHT; + wm8766_registers_init(chip); +} + +static void xonar_ds_handle_hp_jack(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + bool hp_plugged; + unsigned int reg; + + mutex_lock(&chip->mutex); + + hp_plugged = !(oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_DS_HP_DETECT); + + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + hp_plugged ? 0 : GPIO_DS_OUTPUT_FRONTLR, + GPIO_DS_OUTPUT_FRONTLR); + + reg = data->wm8766_regs[WM8766_DAC_CTRL] & ~WM8766_MUTEALL; + if (hp_plugged) + reg |= WM8766_MUTEALL; + wm8766_write_cached(chip, WM8766_DAC_CTRL, reg); + + snd_jack_report(data->hp_jack, hp_plugged ? SND_JACK_HEADPHONE : 0); + + mutex_unlock(&chip->mutex); } static void xonar_ds_init(struct oxygen *chip) @@ -197,10 +225,12 @@ static void xonar_ds_init(struct oxygen *chip) data->generic.output_enable_bit = GPIO_DS_OUTPUT_ENABLE; wm8776_init(chip); - wm8766_registers_init(chip); + wm8766_init(chip); - oxygen_write16_masked(chip, OXYGEN_GPIO_CONTROL, GPIO_DS_INPUT_ROUTE, - GPIO_DS_HP_DETECT | GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DS_INPUT_ROUTE | GPIO_DS_OUTPUT_FRONTLR); + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DS_HP_DETECT); oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DS_INPUT_ROUTE); oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_DS_HP_DETECT); chip->interrupt_mask |= OXYGEN_INT_GPIO; @@ -209,7 +239,7 @@ static void xonar_ds_init(struct oxygen *chip) snd_jack_new(chip->card, "Headphone", SND_JACK_HEADPHONE, &data->hp_jack); - xonar_ds_report_hp_jack(chip); + xonar_ds_handle_hp_jack(chip); snd_component_add(chip->card, "WM8776"); snd_component_add(chip->card, "WM8766"); @@ -231,6 +261,7 @@ static void xonar_ds_resume(struct oxygen *chip) wm8776_registers_init(chip); wm8766_registers_init(chip); xonar_enable_output(chip); + xonar_ds_handle_hp_jack(chip); } static void wm8776_adc_hardware_filter(unsigned int channel, @@ -348,7 +379,7 @@ static void update_wm87x6_mute(struct oxygen *chip) static void xonar_ds_gpio_changed(struct oxygen *chip) { - xonar_ds_report_hp_jack(chip); + xonar_ds_handle_hp_jack(chip); } static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, -- cgit v1.2.3 From 2dbf0ea29c1e4dff4ee5f0c59b367168fa2e5a40 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:24:35 +0200 Subject: ALSA: virtuoso: Xonar DS: add stereo upmixing to center/LFE channels Add the possibility to route a mix of the two channels of stereo data to the center and LFE outputs. Due to a WM8766 restriction, all surround and back channels also get the mixed L/R signal in this case. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index cee07fe3aa36..aceaaa036da6 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -377,6 +377,24 @@ static void update_wm87x6_mute(struct oxygen *chip) (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); } +static void update_wm8766_center_lfe_mix(struct oxygen *chip, bool mixed) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int reg; + + /* + * The WM8766 can mix left and right channels, but this setting + * applies to all three stereo pairs. + */ + reg = data->wm8766_regs[WM8766_DAC_CTRL] & + ~(WM8766_PL_LEFT_MASK | WM8766_PL_RIGHT_MASK); + if (mixed) + reg |= WM8766_PL_LEFT_LRMIX | WM8766_PL_RIGHT_LRMIX; + else + reg |= WM8766_PL_LEFT_LEFT | WM8766_PL_RIGHT_RIGHT; + wm8766_write_cached(chip, WM8766_DAC_CTRL, reg); +} + static void xonar_ds_gpio_changed(struct oxygen *chip) { xonar_ds_handle_hp_jack(chip); @@ -1067,6 +1085,7 @@ static const struct oxygen_model model_xonar_ds = { .set_adc_params = set_wm8776_adc_params, .update_dac_volume = update_wm87x6_volume, .update_dac_mute = update_wm87x6_mute, + .update_center_lfe_mix = update_wm8766_center_lfe_mix, .gpio_changed = xonar_ds_gpio_changed, .dac_tlv = wm87x6_dac_db_scale, .model_data_size = sizeof(struct xonar_wm87x6), -- cgit v1.2.3 From 99f08bf59019ca6c9056f10ee8f7e1ba6663251c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:25:29 +0200 Subject: ALSA: oxygen: fix CONFIG_SND_OXYGEN_LIB dependency selection As the select directive does not handle indirect dependencies, select those explicitly in the driver sections. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index b40f2b8df536..0e75d558f303 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -207,12 +207,12 @@ config SND_CMIPCI config SND_OXYGEN_LIB tristate - select SND_PCM - select SND_MPU401_UART config SND_OXYGEN tristate "C-Media 8788 (Oxygen)" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART help Say Y here to include support for sound cards based on the C-Media CMI8788 (Oxygen HD Audio) chip: @@ -581,6 +581,8 @@ config SND_HDSPM config SND_HIFIER tristate "TempoTec HiFier Fantasia" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART help Say Y here to include support for the MediaTek/TempoTec HiFier Fantasia sound card. @@ -817,6 +819,8 @@ config SND_VIA82XX_MODEM config SND_VIRTUOSO tristate "Asus Virtuoso 100/200 (Xonar)" select SND_OXYGEN_LIB + select SND_PCM + select SND_MPU401_UART select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the -- cgit v1.2.3 From 51485e8e24919be10bd61dba1dede0032de2d952 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Sep 2010 12:26:52 +0200 Subject: ALSA: virtuoso: update Kconfig text Update the Xonar config texts with the latest information about the Xonar DS, HDAV1.3 Slim, and Xense. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 0e75d558f303..12e34653b8a8 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -817,17 +817,17 @@ config SND_VIA82XX_MODEM will be called snd-via82xx-modem. config SND_VIRTUOSO - tristate "Asus Virtuoso 100/200 (Xonar)" + tristate "Asus Virtuoso 66/100/200 (Xonar)" select SND_OXYGEN_LIB select SND_PCM select SND_MPU401_UART select SND_JACK if INPUT=y || INPUT=SND help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, + Asus AV66/AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, DS, Essence ST (Deluxe), and Essence STX. - Support for the DS is experimental. - Support for the HDAV1.3 (Deluxe) is very experimental. + Support for the HDAV1.3 (Deluxe) is incomplete; for the + HDAV1.3 Slim and Xense, missing. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. -- cgit v1.2.3 From b5786e85cb2ffd0b07e86dec38a442bd20765ad8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 14:21:17 +0200 Subject: ALSA: hda - Keep char arrays in input_mux items Keep char array in the input_mux item itself instead of pointing to an external string. This is a preliminary work for improving the input-mux name based on the pin role. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 18 ++++++------------ sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_local.h | 6 +++--- sound/pci/hda/patch_analog.c | 6 +++--- sound/pci/hda/patch_realtek.c | 10 +++++----- sound/pci/hda/patch_sigmatel.c | 16 ++++++++-------- sound/pci/hda/patch_via.c | 15 ++++++++------- 7 files changed, 34 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index bfdde7b0bafb..4348c33c6b85 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4662,17 +4662,8 @@ const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { }; EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); -static const char *input_labels[AUTO_PIN_LAST][4] = { - { "Mic", "Mic 2", "Mic 3", "Mic 4" }, - { "Front Mic", "Front Mic 2", "Front Mic 3", "Front Mic 4" }, - { "Line", "Line 2", "Line 3", "Line 4" }, - { "Front Line", "Front Line 2", "Front Line 3", "Front Line 4" }, - { "CD", "CD 2", "CD 3", "CD 4" }, - { "Aux", "Aux 2", "Aux 3", "Aux 4" }, -}; - -const char *snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, - int input) +void snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, + int input, char *str) { int type = cfg->inputs[input].type; int idx; @@ -4681,7 +4672,10 @@ const char *snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, if (type != cfg->inputs[input].type) break; } - return input_labels[type][idx]; + if (idx > 0) + sprintf(str, "%s %d", auto_pin_cfg_labels[type], idx); + else + strcpy(str, auto_pin_cfg_labels[type]); } EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_label); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5ea21285ee1f..cce18ba8b5a1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -566,7 +566,7 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, } label = spec->cap_labels[spec->input_mux.num_items]; strcpy(label, type); - spec->input_mux.items[spec->input_mux.num_items].label = label; + strcpy(spec->input_mux.items[spec->input_mux.num_items].label, label); /* unmute the PIN external input */ unmute_input(codec, node, 0); /* index = 0? */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index fb561748adb8..b448b0a997b1 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -215,7 +215,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid); */ #define HDA_MAX_NUM_INPUTS 16 struct hda_input_mux_item { - const char *label; + char label[32]; unsigned int index; }; struct hda_input_mux { @@ -391,8 +391,8 @@ struct auto_pin_cfg_item { }; struct auto_pin_cfg; -const char *snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, - int input); +void snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, + int input, char *label); struct auto_pin_cfg { int line_outs; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3409d315f507..8de3a0dc45e4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2926,13 +2926,13 @@ static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, type <= AUTO_PIN_FRONT_MIC); if (err < 0) return err; - imux->items[imux->num_items].label = - snd_hda_get_input_pin_label(cfg, i); + snd_hda_get_input_pin_label(cfg, i, + imux->items[imux->num_items].label); imux->items[imux->num_items].index = ad1988_pin_to_adc_idx(cfg->inputs[i].pin); imux->num_items++; } - imux->items[imux->num_items].label = "Mix"; + strcpy(imux->items[imux->num_items].label, "Mix"); imux->items[imux->num_items].index = 9; imux->num_items++; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c25d22be875..0a7d9d5ea40e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4974,8 +4974,8 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, if (idx < 0 && cap2) idx = get_connection_index(codec, cap2, pin); if (idx >= 0) { - imux->items[imux->num_items].label = - snd_hda_get_input_pin_label(cfg, i); + snd_hda_get_input_pin_label(cfg, i, + imux->items[imux->num_items].label); imux->items[imux->num_items].index = idx; imux->num_items++; } @@ -10626,9 +10626,9 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) break; nid = cfg->inputs[i].pin; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { - char label[32]; - snprintf(label, sizeof(label), "%s Boost", - snd_hda_get_input_pin_label(cfg, i)); + char pinname[32], label[32]; + snd_hda_get_input_pin_label(cfg, i, pinname); + snprintf(label, sizeof(label), "%s Boost", pinname); err = add_control(spec, ALC_CTL_WIDGET_VOL, label, 0, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7f09e140953e..852dae91edb1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1116,7 +1116,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) struct hda_input_mux *smux = &spec->private_smux; /* check for mute support on SPDIF out */ if (wcaps & AC_WCAP_OUT_AMP) { - smux->items[smux->num_items].label = "Off"; + strcpy(smux->items[smux->num_items].label, "Off"); smux->items[smux->num_items].index = 0; smux->num_items++; spec->spdif_mute = 1; @@ -3274,8 +3274,8 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) return -EINVAL; for (i = 0; i < num_cons; i++) { - mono_mux->items[mono_mux->num_items].label = - stac92xx_mono_labels[i]; + strcpy(mono_mux->items[mono_mux->num_items].label, + stac92xx_mono_labels[i]); mono_mux->items[mono_mux->num_items].index = i; mono_mux->num_items++; } @@ -3404,7 +3404,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) labels = stac92xx_spdif_labels; for (i = 0; i < num_cons; i++) { - spdif_mux->items[spdif_mux->num_items].label = labels[i]; + strcpy(spdif_mux->items[spdif_mux->num_items].label, labels[i]); spdif_mux->items[spdif_mux->num_items].index = i; spdif_mux->num_items++; } @@ -3538,7 +3538,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, int err, i; unsigned int def_conf; - dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; + strcpy(dimux->items[dimux->num_items].label, stac92xx_dmic_labels[0]); dimux->items[dimux->num_items].index = 0; dimux->num_items++; @@ -3572,11 +3572,11 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, return err; } - dimux->items[dimux->num_items].label = label; + strcpy(dimux->items[dimux->num_items].label, label); dimux->items[dimux->num_items].index = index; dimux->num_items++; if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) { - imux->items[imux->num_items].label = label; + strcpy(imux->items[imux->num_items].label, label); imux->items[imux->num_items].index = index; imux->num_items++; } @@ -3713,7 +3713,7 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const if (err < 0) return err; - imux->items[imux->num_items].label = label; + strcpy(imux->items[imux->num_items].label, label); imux->items[imux->num_items].index = index; imux->num_items++; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 93b86adbce63..9c1909d398e3 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2376,7 +2376,7 @@ static void create_hp_imux(struct via_spec *spec) /* for hp mode select */ i = 0; while (texts[i] != NULL) { - imux->items[imux->num_items].label = texts[i]; + strcpy(imux->items[imux->num_items].label, texts[i]); imux->items[imux->num_items].index = i; imux->num_items++; i++; @@ -2423,7 +2423,8 @@ static int vt_auto_create_analog_input_ctls(struct via_spec *spec, /* for internal loopback recording select */ for (idx = 0; idx < num_idxs; idx++) { if (pin_idxs[idx] == 0xff) { - imux->items[imux->num_items].label = "Stereo Mixer"; + strcpy(imux->items[imux->num_items].label, + "Stereo Mixer"); imux->items[imux->num_items].index = idx; imux->num_items++; break; @@ -2445,8 +2446,8 @@ static int vt_auto_create_analog_input_ctls(struct via_spec *spec, type_idx, idx, cap_nid); if (err < 0) return err; - imux->items[imux->num_items].label = - snd_hda_get_input_pin_label(cfg, i); + snd_hda_get_input_pin_label(cfg, i, + imux->items[imux->num_items].label); imux->items[imux->num_items].index = idx; imux->num_items++; } @@ -4336,7 +4337,7 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) /* for hp mode select */ i = 0; while (texts[i] != NULL) { - imux->items[imux->num_items].label = texts[i]; + strcpy(imux->items[imux->num_items].label, texts[i]); imux->items[imux->num_items].index = i; imux->num_items++; i++; @@ -5520,7 +5521,7 @@ static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, return err; /* for digital mic select */ - imux->items[imux->num_items].label = "Digital Mic"; + strcpy(imux->items[imux->num_items].label, "Digital Mic"); imux->items[imux->num_items].index = 4; imux->num_items++; @@ -5843,7 +5844,7 @@ static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, return err; /* for digital mic select */ - imux->items[imux->num_items].label = "Digital Mic"; + strcpy(imux->items[imux->num_items].label, "Digital Mic"); imux->items[imux->num_items].index = 6; imux->num_items++; -- cgit v1.2.3 From 86e2959a10828dd2614e037fb2502bc833adca52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 14:50:17 +0200 Subject: ALSA: hda - Remove AUTO_PIN_FRONT_{MIC|LINE} We can assign multiple pins to a single role now, let's reduce the redundant FRONT_MIC and FRONT_LINE. Also, autocfg->input_pins[] is no longer used, so this is removed as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 50 +++++------------------------------------- sound/pci/hda/hda_local.h | 5 +---- sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_cirrus.c | 6 ++--- sound/pci/hda/patch_realtek.c | 8 +++---- sound/pci/hda/patch_sigmatel.c | 20 ++++++----------- sound/pci/hda/patch_via.c | 16 +++++++------- 7 files changed, 31 insertions(+), 78 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4348c33c6b85..0ee4439c68ca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4396,7 +4396,7 @@ static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. * - * The analog input pins are assigned to input_pins array. + * The analog input pins are assigned to inputs array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. */ @@ -4480,39 +4480,16 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, sequences_hp[cfg->hp_outs] = (assoc << 4) | seq; cfg->hp_outs++; break; - case AC_JACK_MIC_IN: { - int preferred, alt; - if (loc == AC_JACK_LOC_FRONT || - (loc & 0x30) == AC_JACK_LOC_INTERNAL) { - preferred = AUTO_PIN_FRONT_MIC; - alt = AUTO_PIN_MIC; - } else { - preferred = AUTO_PIN_MIC; - alt = AUTO_PIN_FRONT_MIC; - } - if (!cfg->input_pins[preferred]) - cfg->input_pins[preferred] = nid; - else if (!cfg->input_pins[alt]) - cfg->input_pins[alt] = nid; - add_auto_cfg_input_pin(cfg, nid, preferred); + case AC_JACK_MIC_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_MIC); break; - } - case AC_JACK_LINE_IN: { - int type; - if (loc == AC_JACK_LOC_FRONT) - type = AUTO_PIN_FRONT_LINE; - else - type = AUTO_PIN_LINE; - cfg->input_pins[type] = nid; - add_auto_cfg_input_pin(cfg, nid, type); + case AC_JACK_LINE_IN: + add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_LINE_IN); break; - } case AC_JACK_CD: - cfg->input_pins[AUTO_PIN_CD] = nid; add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_CD); break; case AC_JACK_AUX: - cfg->input_pins[AUTO_PIN_AUX] = nid; add_auto_cfg_input_pin(cfg, nid, AUTO_PIN_AUX); break; case AC_JACK_SPDIF_OUT: @@ -4570,21 +4547,6 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - /* if we have only one mic, make it AUTO_PIN_MIC */ - if (!cfg->input_pins[AUTO_PIN_MIC] && - cfg->input_pins[AUTO_PIN_FRONT_MIC]) { - cfg->input_pins[AUTO_PIN_MIC] = - cfg->input_pins[AUTO_PIN_FRONT_MIC]; - cfg->input_pins[AUTO_PIN_FRONT_MIC] = 0; - } - /* ditto for line-in */ - if (!cfg->input_pins[AUTO_PIN_LINE] && - cfg->input_pins[AUTO_PIN_FRONT_LINE]) { - cfg->input_pins[AUTO_PIN_LINE] = - cfg->input_pins[AUTO_PIN_FRONT_LINE]; - cfg->input_pins[AUTO_PIN_FRONT_LINE] = 0; - } - /* * FIX-UP: if no line-outs are detected, try to use speaker or HP pin * as a primary output @@ -4658,7 +4620,7 @@ EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); /* labels for input pins - for obsoleted config stuff */ const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { - "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux" + "Mic", "Line", "CD", "Aux" }; EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index b448b0a997b1..72e7b2f210ee 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -366,9 +366,7 @@ struct hda_bus_unsolicited { enum { AUTO_PIN_MIC, - AUTO_PIN_FRONT_MIC, - AUTO_PIN_LINE, - AUTO_PIN_FRONT_LINE, + AUTO_PIN_LINE_IN, AUTO_PIN_CD, AUTO_PIN_AUX, AUTO_PIN_LAST @@ -403,7 +401,6 @@ struct auto_pin_cfg { int hp_outs; int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; - hda_nid_t input_pins[AUTO_PIN_LAST]; /* old config; to be deprecated */ int num_inputs; struct auto_pin_cfg_item inputs[AUTO_CFG_MAX_INS]; int dig_outs; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8de3a0dc45e4..85fc0b954603 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2923,7 +2923,7 @@ static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, type_idx = 0; err = new_analog_input(spec, cfg->inputs[i].pin, auto_pin_cfg_labels[type], type_idx, - type <= AUTO_PIN_FRONT_MIC); + type == AUTO_PIN_MIC); if (err < 0) return err; snd_hda_get_input_pin_label(cfg, i, @@ -3015,7 +3015,7 @@ static void ad1988_auto_init_analog_input(struct hda_codec *codec) break; } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? PIN_VREF80 : PIN_IN); + i == AUTO_PIN_MIC ? PIN_VREF80 : PIN_IN); if (nid != AD1988_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 6adfc5625281..adb5ec50252a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -436,8 +436,8 @@ static int parse_input(struct hda_codec *codec) /* check whether the automatic mic switch is available */ if (spec->num_inputs == 2 && - cfg->inputs[0].type <= AUTO_PIN_FRONT_MIC && - cfg->inputs[1].type == AUTO_PIN_FRONT_MIC) { + cfg->inputs[0].type == AUTO_PIN_MIC && + cfg->inputs[1].type == AUTO_PIN_MIC) { if (is_ext_mic(codec, cfg->inputs[0].pin)) { if (!is_ext_mic(codec, cfg->inputs[1].pin)) { spec->mic_detect = 1; @@ -921,7 +921,7 @@ static void init_input(struct hda_codec *codec) continue; /* set appropriate pin control and mute first */ ctl = PIN_IN; - if (cfg->inputs[i].type <= AUTO_PIN_FRONT_MIC) { + if (cfg->inputs[i].type == AUTO_PIN_MIC) { unsigned int caps = snd_hda_query_pin_caps(codec, pin); caps >>= AC_PINCAP_VREF_SHIFT; if (caps & AC_PINCAP_VREF_80) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0a7d9d5ea40e..8ae30ccf537a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -846,7 +846,7 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, { unsigned int val = PIN_IN; - if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { + if (auto_pin_type == AUTO_PIN_MIC) { unsigned int pincap; unsigned int oldval; oldval = snd_hda_codec_read(codec, nid, 0, @@ -1298,7 +1298,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) /* there must be only two mic inputs exclusively */ for (i = 0; i < cfg->num_inputs; i++) - if (cfg->inputs[i].type >= AUTO_PIN_LINE) + if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN) return; fixed = ext = 0; @@ -10622,7 +10622,7 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) hda_nid_t nid; for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type > AUTO_PIN_FRONT_MIC) + if (cfg->inputs[i].type > AUTO_PIN_MIC) break; nid = cfg->inputs[i].pin; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { @@ -19270,7 +19270,7 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.inputs[0].pin = 0x18; spec->autocfg.inputs[0].type = AUTO_PIN_MIC; spec->autocfg.inputs[1].pin = 0x19; - spec->autocfg.inputs[1].type = AUTO_PIN_LINE; + spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; } static void alc680_unsol_event(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 852dae91edb1..d9c8b4d335d2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2834,7 +2834,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type == AUTO_PIN_LINE) { + if (cfg->inputs[i].type == AUTO_PIN_LINE_IN) { nid = cfg->inputs[i].pin; pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) @@ -2852,16 +2852,14 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int def_conf, pincap; - int i, mic_type; + int i; *dac = 0; if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; - mic_type = AUTO_PIN_MIC; - again: for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; - if (cfg->inputs[i].type != mic_type) + if (cfg->inputs[i].type != AUTO_PIN_MIC) continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone @@ -2875,10 +2873,6 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) } } } - if (mic_type == AUTO_PIN_MIC) { - mic_type = AUTO_PIN_FRONT_MIC; - goto again; - } return 0; } @@ -3222,7 +3216,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } for (idx = 0; idx < cfg->num_inputs; idx++) { - if (cfg->inputs[idx].type > AUTO_PIN_FRONT_LINE) + if (cfg->inputs[idx].type > AUTO_PIN_LINE_IN) break; nid = cfg->inputs[idx].pin; err = stac92xx_add_jack_mode_control(codec, nid, idx); @@ -3621,7 +3615,7 @@ static int set_mic_route(struct hda_codec *codec, if (pin == cfg->inputs[i].pin) break; } - if (i < cfg->num_inputs && cfg->inputs[i].type <= AUTO_PIN_FRONT_MIC) { + if (i < cfg->num_inputs && cfg->inputs[i].type == AUTO_PIN_MIC) { /* analog pin */ i = get_connection_index(codec, spec->mux_nids[0], pin); if (i < 0) @@ -3656,7 +3650,7 @@ static int stac_check_auto_mic(struct hda_codec *codec) int i; for (i = 0; i < cfg->num_inputs; i++) { - if (cfg->inputs[i].type >= AUTO_PIN_LINE) + if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN) return 0; /* must be exclusively mics */ } fixed = ext = 0; @@ -4394,7 +4388,7 @@ static int stac92xx_init(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; int type = cfg->inputs[i].type; unsigned int pinctl, conf; - if (type == AUTO_PIN_MIC || type == AUTO_PIN_FRONT_MIC) { + if (type == AUTO_PIN_MIC) { /* for mic pins, force to initialize */ pinctl = stac92xx_get_default_vref(codec, nid); pinctl |= AC_PINCTL_IN_EN; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9c1909d398e3..de5f61d1b725 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -568,7 +568,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; if (spec->smart51_enabled && is_smart51_pins(spec, nid)) ctl = PIN_OUT; - else if (i <= AUTO_PIN_FRONT_MIC) + else if (i == AUTO_PIN_MIC) ctl = PIN_VREF50; else ctl = PIN_IN; @@ -1328,7 +1328,7 @@ static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) for (i = 0; i < cfg->num_inputs; i++) { if (pin == cfg->inputs[i].pin) - return cfg->inputs[i].type < AUTO_PIN_FRONT_LINE; + return cfg->inputs[i].type <= AUTO_PIN_LINE_IN; } return 0; } @@ -1356,9 +1356,9 @@ static int via_smart51_get(struct snd_kcontrol *kcontrol, hda_nid_t nid = cfg->inputs[i].pin; int ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (cfg->inputs[i].type >= AUTO_PIN_FRONT_LINE) + if (cfg->inputs[i].type > AUTO_PIN_LINE_IN) continue; - if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC && + if (cfg->inputs[i].type == AUTO_PIN_MIC && spec->hp_independent_mode && spec->codec_type != VT1718S) continue; /* ignore FMic for independent HP */ if ((ctl & AC_PINCTL_IN_EN) && !(ctl & AC_PINCTL_OUT_EN)) @@ -1382,9 +1382,9 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, hda_nid_t nid = cfg->inputs[i].pin; unsigned int parm; - if (cfg->inputs[i].type >= AUTO_PIN_FRONT_LINE) + if (cfg->inputs[i].type > AUTO_PIN_LINE_IN) continue; - if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC && + if (cfg->inputs[i].type == AUTO_PIN_MIC && spec->hp_independent_mode && spec->codec_type != VT1718S) continue; /* don't retask FMic for independent HP */ @@ -1404,7 +1404,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); } - if (cfg->inputs[i].type == AUTO_PIN_FRONT_MIC) { + if (cfg->inputs[i].type == AUTO_PIN_MIC) { if (spec->codec_type == VT1708S || spec->codec_type == VT1716S) { /* input = index 1 (AOW3) */ @@ -1450,7 +1450,7 @@ static int via_smart51_build(struct via_spec *spec) for (i = 0; i < cfg->num_inputs; i++) { nid = cfg->inputs[i].pin; - if (cfg->inputs[i].type < AUTO_PIN_FRONT_LINE) { + if (cfg->inputs[i].type <= AUTO_PIN_LINE_IN) { knew = via_clone_control(spec, &via_smart51_mixer[1]); if (knew == NULL) return -ENOMEM; -- cgit v1.2.3 From 10a20af7c944649dc6d1ffa06bc759f5f3a16cd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 16:28:02 +0200 Subject: ALSA: hda - Improve the input source name labels This patch improves the input-source label strings to be generated from the pin information instead of fixed strings per AUTO_PIN_* type. This gives more suitable labels, especially for mic and line-in pins. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 91 +++++++++++++++++++++----- sound/pci/hda/hda_generic.c | 41 ++---------- sound/pci/hda/hda_local.h | 11 ++-- sound/pci/hda/patch_analog.c | 27 ++++---- sound/pci/hda/patch_ca0110.c | 3 +- sound/pci/hda/patch_cirrus.c | 4 +- sound/pci/hda/patch_realtek.c | 20 +++--- sound/pci/hda/patch_sigmatel.c | 144 ++++++----------------------------------- sound/pci/hda/patch_via.c | 100 ++++++++++++---------------- 9 files changed, 174 insertions(+), 267 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0ee4439c68ca..e3284341c484 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4607,7 +4607,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, snd_printd(" inputs:"); for (i = 0; i < cfg->num_inputs; i++) { snd_printdd(" %s=0x%x", - auto_pin_cfg_labels[cfg->inputs[i].type], + hda_get_autocfg_input_label(codec, cfg, i), cfg->inputs[i].pin); } snd_printd("\n"); @@ -4618,28 +4618,87 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); -/* labels for input pins - for obsoleted config stuff */ -const char *auto_pin_cfg_labels[AUTO_PIN_LAST] = { - "Mic", "Line", "CD", "Aux" -}; -EXPORT_SYMBOL_HDA(auto_pin_cfg_labels); +const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, + int check_location) +{ + unsigned int def_conf, loc; + + def_conf = snd_hda_codec_get_pincfg(codec, pin); + loc = get_defcfg_location(def_conf); + + switch (get_defcfg_device(def_conf)) { + case AC_JACK_MIC_IN: + if (!check_location) + return "Mic"; + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || + (loc & 0x30) == AC_JACK_LOC_INTERNAL) + return "Internal Mic"; + if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) + return "Dock Mic"; + if (loc == AC_JACK_LOC_REAR) + return "Rear Mic"; + return "Mic"; + case AC_JACK_LINE_IN: + if (!check_location) + return "Line"; + if ((loc & 0xf0) == AC_JACK_LOC_SEPARATE) + return "Dock Line"; + return "Line"; + case AC_JACK_AUX: + return "Aux"; + case AC_JACK_CD: + return "CD"; + case AC_JACK_SPDIF_IN: + return "SPDIF In"; + case AC_JACK_DIG_OTHER_IN: + return "Digital In"; + default: + return "Misc"; + } +} +EXPORT_SYMBOL_HDA(hda_get_input_pin_label); -void snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, - int input, char *str) +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) { int type = cfg->inputs[input].type; - int idx; + int has_multiple_pins = 0; - for (idx = 0; idx < 3 && --input >= 0; idx++) { - if (type != cfg->inputs[input].type) - break; + if ((input > 0 && cfg->inputs[input - 1].type == type) || + (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) + has_multiple_pins = 1; + return hda_get_input_pin_label(codec, cfg->inputs[input].pin, + has_multiple_pins); +} +EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); + +int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, + int index, int *type_idx) +{ + int i, label_idx = 0; + if (imux->num_items >= HDA_MAX_NUM_INPUTS) { + snd_printd(KERN_ERR "hda_codec: Too many imux items!\n"); + return -EINVAL; + } + for (i = 0; i < imux->num_items; i++) { + if (!strncmp(label, imux->items[i].label, strlen(label))) + label_idx++; } - if (idx > 0) - sprintf(str, "%s %d", auto_pin_cfg_labels[type], idx); + if (type_idx) + *type_idx = label_idx; + if (label_idx > 0) + snprintf(imux->items[imux->num_items].label, + sizeof(imux->items[imux->num_items].label), + "%s %d", label, label_idx); else - strcpy(str, auto_pin_cfg_labels[type]); + strlcpy(imux->items[imux->num_items].label, label, + sizeof(imux->items[imux->num_items].label)); + imux->items[imux->num_items].index = index; + imux->num_items++; + return 0; } -EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_label); +EXPORT_SYMBOL_HDA(snd_hda_add_imux_item); #ifdef CONFIG_PM diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index cce18ba8b5a1..fb0582f8d725 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -61,7 +61,6 @@ struct hda_gspec { struct hda_gnode *cap_vol_node; /* Node for capture volume */ unsigned int cur_cap_src; /* current capture source */ struct hda_input_mux input_mux; - char cap_labels[HDA_MAX_NUM_INPUTS][16]; unsigned int def_amp_in_caps; unsigned int def_amp_out_caps; @@ -506,11 +505,10 @@ static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl) * returns 0 if not found, 1 if found, or a negative error code. */ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, - struct hda_gnode *node) + struct hda_gnode *node, int idx) { int i, err; unsigned int pinctl; - char *label; const char *type; if (node->checked) @@ -523,7 +521,7 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, child = hda_get_node(spec, node->conn_list[i]); if (! child) continue; - err = parse_adc_sub_nodes(codec, spec, child); + err = parse_adc_sub_nodes(codec, spec, child, idx); if (err < 0) return err; if (err > 0) { @@ -564,9 +562,7 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, return 0; type = "Input"; } - label = spec->cap_labels[spec->input_mux.num_items]; - strcpy(label, type); - strcpy(spec->input_mux.items[spec->input_mux.num_items].label, label); + snd_hda_add_imux_item(&spec->input_mux, type, idx, NULL); /* unmute the PIN external input */ unmute_input(codec, node, 0); /* index = 0? */ @@ -577,29 +573,6 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, return 1; /* found */ } -/* add a capture source element */ -static void add_cap_src(struct hda_gspec *spec, int idx) -{ - struct hda_input_mux_item *csrc; - char *buf; - int num, ocap; - - num = spec->input_mux.num_items; - csrc = &spec->input_mux.items[num]; - buf = spec->cap_labels[num]; - for (ocap = 0; ocap < num; ocap++) { - if (! strcmp(buf, spec->cap_labels[ocap])) { - /* same label already exists, - * put the index number to be unique - */ - sprintf(buf, "%s %d", spec->cap_labels[ocap], num); - break; - } - } - csrc->index = idx; - spec->input_mux.num_items++; -} - /* * parse input */ @@ -624,22 +597,18 @@ static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node) for (i = 0; i < adc_node->nconns; i++) { node = hda_get_node(spec, adc_node->conn_list[i]); if (node && node->type == AC_WID_PIN) { - err = parse_adc_sub_nodes(codec, spec, node); + err = parse_adc_sub_nodes(codec, spec, node, i); if (err < 0) return err; - else if (err > 0) - add_cap_src(spec, i); } } /* ... then check the rests, more complicated connections */ for (i = 0; i < adc_node->nconns; i++) { node = hda_get_node(spec, adc_node->conn_list[i]); if (node && node->type != AC_WID_PIN) { - err = parse_adc_sub_nodes(codec, spec, node); + err = parse_adc_sub_nodes(codec, spec, node, i); if (err < 0) return err; - else if (err > 0) - add_cap_src(spec, i); } } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 72e7b2f210ee..6943efc78f66 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -378,8 +378,6 @@ enum { AUTO_PIN_HP_OUT }; -extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST]; - #define AUTO_CFG_MAX_OUTS 5 #define AUTO_CFG_MAX_INS 8 @@ -389,8 +387,13 @@ struct auto_pin_cfg_item { }; struct auto_pin_cfg; -void snd_hda_get_input_pin_label(const struct auto_pin_cfg *cfg, - int input, char *label); +const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, + int check_location); +const char *hda_get_autocfg_input_label(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input); +int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, + int index, int *type_index_ret); struct auto_pin_cfg { int line_outs; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 85fc0b954603..05db1cfcd8b8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2909,32 +2909,27 @@ static int new_analog_input(struct ad198x_spec *spec, hda_nid_t pin, } /* create playback/capture controls for input pins */ -static int ad1988_auto_create_analog_input_ctls(struct ad198x_spec *spec, +static int ad1988_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct ad198x_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; - int i, err, type, type_idx = 0; + int i, err, type, type_idx; for (i = 0; i < cfg->num_inputs; i++) { + const char *label; type = cfg->inputs[i].type; - if (i > 0 && type != cfg->inputs[i - 1].type) - type_idx++; - else - type_idx = 0; + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, + ad1988_pin_to_adc_idx(cfg->inputs[i].pin), + &type_idx); err = new_analog_input(spec, cfg->inputs[i].pin, - auto_pin_cfg_labels[type], type_idx, + label, type_idx, type == AUTO_PIN_MIC); if (err < 0) return err; - snd_hda_get_input_pin_label(cfg, i, - imux->items[imux->num_items].label); - imux->items[imux->num_items].index = - ad1988_pin_to_adc_idx(cfg->inputs[i].pin); - imux->num_items++; } - strcpy(imux->items[imux->num_items].label, "Mix"); - imux->items[imux->num_items].index = 9; - imux->num_items++; + snd_hda_add_imux_item(imux, "Mix", 9, NULL); if ((err = add_control(spec, AD_CTL_WIDGET_VOL, "Analog Mix Playback Volume", @@ -3046,7 +3041,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) "Speaker")) < 0 || (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || - (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) + (err = ad1988_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 42b3fb4cafc4..cca11fdd3d79 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -474,8 +474,7 @@ static void parse_input(struct hda_codec *codec) if (j >= cfg->num_inputs) continue; spec->input_pins[n] = pin; - spec->input_labels[n] = - auto_pin_cfg_labels[cfg->inputs[j].type]; + spec->input_labels[n] = hda_get_input_pin_label(codec, pin, 1); spec->adcs[n] = nid; n++; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index adb5ec50252a..ae75283a5583 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -673,6 +673,7 @@ static int cs_capture_source_info(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int idx; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -681,7 +682,8 @@ static int cs_capture_source_info(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item >= spec->num_inputs) uinfo->value.enumerated.item = spec->num_inputs - 1; idx = spec->input_idx[uinfo->value.enumerated.item]; - strcpy(uinfo->value.enumerated.name, auto_pin_cfg_labels[idx]); + strcpy(uinfo->value.enumerated.name, + hda_get_input_pin_label(codec, cfg->inputs[idx].pin, 1)); return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ae30ccf537a..9c2c19c8b059 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4947,6 +4947,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t pin; + const char *label; pin = cfg->inputs[i].pin; if (!alc_is_input_pin(codec, pin)) @@ -4957,12 +4958,13 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, type_idx++; else type_idx = 0; + label = hda_get_autocfg_input_label(codec, cfg, i); if (mixer) { idx = get_connection_index(codec, mixer, pin); if (idx >= 0) { err = new_analog_input(spec, pin, - auto_pin_cfg_labels[type], - type_idx, idx, mixer); + label, type_idx, + idx, mixer); if (err < 0) return err; } @@ -4973,12 +4975,8 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec, idx = get_connection_index(codec, cap1, pin); if (idx < 0 && cap2) idx = get_connection_index(codec, cap2, pin); - if (idx >= 0) { - snd_hda_get_input_pin_label(cfg, i, - imux->items[imux->num_items].label); - imux->items[imux->num_items].index = idx; - imux->num_items++; - } + if (idx >= 0) + snd_hda_add_imux_item(imux, label, idx, NULL); } return 0; } @@ -10626,9 +10624,9 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec) break; nid = cfg->inputs[i].pin; if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) { - char pinname[32], label[32]; - snd_hda_get_input_pin_label(cfg, i, pinname); - snprintf(label, sizeof(label), "%s Boost", pinname); + char label[32]; + snprintf(label, sizeof(label), "%s Boost", + hda_get_autocfg_input_label(codec, cfg, i)); err = add_control(spec, ALC_CTL_WIDGET_VOL, label, 0, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d9c8b4d335d2..e4e7d43f911e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -191,11 +191,6 @@ struct sigmatel_mic_route { signed char dmux_idx; }; -struct unique_input_names { - int num; - char uname[HDA_MAX_NUM_INPUTS][32]; -}; - struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -312,7 +307,6 @@ struct sigmatel_spec { struct hda_input_mux private_imux; struct hda_input_mux private_smux; struct hda_input_mux private_mono_mux; - struct unique_input_names private_u_inp_names; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -1116,9 +1110,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) struct hda_input_mux *smux = &spec->private_smux; /* check for mute support on SPDIF out */ if (wcaps & AC_WCAP_OUT_AMP) { - strcpy(smux->items[smux->num_items].label, "Off"); - smux->items[smux->num_items].index = 0; - smux->num_items++; + snd_hda_add_imux_item(smux, "Off", 0, NULL); spec->spdif_mute = 1; } stac_smux_mixer.count = spec->num_smuxes; @@ -2797,7 +2789,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, } if (control) { - strcpy(name, auto_pin_cfg_labels[idx]); + strcpy(name, hda_get_input_pin_label(codec, nid, 1)); return stac92xx_add_control(codec->spec, control, strcat(name, " Jack Mode"), nid); } @@ -3267,12 +3259,9 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) return -EINVAL; - for (i = 0; i < num_cons; i++) { - strcpy(mono_mux->items[mono_mux->num_items].label, - stac92xx_mono_labels[i]); - mono_mux->items[mono_mux->num_items].index = i; - mono_mux->num_items++; - } + for (i = 0; i < num_cons; i++) + snd_hda_add_imux_item(mono_mux, stac92xx_mono_labels[i], i, + NULL); return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, "Mono Mux", spec->mono_nid); @@ -3397,11 +3386,8 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) if (!labels) labels = stac92xx_spdif_labels; - for (i = 0; i < num_cons; i++) { - strcpy(spdif_mux->items[spdif_mux->num_items].label, labels[i]); - spdif_mux->items[spdif_mux->num_items].index = i; - spdif_mux->num_items++; - } + for (i = 0; i < num_cons; i++) + snd_hda_add_imux_item(spdif_mux, labels[i], i, NULL); return 0; } @@ -3452,76 +3438,6 @@ static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, return 1; } -static const char *get_input_src_label(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int def_conf; - - def_conf = snd_hda_codec_get_pincfg(codec, nid); - - switch (get_defcfg_device(def_conf)) { - case AC_JACK_MIC_IN: - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || - ((get_defcfg_location(def_conf) & 0xf0) - == AC_JACK_LOC_INTERNAL)) - return "Internal Mic"; - if ((get_defcfg_location(def_conf) & 0xf0) - == AC_JACK_LOC_SEPARATE) - return "Dock Mic"; - if (get_defcfg_location(def_conf) == AC_JACK_LOC_REAR) - return "Rear Mic"; - return "Mic"; - case AC_JACK_LINE_IN: - if ((get_defcfg_location(def_conf) & 0xf0) - == AC_JACK_LOC_SEPARATE) - return "Dock Line"; - return "Line"; - case AC_JACK_AUX: - return "Aux"; - case AC_JACK_CD: - return "CD"; - case AC_JACK_SPDIF_IN: - return "SPDIF In"; - case AC_JACK_DIG_OTHER_IN: - return "Digital In"; - } - - snd_printd("invalid inp pin %02x device config %08x", nid, def_conf); - return NULL; -} - -static const char *get_unique_inp_src_label(struct hda_codec *codec, - hda_nid_t nid) -{ - int i, n; - const char *label; - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux; - struct hda_input_mux *dimux = &spec->private_dimux; - struct unique_input_names *unames = &spec->private_u_inp_names; - - label = get_input_src_label(codec, nid); - n = 0; - - for (i = 0; i < imux->num_items; i++) { - if (!strncmp(label, imux->items[i].label, strlen(label))) - n++; - } - if (snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { - for (i = 0; i < dimux->num_items; i++) { - if (!strncmp(label, dimux->items[i].label, - strlen(label))) - n++; - } - } - if (n > 0 && unames->num < HDA_MAX_NUM_INPUTS) { - sprintf(&unames->uname[unames->num][0], "%.28s %d", label, n); - label = &unames->uname[unames->num][0]; - unames->num++; - } - - return label; -} - /* create playback/capture controls for input pins on dmic capable codecs */ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -3532,13 +3448,11 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, int err, i; unsigned int def_conf; - strcpy(dimux->items[dimux->num_items].label, stac92xx_dmic_labels[0]); - dimux->items[dimux->num_items].index = 0; - dimux->num_items++; + snd_hda_add_imux_item(dimux, stac92xx_dmic_labels[0], 0, NULL); for (i = 0; i < spec->num_dmics; i++) { hda_nid_t nid; - int index; + int index, type_idx; const char *label; nid = spec->dmic_nids[i]; @@ -3552,28 +3466,22 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (index < 0) continue; - label = get_unique_inp_src_label(codec, nid); - if (label == NULL) - return -EINVAL; + label = hda_get_input_pin_label(codec, nid, 1); + snd_hda_add_imux_item(dimux, label, index, &type_idx); - err = create_elem_capture_vol(codec, nid, label, 0, HDA_INPUT); + err = create_elem_capture_vol(codec, nid, label, type_idx, + HDA_INPUT); if (err < 0) return err; if (!err) { - err = create_elem_capture_vol(codec, nid, label, 0, - HDA_OUTPUT); + err = create_elem_capture_vol(codec, nid, label, + type_idx, HDA_OUTPUT); if (err < 0) return err; } - strcpy(dimux->items[dimux->num_items].label, label); - dimux->items[dimux->num_items].index = index; - dimux->num_items++; - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) { - strcpy(imux->items[imux->num_items].label, label); - imux->items[imux->num_items].index = index; - imux->num_items++; - } + if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) + snd_hda_add_imux_item(imux, label, index, NULL); } return 0; @@ -3675,12 +3583,12 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; - int i, j, type_idx = 0; + int i, j; const char *label; for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; - int index, err; + int index, err, type_idx; index = -1; for (j = 0; j < spec->num_muxes; j++) { @@ -3692,24 +3600,14 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const if (index < 0) continue; - if (i > 0 && cfg->inputs[i].type == cfg->inputs[i - 1].type) - type_idx++; - else - type_idx = 0; - - label = get_unique_inp_src_label(codec, nid); - if (label == NULL) - return -EINVAL; + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, index, &type_idx); err = create_elem_capture_vol(codec, nid, label, type_idx, HDA_INPUT); if (err < 0) return err; - - strcpy(imux->items[imux->num_items].label, label); - imux->items[imux->num_items].index = index; - imux->num_items++; } spec->num_analog_muxes = imux->num_items; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de5f61d1b725..d1c3f8defc48 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2374,13 +2374,8 @@ static void create_hp_imux(struct via_spec *spec) static const char *texts[] = { "OFF", "ON", NULL}; /* for hp mode select */ - i = 0; - while (texts[i] != NULL) { - strcpy(imux->items[imux->num_items].label, texts[i]); - imux->items[imux->num_items].index = i; - imux->num_items++; - i++; - } + for (i = 0; texts[i]; i++) + snd_hda_add_imux_item(imux, texts[i], i, NULL); spec->hp_mux = &spec->private_imux[1]; } @@ -2412,26 +2407,25 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg, hda_nid_t cap_nid, hda_nid_t pin_idxs[], int num_idxs) { + struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, type, type_idx = 0; /* for internal loopback recording select */ for (idx = 0; idx < num_idxs; idx++) { if (pin_idxs[idx] == 0xff) { - strcpy(imux->items[imux->num_items].label, - "Stereo Mixer"); - imux->items[imux->num_items].index = idx; - imux->num_items++; + snd_hda_add_imux_item(imux, "Stereo Mixer", idx, NULL); break; } } for (i = 0; i < cfg->num_inputs; i++) { + const char *label; type = cfg->inputs[i].type; for (idx = 0; idx < num_idxs; idx++) if (pin_idxs[idx] == cfg->inputs[i].pin) @@ -2442,24 +2436,21 @@ static int vt_auto_create_analog_input_ctls(struct via_spec *spec, type_idx++; else type_idx = 0; - err = via_new_analog_input(spec, auto_pin_cfg_labels[type], - type_idx, idx, cap_nid); + label = hda_get_autocfg_input_label(codec, cfg, i); + err = via_new_analog_input(spec, label, type_idx, idx, cap_nid); if (err < 0) return err; - snd_hda_get_input_pin_label(cfg, i, - imux->items[imux->num_items].label); - imux->items[imux->num_items].index = idx; - imux->num_items++; + snd_hda_add_imux_item(imux, label, idx, NULL); } return 0; } /* create playback/capture controls for input pins */ -static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1708_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0xff, 0x24, 0x1d, 0x1e, 0x21 }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x17, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x17, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -2559,7 +2550,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) err = vt1708_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1708_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; /* add jack detect on/off control */ @@ -3026,11 +3017,11 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1709_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0xff, 0x23, 0x1d, 0x1e, 0x21 }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x18, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x18, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -3054,7 +3045,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) err = vt1709_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1709_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1709_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -3556,11 +3547,11 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1708B_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0xff, 0x1f, 0x1a, 0x1b, 0x1e }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -3584,7 +3575,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) err = vt1708B_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1708B_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1708B_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -3992,11 +3983,11 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1708S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -4045,7 +4036,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1708S_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1708S_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4335,24 +4326,19 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) imux = &spec->private_imux[1]; /* for hp mode select */ - i = 0; - while (texts[i] != NULL) { - strcpy(imux->items[imux->num_items].label, texts[i]); - imux->items[imux->num_items].index = i; - imux->num_items++; - i++; - } + for (i = 0; texts[i]; i++) + snd_hda_add_imux_item(imux, texts[i], i, NULL); spec->hp_mux = &spec->private_imux[1]; return 0; } /* create playback/capture controls for input pins */ -static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1702_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0x14, 0x15, 0x18, 0xff }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x1a, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x1a, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -4382,7 +4368,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | (1 << AC_AMPCAP_MUTE_SHIFT)); - err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1702_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4733,11 +4719,11 @@ static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1718S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0x2c, 0x2b, 0x2a, 0x29, 0, 0xff }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -4762,7 +4748,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1718S_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -5195,11 +5181,11 @@ static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1716S_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { static hda_nid_t pin_idxs[] = { 0x1f, 0x1a, 0x1b, 0x1e, 0, 0xff }; - return vt_auto_create_analog_input_ctls(spec, cfg, 0x16, pin_idxs, + return vt_auto_create_analog_input_ctls(codec, cfg, 0x16, pin_idxs, ARRAY_SIZE(pin_idxs)); } @@ -5223,7 +5209,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1716S_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -5504,14 +5490,15 @@ static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt2002P_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0xff }; int err; - err = vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, ARRAY_SIZE(pin_idxs)); if (err < 0) return err; @@ -5521,9 +5508,7 @@ static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, return err; /* for digital mic select */ - strcpy(imux->items[imux->num_items].label, "Digital Mic"); - imux->items[imux->num_items].index = 4; - imux->num_items++; + snd_hda_add_imux_item(imux, "Digital Mic", 4, NULL); return 0; } @@ -5551,7 +5536,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt2002P_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -5826,14 +5811,15 @@ static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) } /* create playback/capture controls for input pins */ -static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, +static int vt1812_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct via_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; static hda_nid_t pin_idxs[] = { 0x2b, 0x2a, 0x29, 0, 0, 0xff }; int err; - err = vt_auto_create_analog_input_ctls(spec, cfg, 0x21, pin_idxs, + err = vt_auto_create_analog_input_ctls(codec, cfg, 0x21, pin_idxs, ARRAY_SIZE(pin_idxs)); if (err < 0) return err; @@ -5844,9 +5830,7 @@ static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, return err; /* for digital mic select */ - strcpy(imux->items[imux->num_items].label, "Digital Mic"); - imux->items[imux->num_items].index = 6; - imux->num_items++; + snd_hda_add_imux_item(imux, "Digital Mic", 6, NULL); return 0; } @@ -5874,7 +5858,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = vt1812_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.3 From a1c985158382cbce0b58b3264f771b3b153668a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 21:36:27 +0200 Subject: ALSA: hda - Reduce redundant mic location prefix in input source labels When the mic pins are assigned to the same location, we can omit the redundant location prefix like "Front" or "Rear". Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 90 ++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 77 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e3284341c484..affb4607c6da 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4618,32 +4618,64 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); +enum { + MIC_ATTR_INT, + MIC_ATTR_DOCK, + MIC_ATTR_NORMAL, + MIC_ATTR_FRONT, + MIC_ATTR_REAR, +}; + +static int get_mic_pin_attr(unsigned int def_conf) +{ + unsigned int loc = get_defcfg_location(def_conf); + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || + (loc & 0x30) == AC_JACK_LOC_INTERNAL) + return MIC_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) + return MIC_ATTR_DOCK; + if (loc == AC_JACK_LOC_REAR) + return MIC_ATTR_REAR; + if (loc == AC_JACK_LOC_FRONT) + return MIC_ATTR_FRONT; + return MIC_ATTR_NORMAL; +} + +enum { + LINE_ATTR_DOCK, + LINE_ATTR_NORMAL, +}; + +static int get_line_pin_attr(unsigned int def_conf) +{ + unsigned int loc = get_defcfg_location(def_conf); + if ((loc & 0xf0) == AC_JACK_LOC_SEPARATE) + return LINE_ATTR_DOCK; + return LINE_ATTR_NORMAL; +} + const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, int check_location) { - unsigned int def_conf, loc; + unsigned int def_conf; + static const char *mic_names[] = { + "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", + }; + static const char *line_names[] = { + "Dock Line", "Line", + }; def_conf = snd_hda_codec_get_pincfg(codec, pin); - loc = get_defcfg_location(def_conf); switch (get_defcfg_device(def_conf)) { case AC_JACK_MIC_IN: if (!check_location) return "Mic"; - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || - (loc & 0x30) == AC_JACK_LOC_INTERNAL) - return "Internal Mic"; - if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) - return "Dock Mic"; - if (loc == AC_JACK_LOC_REAR) - return "Rear Mic"; - return "Mic"; + return mic_names[get_mic_pin_attr(def_conf)]; case AC_JACK_LINE_IN: if (!check_location) return "Line"; - if ((loc & 0xf0) == AC_JACK_LOC_SEPARATE) - return "Dock Line"; - return "Line"; + return line_names[get_line_pin_attr(def_conf)]; case AC_JACK_AUX: return "Aux"; case AC_JACK_CD: @@ -4658,6 +4690,36 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, } EXPORT_SYMBOL_HDA(hda_get_input_pin_label); +/* Check whether the location prefix needs to be added to the label. + * If all mic-jacks are in the same location (e.g. rear panel), we don't + * have to put "Front" prefix to each label. In such a case, returns false. + */ +static int check_mic_location_need(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + int input) +{ + unsigned int defc; + int i, attr, attr2; + + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); + attr = get_mic_pin_attr(defc); + /* for internal or docking mics, we need locations */ + if (attr <= MIC_ATTR_NORMAL) + return 1; + + attr = 0; + for (i = 0; i < cfg->num_inputs; i++) { + defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); + attr2 = get_mic_pin_attr(defc); + if (attr2 >= MIC_ATTR_NORMAL) { + if (attr && attr != attr2) + return 1; /* different locations found */ + attr = attr2; + } + } + return 0; +} + const char *hda_get_autocfg_input_label(struct hda_codec *codec, const struct auto_pin_cfg *cfg, int input) @@ -4668,6 +4730,8 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, if ((input > 0 && cfg->inputs[input - 1].type == type) || (input < cfg->num_inputs - 1 && cfg->inputs[input + 1].type == type)) has_multiple_pins = 1; + if (has_multiple_pins && type == AUTO_PIN_MIC) + has_multiple_pins &= check_mic_location_need(codec, cfg, input); return hda_get_input_pin_label(codec, cfg->inputs[input].pin, has_multiple_pins); } -- cgit v1.2.3 From 990061c28ab6c84e1120afb772b69d92d8965da8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 22:08:44 +0200 Subject: ALSA: hda - Add comments to new helper functions Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index affb4607c6da..ec38bdfad81e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4654,6 +4654,14 @@ static int get_line_pin_attr(unsigned int def_conf) return LINE_ATTR_NORMAL; } +/** + * hda_get_input_pin_label - Give a label for the given input pin + * + * When check_location is true, the function checks the pin location + * for mic and line-in pins, and set an appropriate prefix like "Front", + * "Rear", "Internal". + */ + const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, int check_location) { @@ -4720,6 +4728,14 @@ static int check_mic_location_need(struct hda_codec *codec, return 0; } +/** + * hda_get_autocfg_input_label - Get a label for the given input + * + * Get a label for the given input pin defined by the autocfg item. + * Unlike hda_get_input_pin_label(), this function checks all inputs + * defined in autocfg and avoids the redundant mic/line prefix as much as + * possible. + */ const char *hda_get_autocfg_input_label(struct hda_codec *codec, const struct auto_pin_cfg *cfg, int input) @@ -4737,6 +4753,13 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(hda_get_autocfg_input_label); +/** + * snd_hda_add_imux_item - Add an item to input_mux + * + * When the same label is used already in the existing items, the number + * suffix is appended to the label. This label index number is stored + * to type_idx when non-NULL pointer is given. + */ int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, int index, int *type_idx) { -- cgit v1.2.3 From 4a4d4a6985dd37a3c96534027f054be796bf95f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Sep 2010 22:22:02 +0200 Subject: ALSA: hda - Sort input pins in snd_hda_parse_pin_def_config() Sort inputs[] array in autocfg so that the codec parsers can filter out easily per input pin types. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec38bdfad81e..08d81b873022 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4383,6 +4383,23 @@ static void add_auto_cfg_input_pin(struct auto_pin_cfg *cfg, hda_nid_t nid, } } +/* sort inputs in the order of AUTO_PIN_* type */ +static void sort_autocfg_input_pins(struct auto_pin_cfg *cfg) +{ + int i, j; + + for (i = 0; i < cfg->num_inputs; i++) { + for (j = i + 1; j < cfg->num_inputs; j++) { + if (cfg->inputs[i].type > cfg->inputs[j].type) { + struct auto_pin_cfg_item tmp; + tmp = cfg->inputs[i]; + cfg->inputs[i] = cfg->inputs[j]; + cfg->inputs[j] = tmp; + } + } + } +} + /* * Parse all pin widgets and store the useful pin nids to cfg * @@ -4585,6 +4602,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, break; } + sort_autocfg_input_pins(cfg); + /* * debug prints of the parsed results */ -- cgit v1.2.3 From 6008fd5aa4c15f2ea80a9f997983a9cbfa14ba73 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 10 Sep 2010 16:12:34 +0800 Subject: ALSA: snd-usb-caiaq: drop version number Let git do the job. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index cdfb856bddd2..da9cb6dcee2a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -36,7 +36,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.21"); +MODULE_DESCRIPTION("caiaq USB audio"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v1.2.3 From 15c5ab607045e278ebf4d2ca4aea2250617d50ca Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 10 Sep 2010 17:04:57 +0800 Subject: ALSA: snd-usb-caiaq: Add support for Traktor Kontrol S4 This patch adds support for the new Traktor Kontrol S4 by Native Instruments. It features a new audio data streaming model, MIDI in and out ports, a huge number of 174 dimmable LEDs, 96 buttons and 46 absolute encoder axis, including some rotary encoders. All features are supported by the driver now. Did some code refactoring along the way. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 2 + sound/usb/caiaq/audio.c | 175 +++++++++++++++++++++++++++++--- sound/usb/caiaq/control.c | 208 +++++++++++++++++++++++++++++++++++++- sound/usb/caiaq/device.c | 8 +- sound/usb/caiaq/device.h | 6 +- sound/usb/caiaq/input.c | 248 +++++++++++++++++++++++++++++++++++++++++----- 6 files changed, 598 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 44d6d2ec964f..112984f4080f 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -65,6 +65,7 @@ config SND_USB_CAIAQ * Native Instruments Guitar Rig Session I/O * Native Instruments Guitar Rig mobile * Native Instruments Traktor Kontrol X1 + * Native Instruments Traktor Kontrol S4 To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. @@ -82,6 +83,7 @@ config SND_USB_CAIAQ_INPUT * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Traktor Kontrol S4 config SND_USB_US122L tristate "Tascam US-122L USB driver" diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4328cad6c3a2..68b97477577b 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -111,7 +111,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) memset(dev->sub_capture, 0, sizeof(dev->sub_capture)); dev->input_panic = 0; dev->output_panic = 0; - dev->first_packet = 1; + dev->first_packet = 4; dev->streaming = 1; dev->warned = 0; @@ -169,7 +169,7 @@ static int snd_usb_caiaq_substream_close(struct snd_pcm_substream *substream) } static int snd_usb_caiaq_pcm_hw_params(struct snd_pcm_substream *sub, - struct snd_pcm_hw_params *hw_params) + struct snd_pcm_hw_params *hw_params) { debug("%s(%p)\n", __func__, sub); return snd_pcm_lib_malloc_pages(sub, params_buffer_bytes(hw_params)); @@ -189,7 +189,7 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000 }; static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { @@ -201,12 +201,39 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) debug("%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; - dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; + int out_pos; + + switch (dev->spec.data_alignment) { + case 0: + case 2: + out_pos = BYTES_PER_SAMPLE + 1; + break; + case 3: + default: + out_pos = 0; + break; + } + + dev->period_out_count[index] = out_pos; + dev->audio_out_buf_pos[index] = out_pos; } else { - int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; - dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; + int in_pos; + + switch (dev->spec.data_alignment) { + case 0: + in_pos = BYTES_PER_SAMPLE + 2; + break; + case 2: + in_pos = BYTES_PER_SAMPLE; + break; + case 3: + default: + in_pos = 0; + break; + } + + dev->period_in_count[index] = in_pos; + dev->audio_in_buf_pos[index] = in_pos; } if (dev->streaming) @@ -221,7 +248,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; - if (dev->spec.data_alignment == 2) + if (dev->spec.data_alignment >= 2) bytes_per_sample++; bpp = ((runtime->rate / 8000) + CLOCK_DRIFT_TOLERANCE) @@ -253,6 +280,8 @@ static int snd_usb_caiaq_pcm_trigger(struct snd_pcm_substream *sub, int cmd) { struct snd_usb_caiaqdev *dev = snd_pcm_substream_chip(sub); + debug("%s(%p) cmd %d\n", __func__, sub, cmd); + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -402,6 +431,61 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, } } +static void read_in_urb_mode3(struct snd_usb_caiaqdev *dev, + const struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + int stream, i; + + /* paranoia check */ + if (iso->actual_length % (BYTES_PER_SAMPLE_USB * CHANNELS_PER_STREAM)) + return; + + for (i = 0; i < iso->actual_length;) { + for (stream = 0; stream < dev->n_streams; stream++) { + struct snd_pcm_substream *sub = dev->sub_capture[stream]; + char *audio_buf = NULL; + int c, n, sz = 0; + + if (sub && !dev->input_panic) { + struct snd_pcm_runtime *rt = sub->runtime; + audio_buf = rt->dma_area; + sz = frames_to_bytes(rt, rt->buffer_size); + } + + for (c = 0; c < CHANNELS_PER_STREAM; c++) { + /* 3 audio data bytes, followed by 1 check byte */ + if (audio_buf) { + for (n = 0; n < BYTES_PER_SAMPLE; n++) { + audio_buf[dev->audio_in_buf_pos[stream]++] = usb_buf[i+n]; + + if (dev->audio_in_buf_pos[stream] == sz) + dev->audio_in_buf_pos[stream] = 0; + } + + dev->period_in_count[stream] += BYTES_PER_SAMPLE; + } + + i += BYTES_PER_SAMPLE; + + if (usb_buf[i] != ((stream << 1) | c) && + !dev->first_packet) { + if (!dev->input_panic) + printk(" EXPECTED: %02x got %02x, c %d, stream %d, i %d\n", + ((stream << 1) | c), usb_buf[i], c, stream, i); + dev->input_panic = 1; + } + + i++; + } + } + } + + if (dev->first_packet > 0) + dev->first_packet--; +} + static void read_in_urb(struct snd_usb_caiaqdev *dev, const struct urb *urb, const struct usb_iso_packet_descriptor *iso) @@ -419,6 +503,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, case 2: read_in_urb_mode2(dev, urb, iso); break; + case 3: + read_in_urb_mode3(dev, urb, iso); + break; } if ((dev->input_panic || dev->output_panic) && !dev->warned) { @@ -429,9 +516,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, } } -static void fill_out_urb(struct snd_usb_caiaqdev *dev, - struct urb *urb, - const struct usb_iso_packet_descriptor *iso) +static void fill_out_urb_mode_0(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) { unsigned char *usb_buf = urb->transfer_buffer + iso->offset; struct snd_pcm_substream *sub; @@ -457,9 +544,67 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && i % (dev->n_streams * BYTES_PER_SAMPLE_USB) == - (dev->n_streams * CHANNELS_PER_STREAM)) - for (stream = 0; stream < dev->n_streams; stream++, i++) - usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); + (dev->n_streams * CHANNELS_PER_STREAM)) + for (stream = 0; stream < dev->n_streams; stream++, i++) + usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); + } +} + +static void fill_out_urb_mode_3(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + unsigned char *usb_buf = urb->transfer_buffer + iso->offset; + int stream, i; + + for (i = 0; i < iso->length;) { + for (stream = 0; stream < dev->n_streams; stream++) { + struct snd_pcm_substream *sub = dev->sub_playback[stream]; + char *audio_buf = NULL; + int c, n, sz = 0; + + if (sub) { + struct snd_pcm_runtime *rt = sub->runtime; + audio_buf = rt->dma_area; + sz = frames_to_bytes(rt, rt->buffer_size); + } + + for (c = 0; c < CHANNELS_PER_STREAM; c++) { + for (n = 0; n < BYTES_PER_SAMPLE; n++) { + if (audio_buf) { + usb_buf[i+n] = audio_buf[dev->audio_out_buf_pos[stream]++]; + + if (dev->audio_out_buf_pos[stream] == sz) + dev->audio_out_buf_pos[stream] = 0; + } else { + usb_buf[i+n] = 0; + } + } + + if (audio_buf) + dev->period_out_count[stream] += BYTES_PER_SAMPLE; + + i += BYTES_PER_SAMPLE; + + /* fill in the check byte pattern */ + usb_buf[i++] = (stream << 1) | c; + } + } + } +} + +static inline void fill_out_urb(struct snd_usb_caiaqdev *dev, + struct urb *urb, + const struct usb_iso_packet_descriptor *iso) +{ + switch (dev->spec.data_alignment) { + case 0: + case 2: + fill_out_urb_mode_0(dev, urb, iso); + break; + case 3: + fill_out_urb_mode_3(dev, urb, iso); + break; } } diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 91c804cd2782..00e5d0a469e1 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -55,6 +55,10 @@ static int control_info(struct snd_kcontrol *kcontrol, case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): maxval = 127; break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + maxval = 31; + break; } if (is_intval) { @@ -93,6 +97,7 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + int v = ucontrol->value.integer.value[0]; unsigned char cmd = EP1_CMD_WRITE_IO; if (dev->chip.usb_id == @@ -100,12 +105,27 @@ static int control_put(struct snd_kcontrol *kcontrol, cmd = EP1_CMD_DIMM_LEDS; if (pos & CNT_INTVAL) { - dev->control_state[pos & ~CNT_INTVAL] - = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, cmd, - dev->control_state, sizeof(dev->control_state)); + int i = pos & ~CNT_INTVAL; + + dev->control_state[i] = v; + + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4)) { + int actual_len; + + dev->ep8_out_buf[0] = i; + dev->ep8_out_buf[1] = v; + + usb_bulk_msg(dev->chip.dev, + usb_sndbulkpipe(dev->chip.dev, 8), + dev->ep8_out_buf, sizeof(dev->ep8_out_buf), + &actual_len, 200); + } else { + snd_usb_caiaq_send_command(dev, cmd, + dev->control_state, sizeof(dev->control_state)); + } } else { - if (ucontrol->value.integer.value[0]) + if (v) dev->control_state[pos / 8] |= 1 << (pos % 8); else dev->control_state[pos / 8] &= ~(1 << (pos % 8)); @@ -296,6 +316,179 @@ static struct caiaq_controller kontrolx1_controller[] = { { "LED Deck B: SYNC", 8 | CNT_INTVAL }, }; +static struct caiaq_controller kontrols4_controller[] = { + { "LED: Master: Quant", 10 | CNT_INTVAL }, + { "LED: Master: Headphone", 11 | CNT_INTVAL }, + { "LED: Master: Master", 12 | CNT_INTVAL }, + { "LED: Master: Snap", 14 | CNT_INTVAL }, + { "LED: Master: Warning", 15 | CNT_INTVAL }, + { "LED: Master: Master button", 112 | CNT_INTVAL }, + { "LED: Master: Snap button", 113 | CNT_INTVAL }, + { "LED: Master: Rec", 118 | CNT_INTVAL }, + { "LED: Master: Size", 119 | CNT_INTVAL }, + { "LED: Master: Quant button", 120 | CNT_INTVAL }, + { "LED: Master: Browser button", 121 | CNT_INTVAL }, + { "LED: Master: Play button", 126 | CNT_INTVAL }, + { "LED: Master: Undo button", 127 | CNT_INTVAL }, + + { "LED: Channel A: >", 4 | CNT_INTVAL }, + { "LED: Channel A: <", 5 | CNT_INTVAL }, + { "LED: Channel A: Meter 1", 97 | CNT_INTVAL }, + { "LED: Channel A: Meter 2", 98 | CNT_INTVAL }, + { "LED: Channel A: Meter 3", 99 | CNT_INTVAL }, + { "LED: Channel A: Meter 4", 100 | CNT_INTVAL }, + { "LED: Channel A: Meter 5", 101 | CNT_INTVAL }, + { "LED: Channel A: Meter 6", 102 | CNT_INTVAL }, + { "LED: Channel A: Meter clip", 103 | CNT_INTVAL }, + { "LED: Channel A: Active", 114 | CNT_INTVAL }, + { "LED: Channel A: Cue", 116 | CNT_INTVAL }, + { "LED: Channel A: FX1", 149 | CNT_INTVAL }, + { "LED: Channel A: FX2", 148 | CNT_INTVAL }, + + { "LED: Channel B: >", 2 | CNT_INTVAL }, + { "LED: Channel B: <", 3 | CNT_INTVAL }, + { "LED: Channel B: Meter 1", 89 | CNT_INTVAL }, + { "LED: Channel B: Meter 2", 90 | CNT_INTVAL }, + { "LED: Channel B: Meter 3", 91 | CNT_INTVAL }, + { "LED: Channel B: Meter 4", 92 | CNT_INTVAL }, + { "LED: Channel B: Meter 5", 93 | CNT_INTVAL }, + { "LED: Channel B: Meter 6", 94 | CNT_INTVAL }, + { "LED: Channel B: Meter clip", 95 | CNT_INTVAL }, + { "LED: Channel B: Active", 122 | CNT_INTVAL }, + { "LED: Channel B: Cue", 125 | CNT_INTVAL }, + { "LED: Channel B: FX1", 147 | CNT_INTVAL }, + { "LED: Channel B: FX2", 146 | CNT_INTVAL }, + + { "LED: Channel C: >", 6 | CNT_INTVAL }, + { "LED: Channel C: <", 7 | CNT_INTVAL }, + { "LED: Channel C: Meter 1", 105 | CNT_INTVAL }, + { "LED: Channel C: Meter 2", 106 | CNT_INTVAL }, + { "LED: Channel C: Meter 3", 107 | CNT_INTVAL }, + { "LED: Channel C: Meter 4", 108 | CNT_INTVAL }, + { "LED: Channel C: Meter 5", 109 | CNT_INTVAL }, + { "LED: Channel C: Meter 6", 110 | CNT_INTVAL }, + { "LED: Channel C: Meter clip", 111 | CNT_INTVAL }, + { "LED: Channel C: Active", 115 | CNT_INTVAL }, + { "LED: Channel C: Cue", 117 | CNT_INTVAL }, + { "LED: Channel C: FX1", 151 | CNT_INTVAL }, + { "LED: Channel C: FX2", 150 | CNT_INTVAL }, + + { "LED: Channel D: >", 0 | CNT_INTVAL }, + { "LED: Channel D: <", 1 | CNT_INTVAL }, + { "LED: Channel D: Meter 1", 81 | CNT_INTVAL }, + { "LED: Channel D: Meter 2", 82 | CNT_INTVAL }, + { "LED: Channel D: Meter 3", 83 | CNT_INTVAL }, + { "LED: Channel D: Meter 4", 84 | CNT_INTVAL }, + { "LED: Channel D: Meter 5", 85 | CNT_INTVAL }, + { "LED: Channel D: Meter 6", 86 | CNT_INTVAL }, + { "LED: Channel D: Meter clip", 87 | CNT_INTVAL }, + { "LED: Channel D: Active", 123 | CNT_INTVAL }, + { "LED: Channel D: Cue", 124 | CNT_INTVAL }, + { "LED: Channel D: FX1", 145 | CNT_INTVAL }, + { "LED: Channel D: FX2", 144 | CNT_INTVAL }, + + { "LED: Deck A: 1 (blue)", 22 | CNT_INTVAL }, + { "LED: Deck A: 1 (green)", 23 | CNT_INTVAL }, + { "LED: Deck A: 2 (blue)", 20 | CNT_INTVAL }, + { "LED: Deck A: 2 (green)", 21 | CNT_INTVAL }, + { "LED: Deck A: 3 (blue)", 18 | CNT_INTVAL }, + { "LED: Deck A: 3 (green)", 19 | CNT_INTVAL }, + { "LED: Deck A: 4 (blue)", 16 | CNT_INTVAL }, + { "LED: Deck A: 4 (green)", 17 | CNT_INTVAL }, + { "LED: Deck A: Load", 44 | CNT_INTVAL }, + { "LED: Deck A: Deck C button", 45 | CNT_INTVAL }, + { "LED: Deck A: In", 47 | CNT_INTVAL }, + { "LED: Deck A: Out", 46 | CNT_INTVAL }, + { "LED: Deck A: Shift", 24 | CNT_INTVAL }, + { "LED: Deck A: Sync", 27 | CNT_INTVAL }, + { "LED: Deck A: Cue", 26 | CNT_INTVAL }, + { "LED: Deck A: Play", 25 | CNT_INTVAL }, + { "LED: Deck A: Tempo up", 33 | CNT_INTVAL }, + { "LED: Deck A: Tempo down", 32 | CNT_INTVAL }, + { "LED: Deck A: Master", 34 | CNT_INTVAL }, + { "LED: Deck A: Keylock", 35 | CNT_INTVAL }, + { "LED: Deck A: Deck A", 37 | CNT_INTVAL }, + { "LED: Deck A: Deck C", 36 | CNT_INTVAL }, + { "LED: Deck A: Samples", 38 | CNT_INTVAL }, + { "LED: Deck A: On Air", 39 | CNT_INTVAL }, + { "LED: Deck A: Sample 1", 31 | CNT_INTVAL }, + { "LED: Deck A: Sample 2", 30 | CNT_INTVAL }, + { "LED: Deck A: Sample 3", 29 | CNT_INTVAL }, + { "LED: Deck A: Sample 4", 28 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - A", 55 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - B", 54 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - C", 53 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - D", 52 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - E", 51 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - F", 50 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - G", 49 | CNT_INTVAL }, + { "LED: Deck A: Digit 1 - dot", 48 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - A", 63 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - B", 62 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - C", 61 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - D", 60 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - E", 59 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - F", 58 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - G", 57 | CNT_INTVAL }, + { "LED: Deck A: Digit 2 - dot", 56 | CNT_INTVAL }, + + { "LED: Deck B: 1 (blue)", 78 | CNT_INTVAL }, + { "LED: Deck B: 1 (green)", 79 | CNT_INTVAL }, + { "LED: Deck B: 2 (blue)", 76 | CNT_INTVAL }, + { "LED: Deck B: 2 (green)", 77 | CNT_INTVAL }, + { "LED: Deck B: 3 (blue)", 74 | CNT_INTVAL }, + { "LED: Deck B: 3 (green)", 75 | CNT_INTVAL }, + { "LED: Deck B: 4 (blue)", 72 | CNT_INTVAL }, + { "LED: Deck B: 4 (green)", 73 | CNT_INTVAL }, + { "LED: Deck B: Load", 180 | CNT_INTVAL }, + { "LED: Deck B: Deck D button", 181 | CNT_INTVAL }, + { "LED: Deck B: In", 183 | CNT_INTVAL }, + { "LED: Deck B: Out", 182 | CNT_INTVAL }, + { "LED: Deck B: Shift", 64 | CNT_INTVAL }, + { "LED: Deck B: Sync", 67 | CNT_INTVAL }, + { "LED: Deck B: Cue", 66 | CNT_INTVAL }, + { "LED: Deck B: Play", 65 | CNT_INTVAL }, + { "LED: Deck B: Tempo up", 185 | CNT_INTVAL }, + { "LED: Deck B: Tempo down", 184 | CNT_INTVAL }, + { "LED: Deck B: Master", 186 | CNT_INTVAL }, + { "LED: Deck B: Keylock", 187 | CNT_INTVAL }, + { "LED: Deck B: Deck B", 189 | CNT_INTVAL }, + { "LED: Deck B: Deck D", 188 | CNT_INTVAL }, + { "LED: Deck B: Samples", 190 | CNT_INTVAL }, + { "LED: Deck B: On Air", 191 | CNT_INTVAL }, + { "LED: Deck B: Sample 1", 71 | CNT_INTVAL }, + { "LED: Deck B: Sample 2", 70 | CNT_INTVAL }, + { "LED: Deck B: Sample 3", 69 | CNT_INTVAL }, + { "LED: Deck B: Sample 4", 68 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - A", 175 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - B", 174 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - C", 173 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - D", 172 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - E", 171 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - F", 170 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - G", 169 | CNT_INTVAL }, + { "LED: Deck B: Digit 1 - dot", 168 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - A", 167 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - B", 166 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - C", 165 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - D", 164 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - E", 163 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - F", 162 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - G", 161 | CNT_INTVAL }, + { "LED: Deck B: Digit 2 - dot", 160 | CNT_INTVAL }, + + { "LED: FX1: dry/wet", 153 | CNT_INTVAL }, + { "LED: FX1: 1", 154 | CNT_INTVAL }, + { "LED: FX1: 2", 155 | CNT_INTVAL }, + { "LED: FX1: 3", 156 | CNT_INTVAL }, + { "LED: FX1: Mode", 157 | CNT_INTVAL }, + { "LED: FX2: dry/wet", 129 | CNT_INTVAL }, + { "LED: FX2: 1", 130 | CNT_INTVAL }, + { "LED: FX2: 2", 131 | CNT_INTVAL }, + { "LED: FX2: 3", 132 | CNT_INTVAL }, + { "LED: FX2: Mode", 133 | CNT_INTVAL }, +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -354,6 +547,11 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(kontrolx1_controller, ARRAY_SIZE(kontrolx1_controller), dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + ret = add_controls(kontrols4_controller, + ARRAY_SIZE(kontrols4_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index da9cb6dcee2a..6480c3283c05 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -48,7 +48,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," "{Native Instruments, GuitarRig mobile}" - "{Native Instruments, Traktor Kontrol X1}"); + "{Native Instruments, Traktor Kontrol X1}" + "{Native Instruments, Traktor Kontrol S4}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -134,6 +135,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_TRAKTORKONTROLX1 }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_TRAKTORKONTROLS4 + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index f1117ecc84fd..e3d8a3efb35b 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -16,6 +16,7 @@ #define USB_PID_SESSIONIO 0x1915 #define USB_PID_GUITARRIGMOBILE 0x0d8d #define USB_PID_TRAKTORKONTROLX1 0x2305 +#define USB_PID_TRAKTORKONTROLS4 0xbaff #define EP1_BUFSIZE 64 #define EP4_BUFSIZE 512 @@ -99,13 +100,14 @@ struct snd_usb_caiaqdev { struct snd_pcm_substream *sub_capture[MAX_STREAMS]; /* Controls */ - unsigned char control_state[64]; + unsigned char control_state[256]; + unsigned char ep8_out_buf[2]; /* Linux input */ #ifdef CONFIG_SND_USB_CAIAQ_INPUT struct input_dev *input_dev; char phys[64]; /* physical device path */ - unsigned short keycode[64]; + unsigned short keycode[128]; struct urb *ep4_in_urb; unsigned char ep4_in_buf[EP4_BUFSIZE]; #endif diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index dcb620796d9e..4432ef7a70a9 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -67,7 +67,12 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; -#define KONTROLX1_INPUTS 40 +#define KONTROLX1_INPUTS (40) +#define KONTROLS4_BUTTONS (12 * 8) +#define KONTROLS4_AXIS (46) + +#define KONTROLS4_BUTTON(X) ((X) + BTN_MISC) +#define KONTROLS4_ABS(X) ((X) + ABS_HAT0X) #define DEG90 (range / 2) #define DEG180 (range) @@ -139,6 +144,13 @@ static unsigned int decode_erp(unsigned char a, unsigned char b) #undef HIGH_PEAK #undef LOW_PEAK +static inline void snd_caiaq_input_report_abs(struct snd_usb_caiaqdev *dev, + int axis, const unsigned char *buf, + int offset) +{ + input_report_abs(dev->input_dev, axis, + (buf[offset * 2] << 8) | buf[offset * 2 + 1]); +} static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, const unsigned char *buf, @@ -148,36 +160,30 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): - input_report_abs(input_dev, ABS_X, (buf[4] << 8) | buf[5]); - input_report_abs(input_dev, ABS_Y, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Z, (buf[2] << 8) | buf[3]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_X, buf, 2); + snd_caiaq_input_report_abs(dev, ABS_Y, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_Z, buf, 1); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): - input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); - input_sync(input_dev); - break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - input_report_abs(input_dev, ABS_X, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_X, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_Y, buf, 1); + snd_caiaq_input_report_abs(dev, ABS_Z, buf, 2); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): - input_report_abs(input_dev, ABS_HAT0X, (buf[8] << 8) | buf[9]); - input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); - input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); - input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); - input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]); - input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); - input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); - input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); - input_sync(input_dev); + snd_caiaq_input_report_abs(dev, ABS_HAT0X, buf, 4); + snd_caiaq_input_report_abs(dev, ABS_HAT0Y, buf, 2); + snd_caiaq_input_report_abs(dev, ABS_HAT1X, buf, 6); + snd_caiaq_input_report_abs(dev, ABS_HAT1Y, buf, 1); + snd_caiaq_input_report_abs(dev, ABS_HAT2X, buf, 7); + snd_caiaq_input_report_abs(dev, ABS_HAT2Y, buf, 0); + snd_caiaq_input_report_abs(dev, ABS_HAT3X, buf, 5); + snd_caiaq_input_report_abs(dev, ABS_HAT3Y, buf, 3); break; } + + input_sync(input_dev); } static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, @@ -250,6 +256,150 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, input_sync(input_dev); } +#define TKS4_MSGBLOCK_SIZE 16 + +static void snd_usb_caiaq_tks4_dispatch(struct snd_usb_caiaqdev *dev, + const unsigned char *buf, + unsigned int len) +{ + while (len) { + unsigned int i, block_id = (buf[0] << 8) | buf[1]; + + switch (block_id) { + case 0: + /* buttons */ + for (i = 0; i < KONTROLS4_BUTTONS; i++) + input_report_key(dev->input_dev, KONTROLS4_BUTTON(i), + (buf[4 + (i / 8)] >> (i % 8)) & 1); + break; + + case 1: + /* left wheel */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(36), buf[9] | ((buf[8] & 0x3) << 8)); + /* right wheel */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(37), buf[13] | ((buf[12] & 0x3) << 8)); + + /* rotary encoders */ + input_report_abs(dev->input_dev, KONTROLS4_ABS(38), buf[3] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(39), buf[4] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(40), buf[4] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(41), buf[5] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(42), buf[5] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(43), buf[6] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(44), buf[6] & 0xf); + input_report_abs(dev->input_dev, KONTROLS4_ABS(45), buf[7] >> 4); + input_report_abs(dev->input_dev, KONTROLS4_ABS(46), buf[7] & 0xf); + + break; + case 2: + /* Volume Fader Channel D */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(0), buf, 1); + /* Volume Fader Channel B */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(1), buf, 2); + /* Volume Fader Channel A */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(2), buf, 3); + /* Volume Fader Channel C */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(3), buf, 4); + /* Loop Volume */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(4), buf, 6); + /* Crossfader */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(7), buf, 7); + + break; + + case 3: + /* Tempo Fader R */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(6), buf, 3); + /* Tempo Fader L */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(5), buf, 4); + /* Mic Volume */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(8), buf, 6); + /* Cue Mix */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(9), buf, 7); + + break; + + case 4: + /* Wheel distance sensor L */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(10), buf, 1); + /* Wheel distance sensor R */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(11), buf, 2); + /* Channel D EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(12), buf, 3); + /* Channel D EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(13), buf, 4); + /* Channel D EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(14), buf, 5); + /* Channel D EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(15), buf, 6); + /* FX2 - dry/wet */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(16), buf, 7); + + break; + + case 5: + /* FX2 - 1 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(17), buf, 1); + /* FX2 - 2 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(18), buf, 2); + /* FX2 - 3 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(19), buf, 3); + /* Channel B EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(20), buf, 4); + /* Channel B EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(21), buf, 5); + /* Channel B EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(22), buf, 6); + /* Channel B EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(23), buf, 7); + + break; + + case 6: + /* Channel A EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(24), buf, 1); + /* Channel A EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(25), buf, 2); + /* Channel A EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(26), buf, 3); + /* Channel A EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(27), buf, 4); + /* Channel C EQ - Filter */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(28), buf, 5); + /* Channel C EQ - Low */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(29), buf, 6); + /* Channel C EQ - Mid */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(30), buf, 7); + + break; + + case 7: + /* Channel C EQ - Hi */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(31), buf, 1); + /* FX1 - wet/dry */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(32), buf, 2); + /* FX1 - 1 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(33), buf, 3); + /* FX1 - 2 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(34), buf, 4); + /* FX1 - 3 */ + snd_caiaq_input_report_abs(dev, KONTROLS4_ABS(35), buf, 5); + + break; + + default: + debug("%s(): bogus block (id %d)\n", + __func__, block_id); + return; + } + + len -= TKS4_MSGBLOCK_SIZE; + buf += TKS4_MSGBLOCK_SIZE; + } + + input_sync(dev->input_dev); +} + static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) { struct snd_usb_caiaqdev *dev = urb->context; @@ -259,11 +409,11 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) if (urb->status || !dev || urb != dev->ep4_in_urb) return; - if (urb->actual_length < 24) - goto requeue; - switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (urb->actual_length < 24) + goto requeue; + if (buf[0] & 0x3) snd_caiaq_input_read_io(dev, buf + 1, 7); @@ -271,6 +421,10 @@ static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) snd_caiaq_input_read_analog(dev, buf + 8, 16); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + snd_usb_caiaq_tks4_dispatch(dev, buf, urb->actual_length); + break; } requeue: @@ -289,6 +443,7 @@ static int snd_usb_caiaq_input_open(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) return -EIO; break; @@ -306,6 +461,7 @@ static void snd_usb_caiaq_input_close(struct input_dev *idev) switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): usb_kill_urb(dev->ep4_in_urb); break; } @@ -456,6 +612,46 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLS4): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLS4_BUTTONS); + for (i = 0; i < KONTROLS4_BUTTONS; i++) + dev->keycode[i] = KONTROLS4_BUTTON(i); + input->keycodemax = KONTROLS4_BUTTONS; + + for (i = 0; i < KONTROLS4_AXIS; i++) { + int axis = KONTROLS4_ABS(i); + input->absbit[BIT_WORD(axis)] |= BIT_MASK(axis); + } + + /* 36 analog potentiometers and faders */ + for (i = 0; i < 36; i++) + input_set_abs_params(input, KONTROLS4_ABS(i), 0, 0xfff, 0, 10); + + /* 2 encoder wheels */ + input_set_abs_params(input, KONTROLS4_ABS(36), 0, 0x3ff, 0, 1); + input_set_abs_params(input, KONTROLS4_ABS(37), 0, 0x3ff, 0, 1); + + /* 9 rotary encoders */ + for (i = 0; i < 9; i++) + input_set_abs_params(input, KONTROLS4_ABS(38+i), 0, 0xf, 0, 1); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + + break; + default: /* no input methods supported on this device */ goto exit_free_idev; -- cgit v1.2.3 From cea310e8f8702226f982f09386cfd3c5793c5e2f Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Fri, 10 Sep 2010 16:29:56 -0700 Subject: ALSA: hda_intel: ALSA HD Audio patch for Intel Patsburg DeviceIDs This patch adds the Intel Patsburg (PCH) HD Audio Controller DeviceIDs. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1053fff4bd0a..34940a079051 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -126,6 +126,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH10}," "{Intel, PCH}," "{Intel, CPT}," + "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2749,6 +2750,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH }, /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, + /* PBG */ + { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v1.2.3 From 2ca9cac965e81da4b74f2dcec4b87ebfd106b357 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Fri, 10 Sep 2010 15:47:55 +0200 Subject: ALSA: hda - Add quirk for Toshiba C650D using a Conexant CX20585 Add a quirk for laptop Toshiba Satellite C650D to have proper external HP and external Mic support. Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71f9d6475b09..972e7c453b3d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3092,6 +3092,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), -- cgit v1.2.3 From 147fcf1c211f1a87bf4d0711b7e9637f3d6ce080 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 11 Sep 2010 22:10:59 -0700 Subject: sound: Remove pr_ uses of KERN_ Signed-off-by: Joe Perches Acked-by: Mark Brown Acked-by: Geoff Levand Signed-off-by: Takashi Iwai --- sound/ppc/snd_ps3.c | 2 +- sound/soc/s3c24xx/s3c-dma.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 2f12da4da561..581a670e8261 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -579,7 +579,7 @@ static int snd_ps3_delay_to_bytes(struct snd_pcm_substream *substream, rate * delay_ms / 1000) * substream->runtime->channels; - pr_debug(KERN_ERR "%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", + pr_debug("%s: time=%d rate=%d bytes=%ld, frames=%d, ret=%d\n", __func__, delay_ms, rate, diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 1b61c23ff300..f1b1bc4bacfb 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -94,8 +94,7 @@ static void s3c_dma_enqueue(struct snd_pcm_substream *substream) if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); + pr_debug("%s: corrected dma len %ld\n", __func__, len); } ret = s3c2410_dma_enqueue(prtd->params->channel, -- cgit v1.2.3 From 47023ec774b6f73bb11d9f3b00b21f2bbd87e0f2 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 13 Sep 2010 21:24:02 -0700 Subject: sound: Use static const char * const where possible Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e2e73895db12..eb4094270a4f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -142,7 +142,7 @@ int snd_pcm_info_user(struct snd_pcm_substream *substream, #ifdef RULES_DEBUG #define HW_PARAM(v) [SNDRV_PCM_HW_PARAM_##v] = #v -char *snd_pcm_hw_param_names[] = { +static const char * const snd_pcm_hw_param_names[] = { HW_PARAM(ACCESS), HW_PARAM(FORMAT), HW_PARAM(SUBFORMAT), -- cgit v1.2.3 From a254dba37c5a372fc8b44ba29509ba052d4e859d Mon Sep 17 00:00:00 2001 From: Ben Hutchings Date: Sun, 12 Sep 2010 02:41:47 +0100 Subject: ALSA: emux: Add trivial compat ioctl handler Reported-by: Carmen Cru Signed-off-by: Ben Hutchings Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_hwdep.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index ff0b2a8fd25b..5ae1eae9f6db 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -128,6 +128,9 @@ snd_emux_init_hwdep(struct snd_emux *emu) strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME); hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE; hw->ops.ioctl = snd_emux_hwdep_ioctl; + /* The ioctl parameter types are compatible between 32- and + * 64-bit architectures, so use the same function. */ + hw->ops.ioctl_compat = snd_emux_hwdep_ioctl; hw->exclusive = 1; hw->private_data = emu; if ((err = snd_card_register(emu->card)) < 0) -- cgit v1.2.3 From 7b6c3a34e93aafc5dd9adc7dee87c7fa61d8bdbb Mon Sep 17 00:00:00 2001 From: Andreas Schwab Date: Sat, 11 Sep 2010 13:16:41 +0200 Subject: ALSA: sound/ppc/powermac: remove undefined operations Modifying an object twice without an intervening sequence point is undefined. Signed-off-by: Andreas Schwab Signed-off-by: Takashi Iwai --- sound/ppc/tumbler.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 20afdf9772ee..961d98297695 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -785,7 +785,7 @@ static int snapper_set_capture_source(struct pmac_tumbler *mix) if (! mix->i2c.client) return -ENODEV; if (mix->capture_source) - mix->acs = mix->acs |= 2; + mix->acs |= 2; else mix->acs &= ~2; return i2c_smbus_write_byte_data(mix->i2c.client, TAS_REG_ACS, mix->acs); -- cgit v1.2.3 From 645ef9ef1fc0ff70456495b1e21d3420b7b08541 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 14 Sep 2010 21:53:41 +0200 Subject: sound: autoconvert trivial BKL users to private mutex The usage of the BKL in the OSS sound drivers is trivial, and each of them only locks against itself, so it can be turned into per-driver mutexes. This is the script that was used for the conversion: file=$1 name=$2 if grep -q lock_kernel ${file} ; then if grep -q 'include.*linux.mutex.h' ${file} ; then sed -i '/include.*/d' ${file} else sed -i 's/include.*.*$/include /g' ${file} fi sed -i ${file} \ -e "/^#include.*linux.mutex.h/,$ { 1,/^\(static\|int\|long\)/ { /^\(static\|int\|long\)/istatic DEFINE_MUTEX(${name}_mutex); } }" \ -e "s/\(un\)*lock_kernel\>[ ]*()/mutex_\1lock(\&${name}_mutex)/g" \ -e '/[ ]*cycle_kernel_lock();/d' else sed -i -e '/include.*\/d' ${file} \ -e '/cycle_kernel_lock()/d' fi Signed-off-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/oss/au1550_ac97.c | 30 ++++++++++++++-------------- sound/oss/dmasound/dmasound_core.c | 41 +++++++++++++++++++------------------- sound/oss/msnd_pinnacle.c | 15 +++++++------- sound/oss/sh_dac_audio.c | 13 ++++++------ sound/oss/soundcard.c | 41 +++++++++++++++++++------------------- sound/oss/swarm_cs4297a.c | 20 +++++++++---------- sound/oss/vwsnd.c | 30 ++++++++++++++-------------- 7 files changed, 97 insertions(+), 93 deletions(-) (limited to 'sound') diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 8a12621d8b3a..a8f626d99c5b 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -43,7 +43,6 @@ #include #include #include -#include #include #include #include @@ -77,6 +76,7 @@ /* Boot options * 0 = no VRA, 1 = use VRA if codec supports it */ +static DEFINE_MUTEX(au1550_ac97_mutex); static int vra = 1; module_param(vra, bool, 0); MODULE_PARM_DESC(vra, "if 1 use VRA if codec supports it"); @@ -798,9 +798,9 @@ au1550_llseek(struct file *file, loff_t offset, int origin) static int au1550_open_mixdev(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); file->private_data = &au1550_state; - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return 0; } @@ -824,9 +824,9 @@ au1550_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) struct ac97_codec *codec = s->codec; int ret; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); ret = mixdev_ioctl(codec, cmd, arg); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return ret; } @@ -1255,7 +1255,7 @@ au1550_mmap(struct file *file, struct vm_area_struct *vma) unsigned long size; int ret = 0; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); mutex_lock(&s->sem); if (vma->vm_flags & VM_WRITE) db = &s->dma_dac; @@ -1283,7 +1283,7 @@ au1550_mmap(struct file *file, struct vm_area_struct *vma) db->mapped = 1; out: mutex_unlock(&s->sem); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return ret; } @@ -1781,9 +1781,9 @@ au1550_unlocked_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { int ret; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); ret = au1550_ioctl(file, cmd, arg); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return ret; } @@ -1804,7 +1804,7 @@ au1550_open(struct inode *inode, struct file *file) #endif file->private_data = s; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); /* wait for device to become free */ mutex_lock(&s->open_mutex); while (s->open_mode & file->f_mode) { @@ -1861,7 +1861,7 @@ au1550_open(struct inode *inode, struct file *file) out: mutex_unlock(&s->open_mutex); out2: - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return ret; } @@ -1870,12 +1870,12 @@ au1550_release(struct inode *inode, struct file *file) { struct au1550_state *s = file->private_data; - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); if (file->f_mode & FMODE_WRITE) { - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); drain_dac(s, file->f_flags & O_NONBLOCK); - lock_kernel(); + mutex_lock(&au1550_ac97_mutex); } mutex_lock(&s->open_mutex); @@ -1892,7 +1892,7 @@ au1550_release(struct inode *inode, struct file *file) s->open_mode &= ((~file->f_mode) & (FMODE_READ|FMODE_WRITE)); mutex_unlock(&s->open_mutex); wake_up(&s->open_wait); - unlock_kernel(); + mutex_unlock(&au1550_ac97_mutex); return 0; } diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 6ecd41abb066..87e2c72651f5 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -181,7 +181,7 @@ #include #include #include -#include +#include #include @@ -194,6 +194,7 @@ * Declarations */ +static DEFINE_MUTEX(dmasound_core_mutex); int dmasound_catchRadius = 0; module_param(dmasound_catchRadius, int, 0); @@ -323,22 +324,22 @@ static struct { static int mixer_open(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&dmasound_core_mutex); if (!try_module_get(dmasound.mach.owner)) { - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return -ENODEV; } mixer.busy = 1; - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; } static int mixer_release(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&dmasound_core_mutex); mixer.busy = 0; module_put(dmasound.mach.owner); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; } @@ -370,9 +371,9 @@ static long mixer_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) { int ret; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); ret = mixer_ioctl(file, cmd, arg); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return ret; } @@ -752,9 +753,9 @@ static int sq_open(struct inode *inode, struct file *file) { int rc; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); if (!try_module_get(dmasound.mach.owner)) { - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return -ENODEV; } @@ -799,11 +800,11 @@ static int sq_open(struct inode *inode, struct file *file) sound_set_format(AFMT_MU_LAW); } #endif - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; out: module_put(dmasound.mach.owner); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return rc; } @@ -869,7 +870,7 @@ static int sq_release(struct inode *inode, struct file *file) { int rc = 0; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); if (file->f_mode & FMODE_WRITE) { if (write_sq.busy) @@ -900,7 +901,7 @@ static int sq_release(struct inode *inode, struct file *file) write_sq_wake_up(file); /* checks f_mode */ #endif /* blocking open() */ - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return rc; } @@ -1141,9 +1142,9 @@ static long sq_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) { int ret; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); ret = sq_ioctl(file, cmd, arg); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return ret; } @@ -1257,7 +1258,7 @@ static int state_open(struct inode *inode, struct file *file) int len = 0; int ret; - lock_kernel(); + mutex_lock(&dmasound_core_mutex); ret = -EBUSY; if (state.busy) goto out; @@ -1329,16 +1330,16 @@ printk("dmasound: stat buffer used %d bytes\n", len) ; state.len = len; ret = 0; out: - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return ret; } static int state_release(struct inode *inode, struct file *file) { - lock_kernel(); + mutex_lock(&dmasound_core_mutex); state.busy = 0; module_put(dmasound.mach.owner); - unlock_kernel(); + mutex_unlock(&dmasound_core_mutex); return 0; } diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index 2e48b17667d0..b4c1eb504c22 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -39,7 +39,7 @@ #include #include #include -#include +#include #include #include #include @@ -79,6 +79,7 @@ dev.rec_sample_rate / \ dev.rec_channels) +static DEFINE_MUTEX(msnd_pinnacle_mutex); static multisound_dev_t dev; #ifndef HAVE_DSPCODEH @@ -651,12 +652,12 @@ static long dev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) ret = -EINVAL; - lock_kernel(); + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) ret = dsp_ioctl(file, cmd, arg); else if (minor == dev.mixer_minor) ret = mixer_ioctl(cmd, arg); - unlock_kernel(); + mutex_unlock(&msnd_pinnacle_mutex); return ret; } @@ -761,7 +762,7 @@ static int dev_open(struct inode *inode, struct file *file) int minor = iminor(inode); int err = 0; - lock_kernel(); + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) { if ((file->f_mode & FMODE_WRITE && test_bit(F_AUDIO_WRITE_INUSE, &dev.flags)) || @@ -791,7 +792,7 @@ static int dev_open(struct inode *inode, struct file *file) } else err = -EINVAL; out: - unlock_kernel(); + mutex_unlock(&msnd_pinnacle_mutex); return err; } @@ -800,14 +801,14 @@ static int dev_release(struct inode *inode, struct file *file) int minor = iminor(inode); int err = 0; - lock_kernel(); + mutex_lock(&msnd_pinnacle_mutex); if (minor == dev.dsp_minor) err = dsp_release(file); else if (minor == dev.mixer_minor) { /* nothing */ } else err = -EINVAL; - unlock_kernel(); + mutex_unlock(&msnd_pinnacle_mutex); return err; } diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index 479e3025a8a3..53bba16bf709 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -16,7 +16,7 @@ #include #include #include -#include +#include #include #include #include @@ -34,6 +34,7 @@ #define BUFFER_SIZE 48000 +static DEFINE_MUTEX(sh_dac_audio_mutex); static int rate; static int empty; static char *data_buffer, *buffer_begin, *buffer_end; @@ -163,9 +164,9 @@ static long dac_audio_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) { int ret; - lock_kernel(); + mutex_lock(&sh_dac_audio_mutex); ret = dac_audio_ioctl(file, cmd, arg); - unlock_kernel(); + mutex_unlock(&sh_dac_audio_mutex); return ret; } @@ -229,16 +230,16 @@ static int dac_audio_open(struct inode *inode, struct file *file) if (file->f_mode & FMODE_READ) return -ENODEV; - lock_kernel(); + mutex_lock(&sh_dac_audio_mutex); if (in_use) { - unlock_kernel(); + mutex_unlock(&sh_dac_audio_mutex); return -EBUSY; } in_use = 1; dac_audio_start(); - unlock_kernel(); + mutex_unlock(&sh_dac_audio_mutex); return 0; } diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 92aa762ffb7e..938ed94f904f 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -40,7 +40,7 @@ #include #include #include -#include +#include #include #include #include @@ -56,6 +56,7 @@ * Table for permanently allocated memory (used when unloading the module) */ void * sound_mem_blocks[MAX_MEM_BLOCKS]; +static DEFINE_MUTEX(soundcard_mutex); int sound_nblocks = 0; /* Persistent DMA buffers */ @@ -151,7 +152,7 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof * big one anyway, we might as well bandage here.. */ - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { @@ -169,7 +170,7 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof case SND_DEV_MIDIN: ret = MIDIbuf_read(dev, file, buf, count); } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -178,7 +179,7 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou int dev = iminor(file->f_path.dentry->d_inode); int ret = -EINVAL; - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_SEQ: @@ -196,7 +197,7 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou ret = MIDIbuf_write(dev, file, buf, count); break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -210,7 +211,7 @@ static int sound_open(struct inode *inode, struct file *file) printk(KERN_ERR "Invalid minor device %d\n", dev); return -ENXIO; } - lock_kernel(); + mutex_lock(&soundcard_mutex); switch (dev & 0x0f) { case SND_DEV_CTL: dev >>= 4; @@ -247,7 +248,7 @@ static int sound_open(struct inode *inode, struct file *file) retval = -ENXIO; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return 0; } @@ -255,7 +256,7 @@ static int sound_release(struct inode *inode, struct file *file) { int dev = iminor(inode); - lock_kernel(); + mutex_lock(&soundcard_mutex); DEB(printk("sound_release(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_CTL: @@ -280,7 +281,7 @@ static int sound_release(struct inode *inode, struct file *file) default: printk(KERN_ERR "Sound error: Releasing unknown device 0x%02x\n", dev); } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return 0; } @@ -354,7 +355,7 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); - lock_kernel(); + mutex_lock(&soundcard_mutex); if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ (dev & 0x0f) != SND_DEV_CTL) { dtype = dev & 0x0f; @@ -369,7 +370,7 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) ret = sound_mixer_ioctl(dev >> 4, cmd, p); break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -399,7 +400,7 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) break; } - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return ret; } @@ -439,35 +440,35 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) printk(KERN_ERR "Sound: mmap() not supported for other than audio devices\n"); return -EINVAL; } - lock_kernel(); + mutex_lock(&soundcard_mutex); if (vma->vm_flags & VM_WRITE) /* Map write and read/write to the output buf */ dmap = audio_devs[dev]->dmap_out; else if (vma->vm_flags & VM_READ) dmap = audio_devs[dev]->dmap_in; else { printk(KERN_ERR "Sound: Undefined mmap() access\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EINVAL; } if (dmap == NULL) { printk(KERN_ERR "Sound: mmap() error. dmap == NULL\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (dmap->raw_buf == NULL) { printk(KERN_ERR "Sound: mmap() called when raw_buf == NULL\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (dmap->mapping_flags) { printk(KERN_ERR "Sound: mmap() called twice for the same DMA buffer\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EIO; } if (vma->vm_pgoff != 0) { printk(KERN_ERR "Sound: mmap() offset must be 0.\n"); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EINVAL; } size = vma->vm_end - vma->vm_start; @@ -478,7 +479,7 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) if (remap_pfn_range(vma, vma->vm_start, virt_to_phys(dmap->raw_buf) >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot)) { - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return -EAGAIN; } @@ -490,7 +491,7 @@ static int sound_mmap(struct file *file, struct vm_area_struct *vma) memset(dmap->raw_buf, dmap->neutral_byte, dmap->bytes_in_use); - unlock_kernel(); + mutex_unlock(&soundcard_mutex); return 0; } diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index b15840ad2527..44357d877a27 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -68,7 +68,6 @@ #include #include #include -#include #include #include #include @@ -94,6 +93,7 @@ struct cs4297a_state; +static DEFINE_MUTEX(swarm_cs4297a_mutex); static void stop_dac(struct cs4297a_state *s); static void stop_adc(struct cs4297a_state *s); static void start_dac(struct cs4297a_state *s); @@ -1535,7 +1535,7 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()+\n")); - lock_kernel(); + mutex_lock(&swarm_cs4297a_mutex); list_for_each(entry, &cs4297a_devs) { s = list_entry(entry, struct cs4297a_state, list); @@ -1547,7 +1547,7 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) CS_DBGOUT(CS_FUNCTION | CS_OPEN | CS_ERROR, 2, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- -ENODEV\n")); - unlock_kernel(); + mutex_unlock(&swarm_cs4297a_mutex); return -ENODEV; } VALIDATE_STATE(s); @@ -1555,7 +1555,7 @@ static int cs4297a_open_mixdev(struct inode *inode, struct file *file) CS_DBGOUT(CS_FUNCTION | CS_OPEN, 4, printk(KERN_INFO "cs4297a: cs4297a_open_mixdev()- 0\n")); - unlock_kernel(); + mutex_unlock(&swarm_cs4297a_mutex); return nonseekable_open(inode, file); } @@ -1575,10 +1575,10 @@ static int cs4297a_ioctl_mixdev(struct file *file, unsigned int cmd, unsigned long arg) { int ret; - lock_kernel(); + mutex_lock(&swarm_cs4297a_mutex); ret = mixer_ioctl((struct cs4297a_state *) file->private_data, cmd, arg); - unlock_kernel(); + mutex_unlock(&swarm_cs4297a_mutex); return ret; } @@ -2350,9 +2350,9 @@ static long cs4297a_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) { int ret; - lock_kernel(); + mutex_lock(&swarm_cs4297a_mutex); ret = cs4297a_ioctl(file, cmd, arg); - unlock_kernel(); + mutex_unlock(&swarm_cs4297a_mutex); return ret; } @@ -2509,9 +2509,9 @@ static int cs4297a_open(struct inode *inode, struct file *file) { int ret; - lock_kernel(); + mutex_lock(&swarm_cs4297a_mutex); ret = cs4297a_open(inode, file); - unlock_kernel(); + mutex_unlock(&swarm_cs4297a_mutex); return ret; } diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 8cd73cdd88af..643f1113b1d8 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -145,7 +145,6 @@ #include #include -#include #include #include #include @@ -160,6 +159,7 @@ #ifdef VWSND_DEBUG +static DEFINE_MUTEX(vwsnd_mutex); static int shut_up = 1; /* @@ -2891,11 +2891,11 @@ static long vwsnd_audio_ioctl(struct file *file, vwsnd_dev_t *devc = (vwsnd_dev_t *) file->private_data; int ret; - lock_kernel(); + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->io_mutex); ret = vwsnd_audio_do_ioctl(file, cmd, arg); mutex_unlock(&devc->io_mutex); - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return ret; } @@ -2922,7 +2922,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) DBGE("(inode=0x%p, file=0x%p)\n", inode, file); - lock_kernel(); + mutex_lock(&vwsnd_mutex); INC_USE_COUNT; for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) if ((devc->audio_minor & ~0x0F) == (minor & ~0x0F)) @@ -2930,7 +2930,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) if (devc == NULL) { DEC_USE_COUNT; - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return -ENODEV; } @@ -2939,13 +2939,13 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) mutex_unlock(&devc->open_mutex); if (file->f_flags & O_NONBLOCK) { DEC_USE_COUNT; - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return -EBUSY; } interruptible_sleep_on(&devc->open_wait); if (signal_pending(current)) { DEC_USE_COUNT; - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return -ERESTARTSYS; } mutex_lock(&devc->open_mutex); @@ -2998,7 +2998,7 @@ static int vwsnd_audio_open(struct inode *inode, struct file *file) file->private_data = devc; DBGRV(); - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return 0; } @@ -3012,7 +3012,7 @@ static int vwsnd_audio_release(struct inode *inode, struct file *file) vwsnd_port_t *wport = NULL, *rport = NULL; int err = 0; - lock_kernel(); + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->io_mutex); { DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); @@ -3040,7 +3040,7 @@ static int vwsnd_audio_release(struct inode *inode, struct file *file) wake_up(&devc->open_wait); DEC_USE_COUNT; DBGR(); - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return err; } @@ -3068,18 +3068,18 @@ static int vwsnd_mixer_open(struct inode *inode, struct file *file) DBGEV("(inode=0x%p, file=0x%p)\n", inode, file); INC_USE_COUNT; - lock_kernel(); + mutex_lock(&vwsnd_mutex); for (devc = vwsnd_dev_list; devc; devc = devc->next_dev) if (devc->mixer_minor == iminor(inode)) break; if (devc == NULL) { DEC_USE_COUNT; - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return -ENODEV; } file->private_data = devc; - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return 0; } @@ -3223,7 +3223,7 @@ static long vwsnd_mixer_ioctl(struct file *file, DBGEV("(devc=0x%p, cmd=0x%x, arg=0x%lx)\n", devc, cmd, arg); - lock_kernel(); + mutex_lock(&vwsnd_mutex); mutex_lock(&devc->mix_mutex); { if ((cmd & ~nrmask) == MIXER_READ(0)) @@ -3234,7 +3234,7 @@ static long vwsnd_mixer_ioctl(struct file *file, retval = -EINVAL; } mutex_unlock(&devc->mix_mutex); - unlock_kernel(); + mutex_unlock(&vwsnd_mutex); return retval; } -- cgit v1.2.3 From 9ad0e496519d99eb2c34f01e41500a775122c744 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 14 Sep 2010 23:22:00 +0200 Subject: ALSA: hda - Add input jack layer support to Realtek codec Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 98 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 98 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9c2c19c8b059..3b040870365d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -28,6 +28,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" @@ -282,6 +283,12 @@ struct alc_mic_route { unsigned char amix_idx; }; +struct alc_jack { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + #define MUX_IDX_UNDEF ((unsigned char)-1) struct alc_customize_define { @@ -357,6 +364,9 @@ struct alc_spec { /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ + /* jack detection */ + struct snd_array jacks; + /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct alc_customize_define cdefine; @@ -990,6 +1000,91 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, alc_fix_pll(codec); } +#ifdef CONFIG_SND_HDA_INPUT_JACK +static void alc_free_jack_priv(struct snd_jack *jack) +{ + struct alc_jack *jacks = jack->private_data; + jacks->nid = 0; + jacks->jack = NULL; +} + +static int alc_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct alc_spec *spec; + struct alc_jack *jack; + const char *name; + int err; + + spec = codec->spec; + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + if (!jack) + return -ENOMEM; + + jack->nid = nid; + jack->type = type; + name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; + + err = snd_jack_new(codec->bus->card, name, type, &jack->jack); + if (err < 0) + return err; + jack->jack->private_data = jack; + jack->jack->private_free = alc_free_jack_priv; + return 0; +} + +static void alc_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct alc_spec *spec = codec->spec; + struct alc_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int present; + present = snd_hda_jack_detect(codec, nid); + + present = (present) ? jacks->type : 0; + + snd_jack_report(jacks->jack, present); + } + jacks++; + } + } +} + +static int alc_init_jacks(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + unsigned int hp_nid = spec->autocfg.hp_pins[0]; + unsigned int mic_nid = spec->ext_mic.pin; + + err = alc_add_jack(codec, hp_nid, SND_JACK_HEADPHONE); + if (err < 0) + return err; + alc_report_jack(codec, hp_nid); + + err = alc_add_jack(codec, mic_nid, SND_JACK_MICROPHONE); + if (err < 0) + return err; + alc_report_jack(codec, mic_nid); + + return 0; +} +#else +static inline void alc_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ +} + +static inline int alc_init_jacks(struct hda_codec *codec) +{ + return 0; +} +#endif + static void alc_automute_speaker(struct hda_codec *codec, int pinctl) { struct alc_spec *spec = codec->spec; @@ -1006,6 +1101,7 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) spec->jack_present = 1; break; } + alc_report_jack(codec, spec->autocfg.hp_pins[i]); } mute = spec->jack_present ? HDA_AMP_MUTE : 0; @@ -1111,6 +1207,7 @@ static void alc_mic_automute(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, alive->mux_idx); } + alc_report_jack(codec, spec->ext_mic.pin); /* FIXME: analog mixer */ } @@ -14496,6 +14593,7 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); alc_auto_init_digital(codec); + alc_init_jacks(codec); if (spec->unsol_event) alc_inithook(codec); } -- cgit v1.2.3 From 3894335876a6257ac46e14845bd37ae6fb0f7c87 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 14 Sep 2010 10:48:59 -0600 Subject: ALSA: patch_nvhdmi.c: Fix supported sample rate list. 22050 isn't a valid HDMI sample rate. 32000 is. Signed-off-by: Stephen Warren Acked-By: Wei Ni Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 69b950d527c3..baa108b9d6aa 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -84,7 +84,7 @@ static struct hda_verb nvhdmi_basic_init_7x[] = { #else /* support all rates and formats */ #define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ SNDRV_PCM_RATE_192000) #define SUPPORTED_MAXBPS 24 -- cgit v1.2.3 From 1446c5fba73044a1c72153e1203b23b1820431c5 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 15 Sep 2010 08:01:57 +0200 Subject: ALSA: snd-aloop - fix the "PCM Playback Channels" kcontrol Obvious copy-and-paste error. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 3123a15d23f6..f2b8f868d97a 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -809,7 +809,7 @@ static int loopback_channels_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = loopback->setup[kcontrol->id.subdevice] - [kcontrol->id.device].rate; + [kcontrol->id.device].channels; return 0; } -- cgit v1.2.3 From 977ddd6b2e63716cfefe669bbdb30ec0bcea1fe4 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 15 Sep 2010 10:02:29 +0200 Subject: ALSA: hda - Set up COEFs for ALC269 to avoid click noises at power-saving For avoiding the click noises at power-saving, set some COEF values for ALC269* codecs. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 114 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 114 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3b040870365d..ab2947d87232 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1673,6 +1673,15 @@ static int alc_read_coef_idx(struct hda_codec *codec, return val; } +static void alc_write_coef_idx(struct hda_codec *codec, unsigned int coef_idx, + unsigned int coef_val) +{ + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, + coef_idx); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, + coef_val); +} + /* set right pin controls for digital I/O */ static void alc_auto_init_digital(struct hda_codec *codec) { @@ -14598,6 +14607,68 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int alc269_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct alc_spec *spec = codec->spec; + int val; + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { + val = alc_read_coef_idx(codec, 0x04); + /* Power down output pin */ + alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); + } + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + val = alc_read_coef_idx(codec, 0x04); + /* Power down output pin */ + alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); + msleep(150); + } + + alc_shutup(codec); + if (spec && spec->power_hook) + spec->power_hook(codec); + return 0; +} +#endif +#ifdef SND_HDA_NEEDS_RESUME +static int alc269_resume(struct hda_codec *codec) +{ + int val; + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + val = alc_read_coef_idx(codec, 0x04); + /* Power down output pin */ + alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); + msleep(150); + } + + codec->patch_ops.init(codec); + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { + val = alc_read_coef_idx(codec, 0x04); + /* Power up output pin */ + alc_write_coef_idx(codec, 0x04, val | (1<<11)); + msleep(200); + } + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + val = alc_read_coef_idx(codec, 0x04); + /* Power up output pin */ + alc_write_coef_idx(codec, 0x04, val | (1<<11)); + } + + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (codec->patch_ops.check_power_status) + codec->patch_ops.check_power_status(codec, 0x01); +#endif + return 0; +} +#endif + enum { ALC269_FIXUP_SONY_VAIO, }; @@ -14814,6 +14885,41 @@ static struct alc_config_preset alc269_presets[] = { }, }; +static int alc269_fill_coef(struct hda_codec *codec) +{ + int val; + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) < 0x015) { + alc_write_coef_idx(codec, 0xf, 0x960b); + alc_write_coef_idx(codec, 0xe, 0x8817); + } + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x016) { + alc_write_coef_idx(codec, 0xf, 0x960b); + alc_write_coef_idx(codec, 0xe, 0x8814); + } + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { + val = alc_read_coef_idx(codec, 0x04); + /* Power up output pin */ + alc_write_coef_idx(codec, 0x04, val | (1<<11)); + } + + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { + val = alc_read_coef_idx(codec, 0xd); + if ((val & 0x0c00) >> 10 != 0x1) { + /* Capless ramp up clock control */ + alc_write_coef_idx(codec, 0xd, val | 1<<10); + } + val = alc_read_coef_idx(codec, 0x17); + if ((val & 0x01c0) >> 6 != 0x4) { + /* Class D power on reset */ + alc_write_coef_idx(codec, 0x17, val | 1<<7); + } + } + return 0; +} + static int patch_alc269(struct hda_codec *codec) { struct alc_spec *spec; @@ -14839,6 +14945,8 @@ static int patch_alc269(struct hda_codec *codec) } else alc_fix_pll_init(codec, 0x20, 0x04, 15); + alc269_fill_coef(codec); + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -14917,6 +15025,12 @@ static int patch_alc269(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; +#ifdef CONFIG_SND_HDA_POWER_SAVE + codec->patch_ops.suspend = alc269_suspend; +#endif +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops.resume = alc269_resume; +#endif if (board_config == ALC269_AUTO) spec->init_hook = alc269_auto_init; #ifdef CONFIG_SND_HDA_POWER_SAVE -- cgit v1.2.3 From f2e5731dfd3ba08b023d0626d36ccf78f54ab5e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Sep 2010 10:07:08 +0200 Subject: ALSA: hda - Preliminary support for new Conexant audio codecs This patch adds the preliminary support for new Conexant audio codecs with 14f1:5097, 14f1:5098, 14f1:50a1, 14f1:50a2, 14f1:50ab, 14f1:50ac, 14f1:50b8 and 14f1:50b9. Unlike other Conexant parsers, this is designed to be mostly automatic, parsing from BIOS pin configurations. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 640 ++++++++++++++++++++++++++++++++++++++++- 1 file changed, 639 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 972e7c453b3d..7eee0ff65ac9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -57,6 +57,12 @@ struct conexant_jack { }; +struct pin_dac_pair { + hda_nid_t pin; + hda_nid_t dac; + int type; +}; + struct conexant_spec { struct snd_kcontrol_new *mixers[5]; @@ -77,6 +83,7 @@ struct conexant_spec { unsigned int cur_eapd; unsigned int hp_present; unsigned int auto_mic; + int auto_mic_ext; /* autocfg.inputs[] index for ext mic */ unsigned int need_dac_fix; /* capture */ @@ -110,9 +117,12 @@ struct conexant_spec { struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + struct pin_dac_pair dac_info[8]; + int dac_info_filled; - unsigned int dell_automute; unsigned int port_d_mode; + unsigned int auto_mute:1; /* used in auto-parser */ + unsigned int dell_automute:1; unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; @@ -3253,6 +3263,610 @@ static int patch_cxt5066(struct hda_codec *codec) return 0; } +/* + * Automatic parser for CX20641 & co + */ + +static hda_nid_t cx_auto_adc_nids[] = { 0x14 }; + +/* get the connection index of @nid in the widget @mux */ +static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t nid) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int i, nums; + + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + for (i = 0; i < nums; i++) + if (conn[i] == nid) + return i; + return -1; +} + +/* get an unassigned DAC from the given list. + * Return the nid if found and reduce the DAC list, or return zero if + * not found + */ +static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin, + hda_nid_t *dacs, int *num_dacs) +{ + int i, nums = *num_dacs; + hda_nid_t ret = 0; + + for (i = 0; i < nums; i++) { + if (get_connection_index(codec, pin, dacs[i]) >= 0) { + ret = dacs[i]; + break; + } + } + if (!ret) + return 0; + if (--nums > 0) + memmove(dacs, dacs + 1, nums * sizeof(hda_nid_t)); + *num_dacs = nums; + return ret; +} + +#define MAX_AUTO_DACS 5 + +/* fill analog DAC list from the widget tree */ +static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) +{ + hda_nid_t nid, end_nid; + int nums = 0; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int type = get_wcaps_type(wcaps); + if (type == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) { + dacs[nums++] = nid; + if (nums >= MAX_AUTO_DACS) + break; + } + } + return nums; +} + +/* fill pin_dac_pair list from the pin and dac list */ +static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, + int num_pins, hda_nid_t *dacs, int *rest, + struct pin_dac_pair *filled, int type) +{ + int i, nums; + + nums = 0; + for (i = 0; i < num_pins; i++) { + filled[nums].pin = pins[i]; + filled[nums].type = type; + filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); + nums++; + } + return nums; +} + +/* parse analog output paths */ +static void cx_auto_parse_output(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t dacs[MAX_AUTO_DACS]; + int i, j, nums, rest; + + rest = fill_cx_auto_dacs(codec, dacs); + /* parse all analog output pins */ + nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, + dacs, &rest, spec->dac_info, + AUTO_PIN_LINE_OUT); + nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info + nums, + AUTO_PIN_HP_OUT); + nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info + nums, + AUTO_PIN_SPEAKER_OUT); + spec->dac_info_filled = nums; + /* fill multiout struct */ + for (i = 0; i < nums; i++) { + hda_nid_t dac = spec->dac_info[i].dac; + if (!dac) + continue; + switch (spec->dac_info[i].type) { + case AUTO_PIN_LINE_OUT: + spec->private_dac_nids[spec->multiout.num_dacs] = dac; + spec->multiout.num_dacs++; + break; + case AUTO_PIN_HP_OUT: + case AUTO_PIN_SPEAKER_OUT: + if (!spec->multiout.hp_nid) { + spec->multiout.hp_nid = dac; + break; + } + for (j = 0; j < ARRAY_SIZE(spec->multiout.extra_out_nid); j++) + if (!spec->multiout.extra_out_nid[j]) { + spec->multiout.extra_out_nid[j] = dac; + break; + } + break; + } + } + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.max_channels = nums * 2; + + if (cfg->hp_outs > 0) + spec->auto_mute = 1; + spec->vmaster_nid = spec->private_dac_nids[0]; +} + +/* auto-mute/unmute speaker and line outs according to headphone jack */ +static void cx_auto_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, present; + + if (!spec->auto_mute) + return; + present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + if (snd_hda_jack_detect(codec, cfg->hp_pins[i])) { + present = 1; + break; + } + } + for (i = 0; i < cfg->line_outs; i++) { + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); + } + for (i = 0; i < cfg->speaker_outs; i++) { + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); + } +} + +/* automatic switch internal and external mic */ +static void cx_auto_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->private_imux; + int ext_idx = spec->auto_mic_ext; + + if (!spec->auto_mic) + return; + if (snd_hda_jack_detect(codec, cfg->inputs[ext_idx].pin)) { + snd_hda_codec_write(codec, spec->adc_nids[0], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[ext_idx].index); + } else { + snd_hda_codec_write(codec, spec->adc_nids[0], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[!ext_idx].index); + } +} + +static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) +{ + int nid = (res & AC_UNSOL_RES_SUBTAG) >> 20; + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cx_auto_hp_automute(codec); + conexant_report_jack(codec, nid); + break; + case CONEXANT_MIC_EVENT: + cx_auto_automic(codec); + conexant_report_jack(codec, nid); + break; + } +} + +static int is_int_mic_conn(unsigned int def_conf) +{ + unsigned int loc = get_defcfg_location(def_conf); + return get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || + (loc & 0x30) == AC_JACK_LOC_INTERNAL; +} + +/* return true if it's an internal-mic pin */ +static int is_int_mic(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); + return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && + is_int_mic_conn(def_conf); +} + +/* return true if it's an external-mic pin */ +static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) +{ + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); + return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && + !is_int_mic_conn(def_conf); +} + +/* check whether the pin config is suitable for auto-mic switching; + * auto-mic is enabled only when one int-mic and one-ext mic exist + */ +static void cx_auto_check_auto_mic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (is_ext_mic(codec, cfg->inputs[0].pin) && + is_int_mic(codec, cfg->inputs[1].pin)) { + spec->auto_mic = 1; + spec->auto_mic_ext = 1; + return; + } + if (is_int_mic(codec, cfg->inputs[1].pin) && + is_ext_mic(codec, cfg->inputs[0].pin)) { + spec->auto_mic = 1; + spec->auto_mic_ext = 0; + return; + } +} + +static void cx_auto_parse_input(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux; + int i; + + imux = &spec->private_imux; + for (i = 0; i < cfg->num_inputs; i++) { + int idx = get_connection_index(codec, spec->adc_nids[0], + cfg->inputs[i].pin); + if (idx >= 0) { + const char *label; + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, idx, NULL); + } + } + if (imux->num_items == 2 && cfg->num_inputs == 2) + cx_auto_check_auto_mic(codec); + if (imux->num_items > 1 && !spec->auto_mic) + spec->input_mux = imux; +} + +/* get digital-input audio widget corresponding to the given pin */ +static hda_nid_t cx_auto_get_dig_in(struct hda_codec *codec, hda_nid_t pin) +{ + hda_nid_t nid, end_nid; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int type = get_wcaps_type(wcaps); + if (type == AC_WID_AUD_IN && (wcaps & AC_WCAP_DIGITAL)) { + if (get_connection_index(codec, nid, pin) >= 0) + return nid; + } + } + return 0; +} + +static void cx_auto_parse_digital(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + + if (cfg->dig_outs && + snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) == 1) + spec->multiout.dig_out_nid = nid; + if (cfg->dig_in_pin) + spec->dig_in_nid = cx_auto_get_dig_in(codec, cfg->dig_in_pin); +} + +#ifdef CONFIG_SND_HDA_INPUT_BEEP +static void cx_auto_parse_beep(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + hda_nid_t nid, end_nid; + + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) + if (get_wcaps_type(get_wcaps(codec, nid)) == AC_WID_BEEP) { + set_beep_amp(spec, nid, 0, HDA_OUTPUT); + break; + } +} +#else +#define cx_auto_parse_beep(codec) +#endif + +static int cx_auto_parse_auto_config(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + cx_auto_parse_output(codec); + cx_auto_parse_input(codec); + cx_auto_parse_digital(codec); + cx_auto_parse_beep(codec); + return 0; +} + +static void cx_auto_turn_on_eapd(struct hda_codec *codec, int num_pins, + hda_nid_t *pins) +{ + int i; + for (i = 0; i < num_pins; i++) { + if (snd_hda_query_pin_caps(codec, pins[i]) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); + } +} + +static void select_connection(struct hda_codec *codec, hda_nid_t pin, + hda_nid_t src) +{ + int idx = get_connection_index(codec, pin, src); + if (idx >= 0) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_CONNECT_SEL, idx); +} + +static void cx_auto_init_output(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + for (i = 0; i < cfg->hp_outs; i++) + snd_hda_codec_write(codec, cfg->hp_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (spec->auto_mute) { + for (i = 0; i < cfg->hp_outs; i++) { + snd_hda_codec_write(codec, cfg->hp_pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | CONEXANT_HP_EVENT); + } + cx_auto_hp_automute(codec); + } else { + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + } + + for (i = 0; i < spec->dac_info_filled; i++) { + nid = spec->dac_info[i].dac; + if (!nid) + nid = spec->multiout.dac_nids[0]; + select_connection(codec, spec->dac_info[i].pin, nid); + } + + /* turn on EAPD */ + cx_auto_turn_on_eapd(codec, cfg->line_outs, cfg->line_out_pins); + cx_auto_turn_on_eapd(codec, cfg->hp_outs, cfg->hp_pins); + cx_auto_turn_on_eapd(codec, cfg->speaker_outs, cfg->speaker_pins); +} + +static void cx_auto_init_input(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < spec->num_adc_nids; i++) + snd_hda_codec_write(codec, spec->adc_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)); + + for (i = 0; i < cfg->num_inputs; i++) { + unsigned int type; + if (cfg->inputs[i].type == AUTO_PIN_MIC) + type = PIN_VREF80; + else + type = PIN_IN; + snd_hda_codec_write(codec, cfg->inputs[i].pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, type); + } + + if (spec->auto_mic) { + int ext_idx = spec->auto_mic_ext; + snd_hda_codec_write(codec, cfg->inputs[ext_idx].pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | CONEXANT_MIC_EVENT); + cx_auto_automic(codec); + } else { + for (i = 0; i < spec->num_adc_nids; i++) { + snd_hda_codec_write(codec, spec->adc_nids[i], 0, + AC_VERB_SET_CONNECT_SEL, + spec->private_imux.items[0].index); + } + } +} + +static void cx_auto_init_digital(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (spec->multiout.dig_out_nid) + snd_hda_codec_write(codec, cfg->dig_out_pins[0], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + if (spec->dig_in_nid) + snd_hda_codec_write(codec, cfg->dig_in_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); +} + +static int cx_auto_init(struct hda_codec *codec) +{ + /*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/ + cx_auto_init_output(codec); + cx_auto_init_input(codec); + cx_auto_init_digital(codec); + return 0; +} + +static int cx_auto_add_volume(struct hda_codec *codec, const char *basename, + const char *dir, int cidx, + hda_nid_t nid, int hda_dir) +{ + static char name[32]; + static struct snd_kcontrol_new knew[] = { + HDA_CODEC_VOLUME(name, 0, 0, 0), + HDA_CODEC_MUTE(name, 0, 0, 0), + }; + static char *sfx[2] = { "Volume", "Switch" }; + int i, err; + + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir); + knew[i].subdevice = HDA_SUBDEV_AMP_FLAG; + knew[i].index = cidx; + snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]); + kctl = snd_ctl_new1(&knew[i], codec); + if (!kctl) + return -ENOMEM; + err = snd_hda_ctl_add(codec, nid, kctl); + if (err < 0) + return err; + if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE)) + break; + } + return 0; +} + +#define cx_auto_add_pb_volume(codec, nid, str, idx) \ + cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT) + +static int cx_auto_build_output_controls(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + int i, err; + int num_line = 0, num_hp = 0, num_spk = 0; + static const char *texts[3] = { "Front", "Surround", "CLFE" }; + + if (spec->dac_info_filled == 1) + return cx_auto_add_pb_volume(codec, spec->dac_info[0].dac, + "Master", 0); + for (i = 0; i < spec->dac_info_filled; i++) { + const char *label; + int idx, type; + if (!spec->dac_info[i].dac) + continue; + type = spec->dac_info[i].type; + if (type == AUTO_PIN_LINE_OUT) + type = spec->autocfg.line_out_type; + switch (type) { + case AUTO_PIN_LINE_OUT: + default: + label = texts[num_line++]; + idx = 0; + break; + case AUTO_PIN_HP_OUT: + label = "Headphone"; + idx = num_hp++; + break; + case AUTO_PIN_SPEAKER_OUT: + label = "Speaker"; + idx = num_spk++; + break; + } + err = cx_auto_add_pb_volume(codec, spec->dac_info[i].dac, + label, idx); + if (err < 0) + return err; + } + return 0; +} + +static int cx_auto_build_input_controls(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + static const char *prev_label; + int i, err, cidx; + + err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0], + HDA_INPUT); + if (err < 0) + return err; + prev_label = NULL; + cidx = 0; + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t nid = cfg->inputs[i].pin; + const char *label; + if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) + continue; + label = hda_get_autocfg_input_label(codec, cfg, i); + if (label == prev_label) + cidx++; + else + cidx = 0; + prev_label = label; + err = cx_auto_add_volume(codec, label, " Capture", cidx, + nid, HDA_INPUT); + if (err < 0) + return err; + } + return 0; +} + +static int cx_auto_build_controls(struct hda_codec *codec) +{ + int err; + + err = cx_auto_build_output_controls(codec); + if (err < 0) + return err; + err = cx_auto_build_input_controls(codec); + if (err < 0) + return err; + return conexant_build_controls(codec); +} + +static struct hda_codec_ops cx_auto_patch_ops = { + .build_controls = cx_auto_build_controls, + .build_pcms = conexant_build_pcms, + .init = cx_auto_init, + .free = conexant_free, + .unsol_event = cx_auto_unsol_event, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = conexant_suspend, +#endif + .reboot_notify = snd_hda_shutup_pins, +}; + +static int patch_conexant_auto(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + spec->adc_nids = cx_auto_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(cx_auto_adc_nids); + spec->capsrc_nids = spec->adc_nids; + err = cx_auto_parse_auto_config(codec); + if (err < 0) { + kfree(codec->spec); + codec->spec = NULL; + return err; + } + codec->patch_ops = cx_auto_patch_ops; + if (spec->beep_amp) + snd_hda_attach_beep_device(codec, spec->beep_amp); + return 0; +} + /* */ @@ -3271,6 +3885,22 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f15097, .name = "CX20631", + .patch = patch_conexant_auto }, + { .id = 0x14f15098, .name = "CX20632", + .patch = patch_conexant_auto }, + { .id = 0x14f150a1, .name = "CX20641", + .patch = patch_conexant_auto }, + { .id = 0x14f150a2, .name = "CX20642", + .patch = patch_conexant_auto }, + { .id = 0x14f150ab, .name = "CX20651", + .patch = patch_conexant_auto }, + { .id = 0x14f150ac, .name = "CX20652", + .patch = patch_conexant_auto }, + { .id = 0x14f150b8, .name = "CX20664", + .patch = patch_conexant_auto }, + { .id = 0x14f150b9, .name = "CX20665", + .patch = patch_conexant_auto }, {} /* terminator */ }; @@ -3281,6 +3911,14 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f15097"); +MODULE_ALIAS("snd-hda-codec-id:14f15098"); +MODULE_ALIAS("snd-hda-codec-id:14f150a1"); +MODULE_ALIAS("snd-hda-codec-id:14f150a2"); +MODULE_ALIAS("snd-hda-codec-id:14f150ab"); +MODULE_ALIAS("snd-hda-codec-id:14f150ac"); +MODULE_ALIAS("snd-hda-codec-id:14f150b8"); +MODULE_ALIAS("snd-hda-codec-id:14f150b9"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- cgit v1.2.3 From b686453543fd56332e8730a2abd7bf5bca756149 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Sep 2010 10:17:26 +0200 Subject: ALSA: hda - Reduce pci id list for Intel with class id Most of Intel controllers work as generic HD-audio without quirks, and it'll be hopefully so in future. Let's mark pci id with the PCI_CLASS_MULTIMEDIA_HD_AUDIO for Intel so that the driver will work with any new control chips in future. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 34940a079051..5f6f9039a41a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2735,25 +2735,17 @@ static void __devexit azx_remove(struct pci_dev *pci) /* PCI IDs */ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { - /* ICH 6..10 */ - { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x269a), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x284b), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x2911), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x293e), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, - /* PCH */ - { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, - { PCI_DEVICE(0x8086, 0x3b57), .driver_data = AZX_DRIVER_ICH }, /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, + /* Generic Intel */ + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_ICH }, /* ATI SB 450/600 */ { PCI_DEVICE(0x1002, 0x437b), .driver_data = AZX_DRIVER_ATI }, { PCI_DEVICE(0x1002, 0x4383), .driver_data = AZX_DRIVER_ATI }, -- cgit v1.2.3 From 3b119f662d9054d734e3c74d662e7de6d7b35687 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Sep 2010 15:53:50 +0200 Subject: ALSA: hda - Add quirk for Acer laptop with CX20585 codec Its pin configuration is compatible with ideapad. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7eee0ff65ac9..7e22ed14ae8d 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3098,6 +3098,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), -- cgit v1.2.3 From 1a4e34e67c6de2385d9d493e69c0f6fce886b14d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 07:42:34 +0200 Subject: ALSA: usb-audio - Fix an unused-variable compile warning Used only when CONFIG_SND_DEBUG=y sound/usb/mixer.c: In function 'get_min_max': sound/usb/mixer.c:762: warning: unused variable 'chip' Reported-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 3ed3901369ce..5f12e294b0f8 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -759,8 +759,6 @@ static void usb_mixer_elem_free(struct snd_kcontrol *kctl) */ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) { - struct snd_usb_audio *chip = cval->mixer->chip; - /* for failsafe */ cval->min = default_min; cval->max = cval->min + 1; @@ -783,7 +781,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", - cval->id, snd_usb_ctrl_intf(chip), cval->control, cval->id); + cval->id, snd_usb_ctrl_intf(cval->mixer->chip), cval->control, cval->id); return -EINVAL; } if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { -- cgit v1.2.3 From 4437ecdc7190302ed02fb1467c116aff29c325b2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Sep 2010 10:26:54 +0300 Subject: ALSA: core: Allow card id change to the same string When user want to change the card id to the same string on the card via /sys/class/sound/cardX/id, do not report error. Instead return with success without doing anything. Signed-off-by: Peter Ujfalusi Signed-off-by: Takashi Iwai --- sound/core/init.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index ec4a50ce5656..2de45fbd70fb 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -607,11 +607,16 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return -EEXIST; } for (idx = 0; idx < snd_ecards_limit; idx++) { - if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) - goto __exist; + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) { + if (card == snd_cards[idx]) + goto __ok; + else + goto __exist; + } } strcpy(card->id, buf1); snd_info_card_id_change(card); +__ok: mutex_unlock(&snd_card_mutex); return count; -- cgit v1.2.3 From 145a902bfeb1f89a41165bd2d1e633ce070bcb73 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 16 Sep 2010 10:07:53 +0200 Subject: ALSA: HDA: Enable internal speaker on Dell M101z BugLink: http://launchpad.net/bugs/640254 In some cases a magic processing coefficient is needed to enable the internal speaker on Dell M101z. According to Realtek, this processing coefficient is only present on ALC269vb. Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bcbf9160ed81..a1312a6c8af2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14453,6 +14453,7 @@ static void alc269_auto_init(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, + ALC269_FIXUP_DELL_M101Z, }; static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { @@ -14464,11 +14465,20 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_SONY_VAIO] = { .verbs = alc269_sony_vaio_fixup_verbs }, + [ALC269_FIXUP_DELL_M101Z] = { + .verbs = (const struct hda_verb[]) { + /* Enables internal speaker */ + {0x20, AC_VERB_SET_COEF_INDEX, 13}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4040}, + {} + } + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; -- cgit v1.2.3 From 5855fb8076e784a657bc2441cd29f166c7c1ea8c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 18:24:02 +0200 Subject: ALSA: hda - Fix initialization of secondary headphone and speaker The secondary or later headphones or speakers aren't initialized preoprly for some codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab2947d87232..945826da23b6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10634,16 +10634,21 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t pin, dac; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + pin = spec->autocfg.hp_pins[i]; + if (!pin) + break; dac = spec->multiout.hp_nid; if (!dac) dac = spec->multiout.dac_nids[0]; /* to front */ alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); } - pin = spec->autocfg.speaker_pins[0]; - if (pin) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + pin = spec->autocfg.speaker_pins[i]; + if (!pin) + break; dac = spec->multiout.extra_out_nid[0]; if (!dac) dac = spec->multiout.dac_nids[0]; /* to front */ -- cgit v1.2.3 From 30ea098fc000bb05081a1999269658f1a88af36a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 18:47:56 +0200 Subject: ALSA: hda - Fix input-pin setup for Realtek codecs Through the transition of autocfg to individual inputs array, I forgot to rewrite the argument passed to alc_set_input_pin(). This resulted in wrongly setup input pins. Fixed now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 945826da23b6..5df88798895b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5160,7 +5160,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); if (nid != ALC880_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, @@ -6793,7 +6793,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; if (nid >= 0x12) { - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); if (nid != ALC260_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, @@ -10664,7 +10664,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -15856,7 +15856,7 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; if (nid >= 0x0c && nid <= 0x11) - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); } } @@ -16849,7 +16849,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); if (nid != ALC861VD_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, @@ -19086,7 +19086,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; if (alc_is_input_pin(codec, nid)) { - alc_set_input_pin(codec, nid, i); + alc_set_input_pin(codec, nid, cfg->inputs[i].type); if (nid != ALC662_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, -- cgit v1.2.3 From 26e34e9e15aa48e9375ea4e97bc4234ad995b7c8 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 16 Sep 2010 20:13:47 +0200 Subject: ALSA: usb/mixer: remove bogus cast "uinfo->value.enumerated.item" is an unsigned int. If it's negative when we do the comparison: if ((int)uinfo->value.enumerated.item >= cval->max) then we would read past the end of the array on the next line. I also changed the strcpy() to strlcpy() out of paranoia. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 5f12e294b0f8..f2d74d654b3c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1640,9 +1640,10 @@ static int mixer_ctl_selector_info(struct snd_kcontrol *kcontrol, struct snd_ctl uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = cval->max; - if ((int)uinfo->value.enumerated.item >= cval->max) + if (uinfo->value.enumerated.item >= cval->max) uinfo->value.enumerated.item = cval->max - 1; - strcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item]); + strlcpy(uinfo->value.enumerated.name, itemlist[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); return 0; } -- cgit v1.2.3 From 8699a0b657b43fa6401537dfe345bee7aa8115ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 22:52:32 +0200 Subject: ALSA: pcm - Fix unbalanced pm_qos_request The pm_qos_request isn't freed properly when OSS PCM emulation is used because it skips snd_pcm_hw_free() call but directly releases the stream. This resulted in Oops later. Tested-by: Simon Kirby Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 134fc6c2e08d..d4eb2ef80784 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1992,6 +1992,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) substream->ops->close(substream); substream->hw_opened = 0; } + if (pm_qos_request_active(&substream->latency_pm_qos_req)) + pm_qos_remove_request(&substream->latency_pm_qos_req); if (substream->pcm_release) { substream->pcm_release(substream); substream->pcm_release = NULL; -- cgit v1.2.3 From 901d46d5a8eb821b03ca9e8cf005beb0c92f31ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Sep 2010 23:06:50 +0200 Subject: ALSA: pcm - Fix race with proc files The PCM proc files may open a race against substream close, which can end up with an Oops. Use the open_mutex to protect for it. Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 33 ++++++++++++++++++++++++--------- 1 file changed, 24 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 204af48c5cc1..ac242a377aea 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -372,14 +372,17 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format)); @@ -398,20 +401,25 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "OSS period frames: %lu\n", (unsigned long)runtime->oss.period_frames); } #endif + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } if (runtime->status->state == SNDRV_PCM_STATE_OPEN) { snd_iprintf(buffer, "no setup\n"); - return; + goto unlock; } snd_iprintf(buffer, "tstamp_mode: %s\n", snd_pcm_tstamp_mode_name(runtime->tstamp_mode)); snd_iprintf(buffer, "period_step: %u\n", runtime->period_step); @@ -421,24 +429,29 @@ static void snd_pcm_substream_proc_sw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "silence_threshold: %lu\n", runtime->silence_threshold); snd_iprintf(buffer, "silence_size: %lu\n", runtime->silence_size); snd_iprintf(buffer, "boundary: %lu\n", runtime->boundary); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_substream *substream = entry->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_runtime *runtime; struct snd_pcm_status status; int err; + + mutex_lock(&substream->pcm->open_mutex); + runtime = substream->runtime; if (!runtime) { snd_iprintf(buffer, "closed\n"); - return; + goto unlock; } memset(&status, 0, sizeof(status)); err = snd_pcm_status(substream, &status); if (err < 0) { snd_iprintf(buffer, "error %d\n", err); - return; + goto unlock; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); @@ -452,6 +465,8 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); + unlock: + mutex_unlock(&substream->pcm->open_mutex); } #ifdef CONFIG_SND_PCM_XRUN_DEBUG -- cgit v1.2.3 From 9907790aa06bfc04bf78b445e732ea10039c61e4 Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Fri, 17 Sep 2010 10:22:32 +0200 Subject: ALSA: hda - Fix automatic MIC switching and include dock MIC for IDT codecs Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 40 +++++++++++++++++++++++++++++----------- 1 file changed, 29 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e4e7d43f911e..7eb359a030de 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -263,6 +263,7 @@ struct sigmatel_spec { struct sigmatel_mic_route ext_mic; struct sigmatel_mic_route int_mic; + struct sigmatel_mic_route dock_mic; const char **spdif_labels; @@ -3488,7 +3489,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, } static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t *fixed, hda_nid_t *ext) + hda_nid_t *fixed, hda_nid_t *ext, hda_nid_t *dock) { unsigned int cfg; @@ -3496,15 +3497,22 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, return 0; cfg = snd_hda_codec_get_pincfg(codec, nid); switch (get_defcfg_connect(cfg)) { + case AC_JACK_PORT_BOTH: case AC_JACK_PORT_FIXED: if (*fixed) return 1; /* already occupied */ *fixed = nid; break; case AC_JACK_PORT_COMPLEX: - if (*ext) - return 1; /* already occupied */ - *ext = nid; + if ((get_defcfg_location(cfg) & 0xF0) == AC_JACK_LOC_SEPARATE) { + if (*dock) + return 1; /* already occupied */ + *dock = nid; + } else { + if (*ext) + return 1; /* already occupied */ + *ext = nid; + } break; } return 0; @@ -3519,6 +3527,8 @@ static int set_mic_route(struct hda_codec *codec, int i; mic->pin = pin; + if (pin == 0) + return 0; for (i = 0; i < cfg->num_inputs; i++) { if (pin == cfg->inputs[i].pin) break; @@ -3554,26 +3564,29 @@ static int stac_check_auto_mic(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t fixed, ext; + hda_nid_t fixed, ext, dock; int i; for (i = 0; i < cfg->num_inputs; i++) { if (cfg->inputs[i].type >= AUTO_PIN_LINE_IN) return 0; /* must be exclusively mics */ } - fixed = ext = 0; + fixed = ext = dock = 0; for (i = 0; i < cfg->num_inputs; i++) - if (check_mic_pin(codec, cfg->inputs[i].pin, &fixed, &ext)) + if (check_mic_pin(codec, cfg->inputs[i].pin, + &fixed, &ext, &dock)) return 0; for (i = 0; i < spec->num_dmics; i++) - if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext)) + if (check_mic_pin(codec, spec->dmic_nids[i], + &fixed, &ext, &dock)) return 0; - if (!fixed || !ext) - return 0; + if (!fixed && !ext && !dock) + return 0; /* no input to switch */ if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) return 0; /* no unsol support */ if (set_mic_route(codec, &spec->ext_mic, ext) || - set_mic_route(codec, &spec->int_mic, fixed)) + set_mic_route(codec, &spec->int_mic, fixed) || + set_mic_route(codec, &spec->dock_mic, dock)) return 0; /* something is wrong */ return 1; } @@ -4281,6 +4294,9 @@ static int stac92xx_init(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, 0); if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT)) stac_issue_unsol_event(codec, spec->ext_mic.pin); + if (enable_pin_detect(codec, spec->dock_mic.pin, + STAC_MIC_EVENT)) + stac_issue_unsol_event(codec, spec->dock_mic.pin); } for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; @@ -4698,6 +4714,8 @@ static void stac92xx_mic_detect(struct hda_codec *codec) if (get_pin_presence(codec, spec->ext_mic.pin)) mic = &spec->ext_mic; + else if (get_pin_presence(codec, spec->dock_mic.pin)) + mic = &spec->dock_mic; else mic = &spec->int_mic; if (mic->dmux_idx >= 0) -- cgit v1.2.3 From 41c89ef3aafea5f35601fa75edba90e7417f604e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Sep 2010 10:26:37 +0200 Subject: ALSA: hda - Fix mic attribute check for internal mics Now Windows claims that the BIOS sets pins for internal mics to be BOTH connection instead of FIXED. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 08d81b873022..9f668efbe420 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4648,8 +4648,11 @@ enum { static int get_mic_pin_attr(unsigned int def_conf) { unsigned int loc = get_defcfg_location(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || - (loc & 0x30) == AC_JACK_LOC_INTERNAL) + unsigned int conn = get_defcfg_connect(def_conf); + /* Windows may claim the internal mic to be BOTH, too */ + if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) + return MIC_ATTR_INT; + if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) return MIC_ATTR_INT; if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) return MIC_ATTR_DOCK; -- cgit v1.2.3 From 1feba3b7367b333c3fc7deba638a3a1068f22932 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 17 Sep 2010 10:52:50 +0200 Subject: ALSA: HDA: Fix spelling (change VOSTO to VOSTRO) It was just a boring day. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 7e22ed14ae8d..a12a9f6f795a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3075,7 +3075,7 @@ enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ - CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ + CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_HP_LAPTOP, /* HP Laptop */ @@ -3086,7 +3086,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", - [CXT5066_DELL_VOSTO] = "dell-vostro", + [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", [CXT5066_HP_LAPTOP] = "hp-laptop", @@ -3099,8 +3099,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), + SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), + SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), @@ -3207,7 +3207,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->capture_prepare = cxt5066_olpc_capture_prepare; spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; - case CXT5066_DELL_VOSTO: + case CXT5066_DELL_VOSTRO: codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; -- cgit v1.2.3 From 5637edb2e1c2d13b276748508ae17f319fb7f066 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 17 Sep 2010 10:58:03 +0200 Subject: ALSA: HDA: Sort CXT5066 quirk table It was just a boring day. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a12a9f6f795a..e501a85b5612 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3093,19 +3093,19 @@ static const char *cxt5066_models[CXT5066_MODELS] = { }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", - CXT5066_LAPTOP), - SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", - CXT5066_DELL_LAPTOP), - SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), -- cgit v1.2.3 From 99ae28bea984df4c38234eb6d2f29a552def6c1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Sep 2010 14:42:34 +0200 Subject: ALSA: hda - Make snd_hda_get_input_pin_attr() helper Make the helper function to give the input-pin attribute for jack connectivity and location. This simplifies checks of input-pin jacks a bit in some places. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 62 +++++++++++++++++------------------------- sound/pci/hda/hda_local.h | 11 ++++++++ sound/pci/hda/patch_cirrus.c | 2 +- sound/pci/hda/patch_conexant.c | 11 ++------ sound/pci/hda/patch_realtek.c | 10 +++---- sound/pci/hda/patch_sigmatel.c | 30 ++++++++++---------- 6 files changed, 59 insertions(+), 67 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9f668efbe420..e15a75751f57 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4637,44 +4637,26 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_parse_pin_def_config); -enum { - MIC_ATTR_INT, - MIC_ATTR_DOCK, - MIC_ATTR_NORMAL, - MIC_ATTR_FRONT, - MIC_ATTR_REAR, -}; - -static int get_mic_pin_attr(unsigned int def_conf) +int snd_hda_get_input_pin_attr(unsigned int def_conf) { unsigned int loc = get_defcfg_location(def_conf); unsigned int conn = get_defcfg_connect(def_conf); + if (conn == AC_JACK_PORT_NONE) + return INPUT_PIN_ATTR_UNUSED; /* Windows may claim the internal mic to be BOTH, too */ if (conn == AC_JACK_PORT_FIXED || conn == AC_JACK_PORT_BOTH) - return MIC_ATTR_INT; + return INPUT_PIN_ATTR_INT; if ((loc & 0x30) == AC_JACK_LOC_INTERNAL) - return MIC_ATTR_INT; + return INPUT_PIN_ATTR_INT; if ((loc & 0x30) == AC_JACK_LOC_SEPARATE) - return MIC_ATTR_DOCK; + return INPUT_PIN_ATTR_DOCK; if (loc == AC_JACK_LOC_REAR) - return MIC_ATTR_REAR; + return INPUT_PIN_ATTR_REAR; if (loc == AC_JACK_LOC_FRONT) - return MIC_ATTR_FRONT; - return MIC_ATTR_NORMAL; -} - -enum { - LINE_ATTR_DOCK, - LINE_ATTR_NORMAL, -}; - -static int get_line_pin_attr(unsigned int def_conf) -{ - unsigned int loc = get_defcfg_location(def_conf); - if ((loc & 0xf0) == AC_JACK_LOC_SEPARATE) - return LINE_ATTR_DOCK; - return LINE_ATTR_NORMAL; + return INPUT_PIN_ATTR_FRONT; + return INPUT_PIN_ATTR_NORMAL; } +EXPORT_SYMBOL_HDA(snd_hda_get_input_pin_attr); /** * hda_get_input_pin_label - Give a label for the given input pin @@ -4691,9 +4673,7 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, static const char *mic_names[] = { "Internal Mic", "Dock Mic", "Mic", "Front Mic", "Rear Mic", }; - static const char *line_names[] = { - "Dock Line", "Line", - }; + int attr; def_conf = snd_hda_codec_get_pincfg(codec, pin); @@ -4701,11 +4681,19 @@ const char *hda_get_input_pin_label(struct hda_codec *codec, hda_nid_t pin, case AC_JACK_MIC_IN: if (!check_location) return "Mic"; - return mic_names[get_mic_pin_attr(def_conf)]; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + return mic_names[attr - 1]; case AC_JACK_LINE_IN: if (!check_location) return "Line"; - return line_names[get_line_pin_attr(def_conf)]; + attr = snd_hda_get_input_pin_attr(def_conf); + if (!attr) + return "None"; + if (attr == INPUT_PIN_ATTR_DOCK) + return "Dock Line"; + return "Line"; case AC_JACK_AUX: return "Aux"; case AC_JACK_CD: @@ -4732,16 +4720,16 @@ static int check_mic_location_need(struct hda_codec *codec, int i, attr, attr2; defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[input].pin); - attr = get_mic_pin_attr(defc); + attr = snd_hda_get_input_pin_attr(defc); /* for internal or docking mics, we need locations */ - if (attr <= MIC_ATTR_NORMAL) + if (attr <= INPUT_PIN_ATTR_NORMAL) return 1; attr = 0; for (i = 0; i < cfg->num_inputs; i++) { defc = snd_hda_codec_get_pincfg(codec, cfg->inputs[i].pin); - attr2 = get_mic_pin_attr(defc); - if (attr2 >= MIC_ATTR_NORMAL) { + attr2 = snd_hda_get_input_pin_attr(defc); + if (attr2 >= INPUT_PIN_ATTR_NORMAL) { if (attr && attr != attr2) return 1; /* different locations found */ attr = attr2; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6943efc78f66..d7dfa547e2d8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -395,6 +395,17 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec, int snd_hda_add_imux_item(struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); +enum { + INPUT_PIN_ATTR_UNUSED, /* pin not connected */ + INPUT_PIN_ATTR_INT, /* internal mic/line-in */ + INPUT_PIN_ATTR_DOCK, /* docking mic/line-in */ + INPUT_PIN_ATTR_NORMAL, /* mic/line-in jack */ + INPUT_PIN_ATTR_FRONT, /* mic/line-in jack in front */ + INPUT_PIN_ATTR_REAR, /* mic/line-in jack in rear */ +}; + +int snd_hda_get_input_pin_attr(unsigned int def_conf); + struct auto_pin_cfg { int line_outs; /* sorted in the order of Front/Surr/CLFE/Side */ diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index ae75283a5583..483c3f2d8d39 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -334,7 +334,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) if (!(val & AC_PINCAP_PRES_DETECT)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); - return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX); + return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); } static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e501a85b5612..09d573c59bef 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3462,19 +3462,12 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) } } -static int is_int_mic_conn(unsigned int def_conf) -{ - unsigned int loc = get_defcfg_location(def_conf); - return get_defcfg_connect(def_conf) == AC_JACK_PORT_FIXED || - (loc & 0x30) == AC_JACK_LOC_INTERNAL; -} - /* return true if it's an internal-mic pin */ static int is_int_mic(struct hda_codec *codec, hda_nid_t pin) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - is_int_mic_conn(def_conf); + snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT; } /* return true if it's an external-mic pin */ @@ -3482,7 +3475,7 @@ static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - !is_int_mic_conn(def_conf); + snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT; } /* check whether the pin config is suitable for auto-mic switching; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5df88798895b..6045f281b225 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1403,19 +1403,19 @@ static void alc_init_auto_mic(struct hda_codec *codec) hda_nid_t nid = cfg->inputs[i].pin; unsigned int defcfg; defcfg = snd_hda_codec_get_pincfg(codec, nid); - switch (get_defcfg_connect(defcfg)) { - case AC_JACK_PORT_FIXED: + switch (snd_hda_get_input_pin_attr(defcfg)) { + case INPUT_PIN_ATTR_INT: if (fixed) return; /* already occupied */ fixed = nid; break; - case AC_JACK_PORT_COMPLEX: + case INPUT_PIN_ATTR_UNUSED: + return; /* invalid entry */ + default: if (ext) return; /* already occupied */ ext = nid; break; - default: - return; /* invalid entry */ } } if (!ext || !fixed) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7eb359a030de..6bfbc2fe46ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2778,7 +2778,7 @@ static inline int stac92xx_add_jack_mode_control(struct hda_codec *codec, struct sigmatel_spec *spec = codec->spec; char name[22]; - if (!((get_defcfg_connect(def_conf)) & AC_JACK_PORT_FIXED)) { + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { if (stac92xx_get_default_vref(codec, nid) == AC_PINCTL_VREF_GRD && nid == spec->line_switch) control = STAC_CTL_WIDGET_IO_SWITCH; @@ -2857,7 +2857,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec, hda_nid_t *dac) def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { + if (snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT) { pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) { *dac = get_unassigned_dac(codec, nid); @@ -3496,23 +3496,23 @@ static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, if (!nid) return 0; cfg = snd_hda_codec_get_pincfg(codec, nid); - switch (get_defcfg_connect(cfg)) { - case AC_JACK_PORT_BOTH: - case AC_JACK_PORT_FIXED: + switch (snd_hda_get_input_pin_attr(cfg)) { + case INPUT_PIN_ATTR_INT: if (*fixed) return 1; /* already occupied */ *fixed = nid; break; - case AC_JACK_PORT_COMPLEX: - if ((get_defcfg_location(cfg) & 0xF0) == AC_JACK_LOC_SEPARATE) { - if (*dock) - return 1; /* already occupied */ - *dock = nid; - } else { - if (*ext) - return 1; /* already occupied */ - *ext = nid; - } + case INPUT_PIN_ATTR_UNUSED: + break; + case INPUT_PIN_ATTR_DOCK: + if (*dock) + return 1; /* already occupied */ + *dock = nid; + break; + default: + if (*ext) + return 1; /* already occupied */ + *ext = nid; break; } return 0; -- cgit v1.2.3 From f68b3b291d39f1e3361b194a95459f9cbdaf31e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Sep 2010 14:45:14 +0200 Subject: ALSA: hda - Check the external mic pin more strictly for Conexant chips The external mic jack for auto-mic switch must be really an external jack and with a presense-detection capability. This patch makes the check more paranoia. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 09d573c59bef..a6c68cb06ddb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3475,7 +3475,8 @@ static int is_ext_mic(struct hda_codec *codec, hda_nid_t pin) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, pin); return get_defcfg_device(def_conf) == AC_JACK_MIC_IN && - snd_hda_get_input_pin_attr(def_conf) != INPUT_PIN_ATTR_INT; + snd_hda_get_input_pin_attr(def_conf) >= INPUT_PIN_ATTR_NORMAL && + (snd_hda_query_pin_caps(codec, pin) & AC_PINCAP_PRES_DETECT); } /* check whether the pin config is suitable for auto-mic switching; -- cgit v1.2.3 From 84eb01be18df7012ac31bf678da5aaf1accc6a77 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Sep 2010 12:27:25 +0200 Subject: ALSA: hda - Merge all HDMI modules into the unified module This patch merges all three patch_*hdmi variants to the single HDMI parser. There is only one snd-hda-codec-hdmi module now. In this patch, the behavior of each parser isn't changed much. The old ATI parser still doesn't use the dynamic parser yet. In later patches, they'll be cleaned up. Also, this patch gets rid of the individual snd-hda-eld module and builds into snd-hda-codec-hdmi, since this is referred only from the HDMI parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 39 +-- sound/pci/hda/Makefile | 15 +- sound/pci/hda/hda_eld.c | 7 - sound/pci/hda/patch_atihdmi.c | 224 ------------- sound/pci/hda/patch_hdmi.c | 685 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/patch_intelhdmi.c | 220 ------------- sound/pci/hda/patch_nvhdmi.c | 608 ----------------------------------- 7 files changed, 691 insertions(+), 1107 deletions(-) delete mode 100644 sound/pci/hda/patch_atihdmi.c delete mode 100644 sound/pci/hda/patch_intelhdmi.c delete mode 100644 sound/pci/hda/patch_nvhdmi.c (limited to 'sound') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 9194c3c1d04a..0ea5cc60ac78 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -119,47 +119,20 @@ config SND_HDA_CODEC_VIA snd-hda-codec-via. This module is automatically loaded at probing. -config SND_HDA_CODEC_ATIHDMI - bool "Build ATI HDMI HD-audio codec support" - default y - help - Say Y here to include ATI HDMI HD-audio codec support in - snd-hda-intel driver, such as ATI RS600 HDMI. - - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-atihdmi. - This module is automatically loaded at probing. - -config SND_HDA_CODEC_NVHDMI - bool "Build NVIDIA HDMI HD-audio codec support" - default y - help - Say Y here to include NVIDIA HDMI HD-audio codec support in - snd-hda-intel driver, such as NVIDIA MCP78 HDMI. - - When the HD-audio driver is built as a module, the codec - support code is also built as another module, - snd-hda-codec-nvhdmi. - This module is automatically loaded at probing. - -config SND_HDA_CODEC_INTELHDMI - bool "Build INTEL HDMI HD-audio codec support" +config SND_HDA_CODEC_HDMI + bool "Build HDMI/DisplayPort HD-audio codec support" select SND_DYNAMIC_MINORS default y help - Say Y here to include INTEL HDMI HD-audio codec support in - snd-hda-intel driver, such as Eaglelake integrated HDMI. + Say Y here to include HDMI and DisplayPort HD-audio codec + support in snd-hda-intel driver. This includes all AMD/ATI, + Intel and Nvidia HDMI/DisplayPort codecs. When the HD-audio driver is built as a module, the codec support code is also built as another module, - snd-hda-codec-intelhdmi. + snd-hda-codec-hdmi. This module is automatically loaded at probing. -config SND_HDA_ELD - def_bool y - depends on SND_HDA_CODEC_INTELHDMI || SND_HDA_CODEC_NVHDMI - config SND_HDA_CODEC_CIRRUS bool "Build Cirrus Logic codec support" depends on SND_HDA_INTEL diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 24bc195b02da..17ef3658f34b 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -3,7 +3,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-codec-y := hda_codec.o snd-hda-codec-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o -snd-hda-codec-$(CONFIG_SND_HDA_ELD) += hda_eld.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o @@ -12,13 +11,11 @@ snd-hda-codec-cmedia-objs := patch_cmedia.o snd-hda-codec-analog-objs := patch_analog.o snd-hda-codec-idt-objs := patch_sigmatel.o snd-hda-codec-si3054-objs := patch_si3054.o -snd-hda-codec-atihdmi-objs := patch_atihdmi.o snd-hda-codec-cirrus-objs := patch_cirrus.o snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o -snd-hda-codec-nvhdmi-objs := patch_nvhdmi.o -snd-hda-codec-intelhdmi-objs := patch_intelhdmi.o +snd-hda-codec-hdmi-objs := patch_hdmi.o hda_eld.o # common driver obj-$(CONFIG_SND_HDA_INTEL) := snd-hda-codec.o @@ -39,9 +36,6 @@ endif ifdef CONFIG_SND_HDA_CODEC_SI3054 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-si3054.o endif -ifdef CONFIG_SND_HDA_CODEC_ATIHDMI -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o -endif ifdef CONFIG_SND_HDA_CODEC_CIRRUS obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o endif @@ -54,11 +48,8 @@ endif ifdef CONFIG_SND_HDA_CODEC_VIA obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-via.o endif -ifdef CONFIG_SND_HDA_CODEC_NVHDMI -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-nvhdmi.o -endif -ifdef CONFIG_SND_HDA_CODEC_INTELHDMI -obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-intelhdmi.o +ifdef CONFIG_SND_HDA_CODEC_HDMI +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-hdmi.o endif # this must be the last entry after codec drivers; diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 26c3ade73583..cb0c23a6b473 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -332,7 +332,6 @@ int snd_hdmi_get_eld_size(struct hda_codec *codec, hda_nid_t nid) return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, AC_DIPSIZE_ELD_BUF); } -EXPORT_SYMBOL_HDA(snd_hdmi_get_eld_size); int snd_hdmi_get_eld(struct hdmi_eld *eld, struct hda_codec *codec, hda_nid_t nid) @@ -368,7 +367,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, kfree(buf); return ret; } -EXPORT_SYMBOL_HDA(snd_hdmi_get_eld); static void hdmi_show_short_audio_desc(struct cea_sad *a) { @@ -407,7 +405,6 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen) } buf[j] = '\0'; /* necessary when j == 0 */ } -EXPORT_SYMBOL_HDA(snd_print_channel_allocation); void snd_hdmi_show_eld(struct hdmi_eld *e) { @@ -426,7 +423,6 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) for (i = 0; i < e->sad_count; i++) hdmi_show_short_audio_desc(e->sad + i); } -EXPORT_SYMBOL_HDA(snd_hdmi_show_eld); #ifdef CONFIG_PROC_FS @@ -585,7 +581,6 @@ int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, return 0; } -EXPORT_SYMBOL_HDA(snd_hda_eld_proc_new); void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) { @@ -594,7 +589,6 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld) eld->proc_entry = NULL; } } -EXPORT_SYMBOL_HDA(snd_hda_eld_proc_free); #endif /* CONFIG_PROC_FS */ @@ -645,4 +639,3 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max); pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps); } -EXPORT_SYMBOL_HDA(hdmi_eld_update_pcm_info); diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c deleted file mode 100644 index fb684f00156b..000000000000 --- a/sound/pci/hda/patch_atihdmi.c +++ /dev/null @@ -1,224 +0,0 @@ -/* - * Universal Interface for Intel High Definition Audio Codec - * - * HD audio interface patch for ATI HDMI codecs - * - * Copyright (c) 2006 ATI Technologies Inc. - * - * - * This driver is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include -#include -#include -#include -#include "hda_codec.h" -#include "hda_local.h" - -struct atihdmi_spec { - struct hda_multi_out multiout; - - struct hda_pcm pcm_rec; -}; - -#define CVT_NID 0x02 /* audio converter */ -#define PIN_NID 0x03 /* HDMI output pin */ - -static struct hda_verb atihdmi_basic_init[] = { - /* enable digital output on pin widget */ - { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {} /* terminator */ -}; - -/* - * Controls - */ -static int atihdmi_build_controls(struct hda_codec *codec) -{ - struct atihdmi_spec *spec = codec->spec; - int err; - - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); - if (err < 0) - return err; - - return 0; -} - -static int atihdmi_init(struct hda_codec *codec) -{ - snd_hda_sequence_write(codec, atihdmi_basic_init); - /* SI codec requires to unmute the pin */ - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); - return 0; -} - -/* - * Digital out - */ -static int atihdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct atihdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int atihdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct atihdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int atihdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct atihdmi_spec *spec = codec->spec; - int chans = substream->runtime->channels; - int i, err; - - err = snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); - if (err < 0) - return err; - snd_hda_codec_write(codec, CVT_NID, 0, AC_VERB_SET_CVT_CHAN_COUNT, - chans - 1); - /* FIXME: XXX */ - for (i = 0; i < chans; i++) { - snd_hda_codec_write(codec, CVT_NID, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - (i << 4) | i); - } - return 0; -} - -static struct hda_pcm_stream atihdmi_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = CVT_NID, /* NID to query formats and rates and setup streams */ - .ops = { - .open = atihdmi_dig_playback_pcm_open, - .close = atihdmi_dig_playback_pcm_close, - .prepare = atihdmi_dig_playback_pcm_prepare - }, -}; - -static int atihdmi_build_pcms(struct hda_codec *codec) -{ - struct atihdmi_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm_rec; - unsigned int chans; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "ATI HDMI"; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; - - /* FIXME: we must check ELD and change the PCM parameters dynamically - */ - chans = get_wcaps(codec, CVT_NID); - chans = get_wcaps_channels(chans); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; - - return 0; -} - -static void atihdmi_free(struct hda_codec *codec) -{ - kfree(codec->spec); -} - -static struct hda_codec_ops atihdmi_patch_ops = { - .build_controls = atihdmi_build_controls, - .build_pcms = atihdmi_build_pcms, - .init = atihdmi_init, - .free = atihdmi_free, -}; - -static int patch_atihdmi(struct hda_codec *codec) -{ - struct atihdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - /* NID for copying analog to digital, - * seems to be unused in pure-digital - * case. - */ - spec->multiout.dig_out_nid = CVT_NID; - - codec->patch_ops = atihdmi_patch_ops; - - return 0; -} - -/* - * patch entries - */ -static struct hda_codec_preset snd_hda_preset_atihdmi[] = { - { .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, - { .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, - { .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, - { .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, - { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi }, - { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, - {} /* terminator */ -}; - -MODULE_ALIAS("snd-hda-codec-id:1002793c"); -MODULE_ALIAS("snd-hda-codec-id:10027919"); -MODULE_ALIAS("snd-hda-codec-id:1002791a"); -MODULE_ALIAS("snd-hda-codec-id:1002aa01"); -MODULE_ALIAS("snd-hda-codec-id:10951390"); -MODULE_ALIAS("snd-hda-codec-id:17e80047"); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("ATI HDMI HD-audio codec"); - -static struct hda_codec_preset_list atihdmi_list = { - .preset = snd_hda_preset_atihdmi, - .owner = THIS_MODULE, -}; - -static int __init patch_atihdmi_init(void) -{ - return snd_hda_add_codec_preset(&atihdmi_list); -} - -static void __exit patch_atihdmi_exit(void) -{ - snd_hda_delete_codec_preset(&atihdmi_list); -} - -module_init(patch_atihdmi_init) -module_exit(patch_atihdmi_exit) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index afd6022a96a7..cb997ca0fdfa 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3,6 +3,9 @@ * patch_hdmi.c - routines for HDMI/DisplayPort codecs * * Copyright(c) 2008-2010 Intel Corporation. All rights reserved. + * Copyright (c) 2006 ATI Technologies Inc. + * Copyright (c) 2008 NVIDIA Corp. All rights reserved. + * Copyright (c) 2008 Wei Ni * * Authors: * Wu Fengguang @@ -25,6 +28,22 @@ * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +/* + * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device + * could support two independent pipes, each of them can be connected to one or + * more ports (DVI, HDMI or DisplayPort). + * + * The HDA correspondence of pipes/ports are converter/pin nodes. + */ +#define MAX_HDMI_CVTS 3 +#define MAX_HDMI_PINS 3 struct hdmi_spec { int num_cvts; @@ -49,10 +68,10 @@ struct hdmi_spec { struct hda_pcm_stream codec_pcm_pars[MAX_HDMI_CVTS]; /* - * nvhdmi specific + * ati/nvhdmi specific */ struct hda_multi_out multiout; - unsigned int codec_type; + struct hda_pcm_stream *pcm_playback; /* misc flags */ /* PD bit indicates only the update, not the current state */ @@ -791,7 +810,6 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, /* * HDA/HDMI auto parsing */ - static int hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) { struct hdmi_spec *spec = codec->spec; @@ -922,3 +940,664 @@ static int hdmi_parse_codec(struct hda_codec *codec) return 0; } +/* + */ +static char *generic_hdmi_pcm_names[MAX_HDMI_CVTS] = { + "HDMI 0", + "HDMI 1", + "HDMI 2", +}; + +/* + * HDMI callbacks + */ + +static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + hdmi_set_channel_count(codec, hinfo->nid, + substream->runtime->channels); + + hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); + + return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); +} + +static struct hda_pcm_stream generic_hdmi_pcm_playback = { + .substreams = 1, + .channels_min = 2, + .ops = { + .open = hdmi_pcm_open, + .prepare = generic_hdmi_playback_pcm_prepare, + }, +}; + +static int generic_hdmi_build_pcms(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + int i; + + codec->num_pcms = spec->num_cvts; + codec->pcm_info = info; + + for (i = 0; i < codec->num_pcms; i++, info++) { + unsigned int chans; + struct hda_pcm_stream *pstr; + + chans = get_wcaps(codec, spec->cvt[i]); + chans = get_wcaps_channels(chans); + + info->name = generic_hdmi_pcm_names[i]; + info->pcm_type = HDA_PCM_TYPE_HDMI; + pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; + if (spec->pcm_playback) + *pstr = *spec->pcm_playback; + else + *pstr = generic_hdmi_pcm_playback; + pstr->nid = spec->cvt[i]; + if (pstr->channels_max <= 2 && chans && chans <= 16) + pstr->channels_max = chans; + } + + return 0; +} + +static int generic_hdmi_build_controls(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int err; + int i; + + for (i = 0; i < codec->num_pcms; i++) { + err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]); + if (err < 0) + return err; + } + + return 0; +} + +static int generic_hdmi_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int i; + + for (i = 0; spec->pin[i]; i++) { + hdmi_enable_output(codec, spec->pin[i]); + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | spec->pin[i]); + } + return 0; +} + +static void generic_hdmi_free(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); + + kfree(spec); +} + +static struct hda_codec_ops generic_hdmi_patch_ops = { + .init = generic_hdmi_init, + .free = generic_hdmi_free, + .build_pcms = generic_hdmi_build_pcms, + .build_controls = generic_hdmi_build_controls, + .unsol_event = hdmi_unsol_event, +}; + +static int patch_generic_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int i; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + if (hdmi_parse_codec(codec) < 0) { + codec->spec = NULL; + kfree(spec); + return -EINVAL; + } + codec->patch_ops = generic_hdmi_patch_ops; + + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); + + init_channel_allocations(); + + return 0; +} + +/* + * Nvidia specific implementations + */ + +#define Nv_VERB_SET_Channel_Allocation 0xF79 +#define Nv_VERB_SET_Info_Frame_Checksum 0xF7A +#define Nv_VERB_SET_Audio_Protection_On 0xF98 +#define Nv_VERB_SET_Audio_Protection_Off 0xF99 + +#define nvhdmi_master_con_nid_7x 0x04 +#define nvhdmi_master_pin_nid_7x 0x05 + +static hda_nid_t nvhdmi_con_nids_7x[4] = { + /*front, rear, clfe, rear_surr */ + 0x6, 0x8, 0xa, 0xc, +}; + +static struct hda_verb nvhdmi_basic_init_7x[] = { + /* set audio protect on */ + { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, + /* enable digital output on pin widget */ + { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x7, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0x9, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xb, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + { 0xd, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, + {} /* terminator */ +}; + +#ifdef LIMITED_RATE_FMT_SUPPORT +/* support only the safe format and rate */ +#define SUPPORTED_RATES SNDRV_PCM_RATE_48000 +#define SUPPORTED_MAXBPS 16 +#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#else +/* support all rates and formats */ +#define SUPPORTED_RATES \ + (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define SUPPORTED_MAXBPS 24 +#define SUPPORTED_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#endif + +static int nvhdmi_7x_init(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, nvhdmi_basic_init_7x); + return 0; +} + +static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int simple_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int simple_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + +static int nvhdmi_8ch_7x_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + int i; + + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, + 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + for (i = 0; i < 4; i++) { + /* set the stream id */ + snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, + AC_VERB_SET_CHANNEL_STREAMID, 0); + /* set the stream format */ + snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, + AC_VERB_SET_STREAM_FORMAT, 0); + } + + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int nvhdmi_8ch_7x_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + int chs; + unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; + int i; + + mutex_lock(&codec->spdif_mutex); + + chs = substream->runtime->channels; + chan = chs ? (chs - 1) : 1; + + switch (chs) { + default: + case 0: + case 2: + chanmask = 0x00; + break; + case 4: + chanmask = 0x08; + break; + case 6: + chanmask = 0x0b; + break; + case 8: + chanmask = 0x13; + break; + } + dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; + dataDCC2 = 0x2; + + /* set the Audio InforFrame Channel Allocation */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Channel_Allocation, chanmask); + + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + nvhdmi_master_con_nid_7x, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + + /* set the stream id */ + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, + AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0); + + /* set the stream format */ + snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, + AC_VERB_SET_STREAM_FORMAT, format); + + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + nvhdmi_master_con_nid_7x, + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + nvhdmi_master_con_nid_7x, + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + + for (i = 0; i < 4; i++) { + if (chs == 2) + channel_id = 0; + else + channel_id = i * 2; + + /* turn off SPDIF once; + *otherwise the IEC958 bits won't be updated + */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) + snd_hda_codec_write(codec, + nvhdmi_con_nids_7x[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + /* set the stream id */ + snd_hda_codec_write(codec, + nvhdmi_con_nids_7x[i], + 0, + AC_VERB_SET_CHANNEL_STREAMID, + (stream_tag << 4) | channel_id); + /* set the stream format */ + snd_hda_codec_write(codec, + nvhdmi_con_nids_7x[i], + 0, + AC_VERB_SET_STREAM_FORMAT, + format); + /* turn on again (if needed) */ + /* enable and set the channel status audio/data flag */ + if (codec->spdif_status_reset && + (codec->spdif_ctls & AC_DIG1_ENABLE)) { + snd_hda_codec_write(codec, + nvhdmi_con_nids_7x[i], + 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + snd_hda_codec_write(codec, + nvhdmi_con_nids_7x[i], + 0, + AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); + } + } + + /* set the Audio Info Frame Checksum */ + snd_hda_codec_write(codec, 0x1, 0, + Nv_VERB_SET_Info_Frame_Checksum, + (0x71 - chan - chanmask)); + + mutex_unlock(&codec->spdif_mutex); + return 0; +} + +static struct hda_pcm_stream nvhdmi_pcm_playback_8ch_7x = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .nid = nvhdmi_master_con_nid_7x, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, + .ops = { + .open = simple_playback_pcm_open, + .close = nvhdmi_8ch_7x_pcm_close, + .prepare = nvhdmi_8ch_7x_pcm_prepare + }, +}; + +static struct hda_pcm_stream nvhdmi_pcm_playback_2ch = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = nvhdmi_master_con_nid_7x, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, + .ops = { + .open = simple_playback_pcm_open, + .close = simple_playback_pcm_close, + .prepare = simple_playback_pcm_prepare + }, +}; + +static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { + .build_controls = generic_hdmi_build_controls, + .build_pcms = generic_hdmi_build_pcms, + .init = nvhdmi_7x_init, + .free = generic_hdmi_free, +}; + +static struct hda_codec_ops nvhdmi_patch_ops_2ch = { + .build_controls = generic_hdmi_build_controls, + .build_pcms = generic_hdmi_build_pcms, + .init = nvhdmi_7x_init, + .free = generic_hdmi_free, +}; + +static int patch_nvhdmi_8ch_89(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err = patch_generic_hdmi(codec); + + if (err < 0) + return err; + spec = codec->spec; + spec->old_pin_detect = 1; + return 0; +} + +static int patch_nvhdmi_2ch(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; + spec->old_pin_detect = 1; + spec->num_cvts = 1; + spec->cvt[0] = nvhdmi_master_con_nid_7x; + spec->pcm_playback = &nvhdmi_pcm_playback_2ch; + + codec->patch_ops = nvhdmi_patch_ops_2ch; + + return 0; +} + +static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err = patch_nvhdmi_2ch(codec); + + if (err < 0) + return err; + spec = codec->spec; + spec->multiout.max_channels = 8; + spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x; + codec->patch_ops = nvhdmi_patch_ops_8ch_7x; + return 0; +} + +/* + * ATI-specific implementations + * + * FIXME: we may omit the whole this and use the generic code once after + * it's confirmed to work. + */ + +#define ATIHDMI_CVT_NID 0x02 /* audio converter */ +#define ATIHDMI_PIN_NID 0x03 /* HDMI output pin */ + +static int atihdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + int chans = substream->runtime->channels; + int i, err; + + err = simple_playback_pcm_prepare(hinfo, codec, stream_tag, format, + substream); + if (err < 0) + return err; + snd_hda_codec_write(codec, spec->cvt[0], 0, AC_VERB_SET_CVT_CHAN_COUNT, + chans - 1); + /* FIXME: XXX */ + for (i = 0; i < chans; i++) { + snd_hda_codec_write(codec, spec->cvt[0], 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (i << 4) | i); + } + return 0; +} + +static struct hda_pcm_stream atihdmi_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = ATIHDMI_CVT_NID, + .ops = { + .open = simple_playback_pcm_open, + .close = simple_playback_pcm_close, + .prepare = atihdmi_playback_pcm_prepare + }, +}; + +static struct hda_verb atihdmi_basic_init[] = { + /* enable digital output on pin widget */ + { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {} /* terminator */ +}; + +static int atihdmi_init(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, atihdmi_basic_init); + /* SI codec requires to unmute the pin */ + if (get_wcaps(codec, spec->pin[0]) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, spec->pin[0], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + return 0; +} + +static struct hda_codec_ops atihdmi_patch_ops = { + .build_controls = generic_hdmi_build_controls, + .build_pcms = generic_hdmi_build_pcms, + .init = atihdmi_init, + .free = generic_hdmi_free, +}; + + +static int patch_atihdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = ATIHDMI_CVT_NID; + spec->num_cvts = 1; + spec->cvt[0] = ATIHDMI_CVT_NID; + spec->pin[0] = ATIHDMI_PIN_NID; + spec->pcm_playback = &atihdmi_pcm_digital_playback; + + codec->patch_ops = atihdmi_patch_ops; + + return 0; +} + + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_hdmi[] = { +{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi }, +{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi }, +{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi }, +{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi }, +{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, +{ .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, +{ .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, +{ .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, +{ .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, +{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +{} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:1002793c"); +MODULE_ALIAS("snd-hda-codec-id:10027919"); +MODULE_ALIAS("snd-hda-codec-id:1002791a"); +MODULE_ALIAS("snd-hda-codec-id:1002aa01"); +MODULE_ALIAS("snd-hda-codec-id:10951390"); +MODULE_ALIAS("snd-hda-codec-id:10951392"); +MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0003"); +MODULE_ALIAS("snd-hda-codec-id:10de0005"); +MODULE_ALIAS("snd-hda-codec-id:10de0006"); +MODULE_ALIAS("snd-hda-codec-id:10de0007"); +MODULE_ALIAS("snd-hda-codec-id:10de000a"); +MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000c"); +MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0010"); +MODULE_ALIAS("snd-hda-codec-id:10de0011"); +MODULE_ALIAS("snd-hda-codec-id:10de0012"); +MODULE_ALIAS("snd-hda-codec-id:10de0013"); +MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0018"); +MODULE_ALIAS("snd-hda-codec-id:10de0019"); +MODULE_ALIAS("snd-hda-codec-id:10de001a"); +MODULE_ALIAS("snd-hda-codec-id:10de001b"); +MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0040"); +MODULE_ALIAS("snd-hda-codec-id:10de0041"); +MODULE_ALIAS("snd-hda-codec-id:10de0042"); +MODULE_ALIAS("snd-hda-codec-id:10de0043"); +MODULE_ALIAS("snd-hda-codec-id:10de0044"); +MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); +MODULE_ALIAS("snd-hda-codec-id:17e80047"); +MODULE_ALIAS("snd-hda-codec-id:80860054"); +MODULE_ALIAS("snd-hda-codec-id:80862801"); +MODULE_ALIAS("snd-hda-codec-id:80862802"); +MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:80862804"); +MODULE_ALIAS("snd-hda-codec-id:80862805"); +MODULE_ALIAS("snd-hda-codec-id:808629fb"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("HDMI HD-audio codec"); +MODULE_ALIAS("snd-hda-codec-intelhdmi"); +MODULE_ALIAS("snd-hda-codec-nvhdmi"); +MODULE_ALIAS("snd-hda-codec-atihdmi"); + +static struct hda_codec_preset_list intel_list = { + .preset = snd_hda_preset_hdmi, + .owner = THIS_MODULE, +}; + +static int __init patch_hdmi_init(void) +{ + return snd_hda_add_codec_preset(&intel_list); +} + +static void __exit patch_hdmi_exit(void) +{ + snd_hda_delete_codec_preset(&intel_list); +} + +module_init(patch_hdmi_init) +module_exit(patch_hdmi_exit) diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c deleted file mode 100644 index 36a9b83a6174..000000000000 --- a/sound/pci/hda/patch_intelhdmi.c +++ /dev/null @@ -1,220 +0,0 @@ -/* - * - * patch_intelhdmi.c - Patch for Intel HDMI codecs - * - * Copyright(c) 2008 Intel Corporation. All rights reserved. - * - * Authors: - * Jiang Zhe - * Wu Fengguang - * - * Maintained by: - * Wu Fengguang - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY - * or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License - * for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software Foundation, - * Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. - */ - -#include -#include -#include -#include -#include "hda_codec.h" -#include "hda_local.h" - -/* - * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device - * could support two independent pipes, each of them can be connected to one or - * more ports (DVI, HDMI or DisplayPort). - * - * The HDA correspondence of pipes/ports are converter/pin nodes. - */ -#define MAX_HDMI_CVTS 3 -#define MAX_HDMI_PINS 3 - -#include "patch_hdmi.c" - -static char *intel_hdmi_pcm_names[MAX_HDMI_CVTS] = { - "INTEL HDMI 0", - "INTEL HDMI 1", - "INTEL HDMI 2", -}; - -/* - * HDMI callbacks - */ - -static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - hdmi_set_channel_count(codec, hinfo->nid, - substream->runtime->channels); - - hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - - return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); -} - -static struct hda_pcm_stream intel_hdmi_pcm_playback = { - .substreams = 1, - .channels_min = 2, - .ops = { - .open = hdmi_pcm_open, - .prepare = intel_hdmi_playback_pcm_prepare, - }, -}; - -static int intel_hdmi_build_pcms(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - int i; - - codec->num_pcms = spec->num_cvts; - codec->pcm_info = info; - - for (i = 0; i < codec->num_pcms; i++, info++) { - unsigned int chans; - - chans = get_wcaps(codec, spec->cvt[i]); - chans = get_wcaps_channels(chans); - - info->name = intel_hdmi_pcm_names[i]; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - intel_hdmi_pcm_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; - } - - return 0; -} - -static int intel_hdmi_build_controls(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int err; - int i; - - for (i = 0; i < codec->num_pcms; i++) { - err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]); - if (err < 0) - return err; - } - - return 0; -} - -static int intel_hdmi_init(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int i; - - for (i = 0; spec->pin[i]; i++) { - hdmi_enable_output(codec, spec->pin[i]); - snd_hda_codec_write(codec, spec->pin[i], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | spec->pin[i]); - } - return 0; -} - -static void intel_hdmi_free(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) - snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); - - kfree(spec); -} - -static struct hda_codec_ops intel_hdmi_patch_ops = { - .init = intel_hdmi_init, - .free = intel_hdmi_free, - .build_pcms = intel_hdmi_build_pcms, - .build_controls = intel_hdmi_build_controls, - .unsol_event = hdmi_unsol_event, -}; - -static int patch_intel_hdmi(struct hda_codec *codec) -{ - struct hdmi_spec *spec; - int i; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - if (hdmi_parse_codec(codec) < 0) { - codec->spec = NULL; - kfree(spec); - return -EINVAL; - } - codec->patch_ops = intel_hdmi_patch_ops; - - for (i = 0; i < spec->num_pins; i++) - snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); - - init_channel_allocations(); - - return 0; -} - -static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { -{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_intel_hdmi }, -{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, -{} /* terminator */ -}; - -MODULE_ALIAS("snd-hda-codec-id:808629fb"); -MODULE_ALIAS("snd-hda-codec-id:80862801"); -MODULE_ALIAS("snd-hda-codec-id:80862802"); -MODULE_ALIAS("snd-hda-codec-id:80862803"); -MODULE_ALIAS("snd-hda-codec-id:80862804"); -MODULE_ALIAS("snd-hda-codec-id:80862805"); -MODULE_ALIAS("snd-hda-codec-id:80860054"); -MODULE_ALIAS("snd-hda-codec-id:10951392"); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Intel HDMI HD-audio codec"); - -static struct hda_codec_preset_list intel_list = { - .preset = snd_hda_preset_intelhdmi, - .owner = THIS_MODULE, -}; - -static int __init patch_intelhdmi_init(void) -{ - return snd_hda_add_codec_preset(&intel_list); -} - -static void __exit patch_intelhdmi_exit(void) -{ - snd_hda_delete_codec_preset(&intel_list); -} - -module_init(patch_intelhdmi_init) -module_exit(patch_intelhdmi_exit) diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c deleted file mode 100644 index baa108b9d6aa..000000000000 --- a/sound/pci/hda/patch_nvhdmi.c +++ /dev/null @@ -1,608 +0,0 @@ -/* - * Universal Interface for Intel High Definition Audio Codec - * - * HD audio interface patch for NVIDIA HDMI codecs - * - * Copyright (c) 2008 NVIDIA Corp. All rights reserved. - * Copyright (c) 2008 Wei Ni - * - * - * This driver is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This driver is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include -#include -#include -#include -#include "hda_codec.h" -#include "hda_local.h" - -#define MAX_HDMI_CVTS 1 -#define MAX_HDMI_PINS 1 - -#include "patch_hdmi.c" - -static char *nvhdmi_pcm_names[MAX_HDMI_CVTS] = { - "NVIDIA HDMI", -}; - -/* define below to restrict the supported rates and formats */ -/* #define LIMITED_RATE_FMT_SUPPORT */ - -enum HDACodec { - HDA_CODEC_NVIDIA_MCP7X, - HDA_CODEC_NVIDIA_MCP89, - HDA_CODEC_NVIDIA_GT21X, - HDA_CODEC_INVALID -}; - -#define Nv_VERB_SET_Channel_Allocation 0xF79 -#define Nv_VERB_SET_Info_Frame_Checksum 0xF7A -#define Nv_VERB_SET_Audio_Protection_On 0xF98 -#define Nv_VERB_SET_Audio_Protection_Off 0xF99 - -#define nvhdmi_master_con_nid_7x 0x04 -#define nvhdmi_master_pin_nid_7x 0x05 - -#define nvhdmi_master_con_nid_89 0x04 -#define nvhdmi_master_pin_nid_89 0x05 - -static hda_nid_t nvhdmi_con_nids_7x[4] = { - /*front, rear, clfe, rear_surr */ - 0x6, 0x8, 0xa, 0xc, -}; - -static struct hda_verb nvhdmi_basic_init_7x[] = { - /* set audio protect on */ - { 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1}, - /* enable digital output on pin widget */ - { 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, - { 0x7, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, - { 0x9, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, - { 0xb, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, - { 0xd, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 }, - {} /* terminator */ -}; - -#ifdef LIMITED_RATE_FMT_SUPPORT -/* support only the safe format and rate */ -#define SUPPORTED_RATES SNDRV_PCM_RATE_48000 -#define SUPPORTED_MAXBPS 16 -#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#else -/* support all rates and formats */ -#define SUPPORTED_RATES \ - (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ - SNDRV_PCM_RATE_192000) -#define SUPPORTED_MAXBPS 24 -#define SUPPORTED_FORMATS \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) -#endif - -/* - * Controls - */ -static int nvhdmi_build_controls(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int err; - int i; - - if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) - || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { - for (i = 0; i < codec->num_pcms; i++) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->cvt[i]); - if (err < 0) - return err; - } - } else { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - } - - return 0; -} - -static int nvhdmi_init(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int i; - if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) - || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { - for (i = 0; spec->pin[i]; i++) { - hdmi_enable_output(codec, spec->pin[i]); - snd_hda_codec_write(codec, spec->pin[i], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | spec->pin[i]); - } - } else { - snd_hda_sequence_write(codec, nvhdmi_basic_init_7x); - } - return 0; -} - -static void nvhdmi_free(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - int i; - - if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) - || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { - for (i = 0; i < spec->num_pins; i++) - snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); - } - - kfree(spec); -} - -/* - * Digital out - */ -static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int nvhdmi_dig_playback_pcm_close_8ch_7x(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hdmi_spec *spec = codec->spec; - int i; - - snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, - 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - for (i = 0; i < 4; i++) { - /* set the stream id */ - snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, - AC_VERB_SET_CHANNEL_STREAMID, 0); - /* set the stream format */ - snd_hda_codec_write(codec, nvhdmi_con_nids_7x[i], 0, - AC_VERB_SET_STREAM_FORMAT, 0); - } - - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int nvhdmi_dig_playback_pcm_close_2ch(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int nvhdmi_dig_playback_pcm_prepare_8ch_89(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - hdmi_set_channel_count(codec, hinfo->nid, - substream->runtime->channels); - - hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - - return hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); -} - -static int nvhdmi_dig_playback_pcm_prepare_8ch(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - int chs; - unsigned int dataDCC1, dataDCC2, chan, chanmask, channel_id; - int i; - - mutex_lock(&codec->spdif_mutex); - - chs = substream->runtime->channels; - chan = chs ? (chs - 1) : 1; - - switch (chs) { - default: - case 0: - case 2: - chanmask = 0x00; - break; - case 4: - chanmask = 0x08; - break; - case 6: - chanmask = 0x0b; - break; - case 8: - chanmask = 0x13; - break; - } - dataDCC1 = AC_DIG1_ENABLE | AC_DIG1_COPYRIGHT; - dataDCC2 = 0x2; - - /* set the Audio InforFrame Channel Allocation */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Channel_Allocation, chanmask); - - /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) - snd_hda_codec_write(codec, - nvhdmi_master_con_nid_7x, - 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - - /* set the stream id */ - snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, - AC_VERB_SET_CHANNEL_STREAMID, (stream_tag << 4) | 0x0); - - /* set the stream format */ - snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, 0, - AC_VERB_SET_STREAM_FORMAT, format); - - /* turn on again (if needed) */ - /* enable and set the channel status audio/data flag */ - if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) { - snd_hda_codec_write(codec, - nvhdmi_master_con_nid_7x, - 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); - snd_hda_codec_write(codec, - nvhdmi_master_con_nid_7x, - 0, - AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); - } - - for (i = 0; i < 4; i++) { - if (chs == 2) - channel_id = 0; - else - channel_id = i * 2; - - /* turn off SPDIF once; - *otherwise the IEC958 bits won't be updated - */ - if (codec->spdif_status_reset && - (codec->spdif_ctls & AC_DIG1_ENABLE)) - snd_hda_codec_write(codec, - nvhdmi_con_nids_7x[i], - 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - /* set the stream id */ - snd_hda_codec_write(codec, - nvhdmi_con_nids_7x[i], - 0, - AC_VERB_SET_CHANNEL_STREAMID, - (stream_tag << 4) | channel_id); - /* set the stream format */ - snd_hda_codec_write(codec, - nvhdmi_con_nids_7x[i], - 0, - AC_VERB_SET_STREAM_FORMAT, - format); - /* turn on again (if needed) */ - /* enable and set the channel status audio/data flag */ - if (codec->spdif_status_reset && - (codec->spdif_ctls & AC_DIG1_ENABLE)) { - snd_hda_codec_write(codec, - nvhdmi_con_nids_7x[i], - 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); - snd_hda_codec_write(codec, - nvhdmi_con_nids_7x[i], - 0, - AC_VERB_SET_DIGI_CONVERT_2, dataDCC2); - } - } - - /* set the Audio Info Frame Checksum */ - snd_hda_codec_write(codec, 0x1, 0, - Nv_VERB_SET_Info_Frame_Checksum, - (0x71 - chan - chanmask)); - - mutex_unlock(&codec->spdif_mutex); - return 0; -} - -static int nvhdmi_dig_playback_pcm_prepare_2ch(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct hdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_89 = { - .substreams = 1, - .channels_min = 2, - .ops = { - .open = hdmi_pcm_open, - .prepare = nvhdmi_dig_playback_pcm_prepare_8ch_89, - }, -}; - -static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch_7x = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, - .nid = nvhdmi_master_con_nid_7x, - .rates = SUPPORTED_RATES, - .maxbps = SUPPORTED_MAXBPS, - .formats = SUPPORTED_FORMATS, - .ops = { - .open = nvhdmi_dig_playback_pcm_open, - .close = nvhdmi_dig_playback_pcm_close_8ch_7x, - .prepare = nvhdmi_dig_playback_pcm_prepare_8ch - }, -}; - -static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = nvhdmi_master_con_nid_7x, - .rates = SUPPORTED_RATES, - .maxbps = SUPPORTED_MAXBPS, - .formats = SUPPORTED_FORMATS, - .ops = { - .open = nvhdmi_dig_playback_pcm_open, - .close = nvhdmi_dig_playback_pcm_close_2ch, - .prepare = nvhdmi_dig_playback_pcm_prepare_2ch - }, -}; - -static int nvhdmi_build_pcms_8ch_89(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - int i; - - codec->num_pcms = spec->num_cvts; - codec->pcm_info = info; - - for (i = 0; i < codec->num_pcms; i++, info++) { - unsigned int chans; - - chans = get_wcaps(codec, spec->cvt[i]); - chans = get_wcaps_channels(chans); - - info->name = nvhdmi_pcm_names[i]; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] - = nvhdmi_pcm_digital_playback_8ch_89; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; - } - - return 0; -} - -static int nvhdmi_build_pcms_8ch_7x(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "NVIDIA HDMI"; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] - = nvhdmi_pcm_digital_playback_8ch_7x; - - return 0; -} - -static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) -{ - struct hdmi_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "NVIDIA HDMI"; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] - = nvhdmi_pcm_digital_playback_2ch; - - return 0; -} - -static struct hda_codec_ops nvhdmi_patch_ops_8ch_89 = { - .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms_8ch_89, - .init = nvhdmi_init, - .free = nvhdmi_free, - .unsol_event = hdmi_unsol_event, -}; - -static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { - .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms_8ch_7x, - .init = nvhdmi_init, - .free = nvhdmi_free, -}; - -static struct hda_codec_ops nvhdmi_patch_ops_2ch = { - .build_controls = nvhdmi_build_controls, - .build_pcms = nvhdmi_build_pcms_2ch, - .init = nvhdmi_init, - .free = nvhdmi_free, -}; - -static int patch_nvhdmi_8ch_89(struct hda_codec *codec) -{ - struct hdmi_spec *spec; - int i; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - spec->codec_type = HDA_CODEC_NVIDIA_MCP89; - spec->old_pin_detect = 1; - - if (hdmi_parse_codec(codec) < 0) { - codec->spec = NULL; - kfree(spec); - return -EINVAL; - } - codec->patch_ops = nvhdmi_patch_ops_8ch_89; - - for (i = 0; i < spec->num_pins; i++) - snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); - - init_channel_allocations(); - - return 0; -} - -static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) -{ - struct hdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; - spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; - spec->old_pin_detect = 1; - - codec->patch_ops = nvhdmi_patch_ops_8ch_7x; - - return 0; -} - -static int patch_nvhdmi_2ch(struct hda_codec *codec) -{ - struct hdmi_spec *spec; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x; - spec->codec_type = HDA_CODEC_NVIDIA_MCP7X; - spec->old_pin_detect = 1; - - codec->patch_ops = nvhdmi_patch_ops_2ch; - - return 0; -} - -/* - * patch entries - */ -static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, - {} /* terminator */ -}; - -MODULE_ALIAS("snd-hda-codec-id:10de0002"); -MODULE_ALIAS("snd-hda-codec-id:10de0003"); -MODULE_ALIAS("snd-hda-codec-id:10de0005"); -MODULE_ALIAS("snd-hda-codec-id:10de0006"); -MODULE_ALIAS("snd-hda-codec-id:10de0007"); -MODULE_ALIAS("snd-hda-codec-id:10de000a"); -MODULE_ALIAS("snd-hda-codec-id:10de000b"); -MODULE_ALIAS("snd-hda-codec-id:10de000c"); -MODULE_ALIAS("snd-hda-codec-id:10de000d"); -MODULE_ALIAS("snd-hda-codec-id:10de0010"); -MODULE_ALIAS("snd-hda-codec-id:10de0011"); -MODULE_ALIAS("snd-hda-codec-id:10de0012"); -MODULE_ALIAS("snd-hda-codec-id:10de0013"); -MODULE_ALIAS("snd-hda-codec-id:10de0014"); -MODULE_ALIAS("snd-hda-codec-id:10de0018"); -MODULE_ALIAS("snd-hda-codec-id:10de0019"); -MODULE_ALIAS("snd-hda-codec-id:10de001a"); -MODULE_ALIAS("snd-hda-codec-id:10de001b"); -MODULE_ALIAS("snd-hda-codec-id:10de001c"); -MODULE_ALIAS("snd-hda-codec-id:10de0040"); -MODULE_ALIAS("snd-hda-codec-id:10de0041"); -MODULE_ALIAS("snd-hda-codec-id:10de0042"); -MODULE_ALIAS("snd-hda-codec-id:10de0043"); -MODULE_ALIAS("snd-hda-codec-id:10de0044"); -MODULE_ALIAS("snd-hda-codec-id:10de0067"); -MODULE_ALIAS("snd-hda-codec-id:10de8001"); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec"); - -static struct hda_codec_preset_list nvhdmi_list = { - .preset = snd_hda_preset_nvhdmi, - .owner = THIS_MODULE, -}; - -static int __init patch_nvhdmi_init(void) -{ - return snd_hda_add_codec_preset(&nvhdmi_list); -} - -static void __exit patch_nvhdmi_exit(void) -{ - snd_hda_delete_codec_preset(&nvhdmi_list); -} - -module_init(patch_nvhdmi_init) -module_exit(patch_nvhdmi_exit) -- cgit v1.2.3 From f6837bbd599c2a4e1f621441f84286434bcc91ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Sep 2010 14:56:32 +0200 Subject: ALSA: hda - Fix up autocfg output pin numbers in realtek parser When quirks are applied, the numbers of output pins in autocfg aren't set up properly but only pin arrays are changed. Let's fix it up so that the rest of the parser can use autocfg.line_outs & co safely. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6045f281b225..bb3cf3b7282b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -876,6 +876,28 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); } +static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + + if (!cfg->line_outs) { + while (cfg->line_outs < AUTO_CFG_MAX_OUTS && + cfg->line_out_pins[cfg->line_outs]) + cfg->line_outs++; + } + if (!cfg->speaker_outs) { + while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS && + cfg->speaker_pins[cfg->speaker_outs]) + cfg->speaker_outs++; + } + if (!cfg->hp_outs) { + while (cfg->hp_outs < AUTO_CFG_MAX_OUTS && + cfg->hp_pins[cfg->hp_outs]) + cfg->hp_outs++; + } +} + /* */ static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) @@ -944,6 +966,8 @@ static void setup_preset(struct hda_codec *codec, if (preset->setup) preset->setup(codec); + + alc_fixup_autocfg_pin_nums(codec); } /* Enable GPIO mask and set output */ -- cgit v1.2.3 From e1ca7b4ea29707920650d86b22afdb7e94ad5986 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Sep 2010 14:58:57 +0200 Subject: ALSA: hda - Fix initialization of multiple output pins for ALC268/269 When multiple pins are assigned to headphones or speakers, they haven't been initialized properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bb3cf3b7282b..c4d9ad70fde7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13484,8 +13484,10 @@ static void alc268_auto_set_output_and_unmute(struct hda_codec *codec, static void alc268_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.line_out_pins[0]; - if (nid) { + int i; + + for (i = 0; i < spec->autocfg.line_outs; i++) { + hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); alc268_auto_set_output_and_unmute(codec, nid, pin_type); } @@ -13495,13 +13497,19 @@ static void alc268_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; alc268_auto_set_output_and_unmute(codec, pin, PIN_HP); - pin = spec->autocfg.speaker_pins[0]; - if (pin) + } + for (i = 0; i < spec->autocfg.speaker_outs; i++) { + pin = spec->autocfg.speaker_pins[i]; alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT); + } + if (spec->autocfg.mono_out_pin) + snd_hda_codec_write(codec, spec->autocfg.mono_out_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); } static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) -- cgit v1.2.3 From d433a67831ab2c470cc53a3ff9b60f656767be15 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Sep 2010 15:11:54 +0200 Subject: ALSA: hda - Optimize the check of ALC269 codec variants Don't call the COEF check for checking ACL269 codec variants at each time in init but remember the type at the initialization. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4d9ad70fde7..9bcf34eab679 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -393,6 +393,7 @@ struct alc_spec { unsigned int no_analog :1; /* digital I/O only */ unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ int init_amp; + int codec_variant; /* flag for other variants */ /* for virtual master */ hda_nid_t vmaster_nid; @@ -14568,6 +14569,13 @@ static int alc275_setup_dual_adc(struct hda_codec *codec) return 0; } +/* different alc269-variants */ +enum { + ALC269_TYPE_NORMAL, + ALC269_TYPE_ALC259, + ALC269_TYPE_ALC271X, +}; + /* * BIOS auto configuration */ @@ -14596,7 +14604,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { + if (spec->codec_variant != ALC269_TYPE_NORMAL) { add_verb(spec, alc269vb_init_verbs); alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { @@ -14962,7 +14970,6 @@ static int patch_alc269(struct hda_codec *codec) struct alc_spec *spec; int board_config; int err; - int is_alc269vb = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -14974,11 +14981,13 @@ static int patch_alc269(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){ if (codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) + spec->cdefine.platform_type == 1) { alc_codec_rename(codec, "ALC271X"); - else + spec->codec_variant = ALC269_TYPE_ALC271X; + } else { alc_codec_rename(codec, "ALC259"); - is_alc269vb = 1; + spec->codec_variant = ALC269_TYPE_ALC259; + } } else alc_fix_pll_init(codec, 0x20, 0x04, 15); @@ -15040,7 +15049,7 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_capture = &alc269_pcm_digital_capture; if (!spec->adc_nids) { /* wasn't filled automatically? use default */ - if (!is_alc269vb) { + if (spec->codec_variant != ALC269_TYPE_NORMAL) { spec->adc_nids = alc269_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); spec->capsrc_nids = alc269_capsrc_nids; -- cgit v1.2.3 From f3550d1b052a8acf4159b407dbdd1def47f223f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Sep 2010 15:09:03 +0200 Subject: ALSA: hda - Fix capture widget for ALC269vb and co ALC269vb and other variants don't use the widgets 0x24 but prefer the widget 0x22 instead. We need to fix the input parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bcf34eab679..4abe3da1240c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14593,7 +14593,11 @@ static int alc269_parse_auto_config(struct hda_codec *codec) err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; - err = alc269_auto_create_input_ctls(codec, &spec->autocfg); + if (spec->codec_variant == ALC269_TYPE_NORMAL) + err = alc269_auto_create_input_ctls(codec, &spec->autocfg); + else + err = alc_auto_create_input_ctls(codec, &spec->autocfg, 0, + 0x22, 0); if (err < 0) return err; -- cgit v1.2.3 From 0ec33d1f952934ea3251cefc6d108b47818eedd0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Sep 2010 15:20:52 +0200 Subject: ALSA: hda - Refactor ALC269 power-ups/downs in PM callbacks Create a helper function to simplify the code. Also, cleaned up the ifdef SND_HDA_NEEDS_RESUME and CONFIG_SND_HDA_POWER_SAVE. The former is always defined when the latter is set. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 51 ++++++++++++++++++------------------------- 1 file changed, 21 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4abe3da1240c..8689216fdcce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4128,10 +4128,8 @@ static int alc_resume(struct hda_codec *codec) codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE if (codec->patch_ops.check_power_status) codec->patch_ops.check_power_status(codec, 0x01); -#endif return 0; } #endif @@ -14656,22 +14654,26 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +#ifdef SND_HDA_NEEDS_RESUME +static void alc269_toggle_power_output(struct hda_codec *codec, int power_up) +{ + int val = alc_read_coef_idx(codec, 0x04); + if (power_up) + val |= 1 << 11; + else + val &= ~(1 << 11); + alc_write_coef_idx(codec, 0x04, val); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE static int alc269_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; - int val; - - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { - val = alc_read_coef_idx(codec, 0x04); - /* Power down output pin */ - alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); - } + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) + alc269_toggle_power_output(codec, 0); if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { - val = alc_read_coef_idx(codec, 0x04); - /* Power down output pin */ - alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); + alc269_toggle_power_output(codec, 0); msleep(150); } @@ -14680,43 +14682,32 @@ static int alc269_suspend(struct hda_codec *codec, pm_message_t state) spec->power_hook(codec); return 0; } -#endif -#ifdef SND_HDA_NEEDS_RESUME +#endif /* CONFIG_SND_HDA_POWER_SAVE */ + static int alc269_resume(struct hda_codec *codec) { - int val; - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { - val = alc_read_coef_idx(codec, 0x04); - /* Power down output pin */ - alc_write_coef_idx(codec, 0x04, val & ~(1<<11)); + alc269_toggle_power_output(codec, 0); msleep(150); } codec->patch_ops.init(codec); if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x017) { - val = alc_read_coef_idx(codec, 0x04); - /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); + alc269_toggle_power_output(codec, 1); msleep(200); } - if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { - val = alc_read_coef_idx(codec, 0x04); - /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); - } + if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) + alc269_toggle_power_output(codec, 1); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE if (codec->patch_ops.check_power_status) codec->patch_ops.check_power_status(codec, 0x01); -#endif return 0; } -#endif +#endif /* SND_HDA_NEEDS_RESUME */ enum { ALC269_FIXUP_SONY_VAIO, -- cgit v1.2.3 From 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Tue, 21 Sep 2010 14:25:49 +0800 Subject: ALSA: hdmi - support infoframe for DisplayPort DisplayPort works mostly in the same way as HDMI, except that it expects a slightly different audio infoframe format. Citations from "HDA036-A: Display Port Support and HDMI Miscellaneous Corrections": The HDMI specification defines a data island packet with a header of 4 bytes (3 bytes content + 1 byte ECC) and packet body of 32 bytes (28 bytes content and 4 bytes ECC). Display Port specification on the other hand defines a data island packet (secondary data packet) with header of 4 bytes protected by 4 bytes of parity, and data of theoretically up to 1024 bytes with each 16 bytes chunk of data protected by 4 bytes of parity. Note that the ECC or parity bytes are not present in the DIP content populated by software and are hardware generated. It tests DP connection based on the ELD conn_type field, which will be set by the graphics driver and can be overriden manually by users through the /proc/asound/card0/eld* interface. The DP infoframe is tested OK on Intel SandyBridge/CougarPoint platform. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 110 ++++++++++++++++++++++++++++++--------------- 1 file changed, 73 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index cb997ca0fdfa..1f4ae1aeca44 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -84,13 +84,25 @@ struct hdmi_audio_infoframe { u8 ver; /* 0x01 */ u8 len; /* 0x0a */ - u8 checksum; /* PB0 */ + u8 checksum; + u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ u8 SS01_SF24; u8 CXT04; u8 CA; u8 LFEPBL01_LSV36_DM_INH7; - u8 reserved[5]; /* PB6 - PB10 */ +}; + +struct dp_audio_infoframe { + u8 type; /* 0x84 */ + u8 len; /* 0x1b */ + u8 ver; /* 0x11 << 2 */ + + u8 CC02_CT47; /* match with HDMI infoframe from this on */ + u8 SS01_SF24; + u8 CXT04; + u8 CA; + u8 LFEPBL01_LSV36_DM_INH7; }; /* @@ -194,7 +206,7 @@ static int hdmi_channel_mapping[0x32][8] = { * This is an ordered list! * * The preceding ones have better chances to be selected by - * hdmi_setup_channel_allocation(). + * hdmi_channel_allocation(). */ static struct cea_channel_speaker_allocation channel_allocations[] = { /* channel: 7 6 5 4 3 2 1 0 */ @@ -371,14 +383,14 @@ static void init_channel_allocations(void) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, - struct hdmi_audio_infoframe *ai) +static int hdmi_channel_allocation(struct hda_codec *codec, hda_nid_t nid, + int channels) { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld; int i; + int ca = 0; int spk_mask = 0; - int channels = 1 + (ai->CC02_CT47 & 0x7); char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; /* @@ -416,16 +428,16 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, if (channels == channel_allocations[i].channels && (spk_mask & channel_allocations[i].spk_mask) == channel_allocations[i].spk_mask) { - ai->CA = channel_allocations[i].ca_index; + ca = channel_allocations[i].ca_index; break; } } snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); snd_printdd("HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); + ca, channels, buf); - return ai->CA; + return ca; } static void hdmi_debug_channel_mapping(struct hda_codec *codec, @@ -447,10 +459,9 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) + int ca) { int i; - int ca = ai->CA; int err; if (hdmi_channel_mapping[ca][1] == 0) { @@ -547,41 +558,37 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) #endif } -static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *hdmi_ai) { - u8 *bytes = (u8 *)ai; + u8 *bytes = (u8 *)hdmi_ai; u8 sum = 0; int i; - ai->checksum = 0; + hdmi_ai->checksum = 0; - for (i = 0; i < sizeof(*ai); i++) + for (i = 0; i < sizeof(*hdmi_ai); i++) sum += bytes[i]; - ai->checksum = -sum; + hdmi_ai->checksum = -sum; } static void hdmi_fill_audio_infoframe(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) + u8 *dip, int size) { - u8 *bytes = (u8 *)ai; int i; hdmi_debug_dip_size(codec, pin_nid); hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - hdmi_checksum_audio_infoframe(ai); - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, bytes[i]); + for (i = 0; i < size; i++) + hdmi_write_dip_byte(codec, pin_nid, dip[i]); } static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) + u8 *dip, int size) { - u8 *bytes = (u8 *)ai; u8 val; int i; @@ -590,10 +597,10 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, return false; hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) { + for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); - if (val != bytes[i]) + if (val != dip[i]) return false; } @@ -605,15 +612,13 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, { struct hdmi_spec *spec = codec->spec; hda_nid_t pin_nid; + int channels = substream->runtime->channels; + int ca; int i; - struct hdmi_audio_infoframe ai = { - .type = 0x84, - .ver = 0x01, - .len = 0x0a, - .CC02_CT47 = substream->runtime->channels - 1, - }; + u8 ai[max(sizeof(struct hdmi_audio_infoframe), + sizeof(struct dp_audio_infoframe))]; - hdmi_setup_channel_allocation(codec, nid, &ai); + ca = hdmi_channel_allocation(codec, nid, channels); for (i = 0; i < spec->num_pins; i++) { if (spec->pin_cvt[i] != nid) @@ -622,14 +627,45 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, continue; pin_nid = spec->pin[i]; - if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + + memset(ai, 0, sizeof(ai)); + if (spec->sink_eld[i].conn_type == 0) { /* HDMI */ + struct hdmi_audio_infoframe *hdmi_ai; + + hdmi_ai = (struct hdmi_audio_infoframe *)ai; + hdmi_ai->type = 0x84; + hdmi_ai->ver = 0x01; + hdmi_ai->len = 0x0a; + hdmi_ai->CC02_CT47 = channels - 1; + hdmi_checksum_audio_infoframe(hdmi_ai); + } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */ + struct dp_audio_infoframe *dp_ai; + + dp_ai = (struct dp_audio_infoframe *)ai; + dp_ai->type = 0x84; + dp_ai->len = 0x1b; + dp_ai->ver = 0x11 << 2; + dp_ai->CC02_CT47 = channels - 1; + } else { + snd_printd("HDMI: unknown connection type at pin %d\n", + pin_nid); + continue; + } + + /* + * sizeof(ai) is used instead of sizeof(*hdmi_ai) or + * sizeof(*dp_ai) to avoid partial match/update problems when + * the user switches between HDMI/DP monitors. + */ + if (!hdmi_infoframe_uptodate(codec, pin_nid, ai, sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "cvt=%d pin=%d channels=%d\n", nid, pin_nid, - substream->runtime->channels); - hdmi_setup_channel_mapping(codec, pin_nid, &ai); + channels); + hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_fill_audio_infoframe(codec, pin_nid, + ai, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } } -- cgit v1.2.3 From 9396d3174b761685d6fefb1103e66b96a2e5db6d Mon Sep 17 00:00:00 2001 From: Jerry Zhou Date: Tue, 21 Sep 2010 14:44:51 +0800 Subject: ALSA: hdmi - fix surround41 channel mapping Channel 2 and channel 3 were all wrongly mapped to HDMI slot 4. This shows up as a bug that one channel is "lost" when playing in surround41 mode. Signed-off-by: Jerry Zhou Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1f4ae1aeca44..d3e49aa5b9ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -193,7 +193,7 @@ static int hdmi_channel_mapping[0x32][8] = { /* 4ch */ [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, /* surround41 */ - [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, + [0x09] = { 0x00, 0x11, 0x24, 0x35, 0x42, 0xf3, 0xf6, 0xf7 }, /* surround50 */ [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, /* surround51 */ -- cgit v1.2.3 From 9e5341b92d1d2dde11691b394721b45b36416bef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Sep 2010 09:57:06 +0200 Subject: ALSA: hda - Introduce hda_call_check_power_status() helper Replace the explicit ifdef check and call of check_power_status ops with a new helper function, hda_call_check_power_status(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +---- sound/pci/hda/hda_codec.h | 12 ++++++++++++ sound/pci/hda/patch_realtek.c | 11 +++-------- sound/pci/hda/patch_sigmatel.c | 12 ++++-------- 4 files changed, 20 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e15a75751f57..053f827d2c2c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2228,10 +2228,7 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, HDA_AMP_MUTE, *valp ? 0 : HDA_AMP_MUTE); -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, nid); -#endif + hda_call_check_power_status(codec, nid); snd_hda_power_down(codec); return change; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 62c702240108..ebf8eb02e3c2 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -989,6 +989,18 @@ int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); #endif +#ifdef CONFIG_SND_HDA_POWER_SAVE +static inline +int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + if (codec->patch_ops.check_power_status) + return codec->patch_ops.check_power_status(codec, nid); + return 0; +} +#else +#define hda_call_check_power_status(codec, nid) 0 +#endif + /* * get widget information */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8689216fdcce..9bedca073e9a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3729,10 +3729,7 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, 0x01); -#endif + hda_call_check_power_status(codec, 0x01); return 0; } @@ -4128,8 +4125,7 @@ static int alc_resume(struct hda_codec *codec) codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); - if (codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, 0x01); + hda_call_check_power_status(codec, 0x01); return 0; } #endif @@ -14703,8 +14699,7 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); - if (codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, 0x01); + hda_call_check_power_status(codec, 0x01); return 0; } #endif /* SND_HDA_NEEDS_RESUME */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6bfbc2fe46ed..a90327b0cc3e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4372,11 +4372,9 @@ static int stac92xx_init(struct hda_codec *codec) stac_issue_unsol_event(codec, nid); } -#ifdef CONFIG_SND_HDA_POWER_SAVE /* sync mute LED */ - if (spec->gpio_led && codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, 0x01); -#endif + if (spec->gpio_led) + hda_call_check_power_status(codec, 0x01); if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -4958,11 +4956,9 @@ static int stac92xx_resume(struct hda_codec *codec) stac_issue_unsol_event(codec, spec->autocfg.line_out_pins[0]); } -#ifdef CONFIG_SND_HDA_POWER_SAVE /* sync mute LED */ - if (spec->gpio_led && codec->patch_ops.check_power_status) - codec->patch_ops.check_power_status(codec, 0x01); -#endif + if (spec->gpio_led) + hda_call_check_power_status(codec, 0x01); return 0; } -- cgit v1.2.3 From 0f9f1ee9d1412d45a22bfd69dfd4d4324b506e9e Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Tue, 21 Sep 2010 17:05:46 +1000 Subject: ALSA: hda - Add Dell Latitude E6400 model quirk BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/643891 Set the Dell Latitude E6400 (1028:0233) SSID to use AD1984_DELL_DESKTOP Cc: stable@kernel.org Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b697fd2a6f8b..10bbbaf6ebc3 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3641,6 +3641,7 @@ static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), + SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP), {} }; -- cgit v1.2.3 From 265a02478db5217eda8063004ded1ef0a461c240 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Sep 2010 11:26:21 +0200 Subject: ALSA: hda - Check invalid NIDs in alc_init_jacks() The headphone and external-mic pin NIDs can be null, and the jack input elements should be skipped in such a case. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9bedca073e9a..f5ccba0fd189 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1087,15 +1087,19 @@ static int alc_init_jacks(struct hda_codec *codec) unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int mic_nid = spec->ext_mic.pin; - err = alc_add_jack(codec, hp_nid, SND_JACK_HEADPHONE); - if (err < 0) - return err; - alc_report_jack(codec, hp_nid); + if (hp_nid) { + err = alc_add_jack(codec, hp_nid, SND_JACK_HEADPHONE); + if (err < 0) + return err; + alc_report_jack(codec, hp_nid); + } - err = alc_add_jack(codec, mic_nid, SND_JACK_MICROPHONE); - if (err < 0) - return err; - alc_report_jack(codec, mic_nid); + if (mic_nid) { + err = alc_add_jack(codec, mic_nid, SND_JACK_MICROPHONE); + if (err < 0) + return err; + alc_report_jack(codec, mic_nid); + } return 0; } -- cgit v1.2.3 From abdd8f510686da0a58e475bc0143d1069e5f53da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Sep 2010 17:38:14 +0200 Subject: ALSA: hda - Apply ALC269 VAIO fix-up to all Sony laptops with ALC269 We've applied a fix-up for ALC269 VAIO only for two models. But all Sony VAIO models with ALC269 codec seem to require the similar fix. Let's apply it with vendor-id mask. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2a30447f26f..eea88b7ddb9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14731,8 +14731,7 @@ static const struct alc_fixup alc269_fixups[] = { }; static struct snd_pci_quirk alc269_fixup_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), - SND_PCI_QUIRK(0x104d, 0x9077, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), {} }; -- cgit v1.2.3 From 0873a5ae747847ee55a63db409dff3476e45bcd9 Mon Sep 17 00:00:00 2001 From: "Erik J. Staab" Date: Wed, 22 Sep 2010 11:07:41 +0200 Subject: ALSA: oxygen: fix analog capture on Claro halo cards On the HT-Omega Claro halo card, the ADC data must be captured from the second I2S input. Using the default first input, which isn't connected to anything, would result in silence. Signed-off-by: Erik J. Staab Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 289cb4dacfc7..6c0a11adb2a8 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -543,6 +543,10 @@ static int __devinit get_oxygen_model(struct oxygen *chip, chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF; break; } if (id->driver_data == MODEL_MERIDIAN || -- cgit v1.2.3 From 095a0f6df246bdc57b57d616c4698e41fbd3bf43 Mon Sep 17 00:00:00 2001 From: John Kacur Date: Wed, 22 Sep 2010 13:47:01 +0200 Subject: SOUND-OSS: Remove sh_dac_audio Remove the SH DAC oss driver since there is an equivalent alsa driver. oss has been deprecated for years. Furthermore this driver has BKL code which we are trying to remove. Rather than attempt to fix this, simply remove the driver. Signed-off-by: John Kacur Acked-by: Paul Mundt Acked-by: Ralf Baechle Signed-off-by: Takashi Iwai --- sound/oss/Kconfig | 8 -- sound/oss/Makefile | 1 - sound/oss/sh_dac_audio.c | 326 ----------------------------------------------- 3 files changed, 335 deletions(-) delete mode 100644 sound/oss/sh_dac_audio.c (limited to 'sound') diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index a513651fa149..76c090218073 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -545,11 +545,3 @@ config SOUND_KAHLUA endif # SOUND_OSS -config SOUND_SH_DAC_AUDIO - tristate "SuperH DAC audio support" - depends on CPU_SH3 && HIGH_RES_TIMERS - -config SOUND_SH_DAC_AUDIO_CHANNEL - int "DAC channel" - default "1" - depends on SOUND_SH_DAC_AUDIO diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 567b8a74178a..96f14dcd0cd1 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -9,7 +9,6 @@ obj-$(CONFIG_SOUND_OSS) += sound.o # Please leave it as is, cause the link order is significant ! -obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c deleted file mode 100644 index 53bba16bf709..000000000000 --- a/sound/oss/sh_dac_audio.c +++ /dev/null @@ -1,326 +0,0 @@ -/* - * sound/oss/sh_dac_audio.c - * - * SH DAC based sound :( - * - * Copyright (C) 2004,2005 Andriy Skulysh - * - * This file is subject to the terms and conditions of the GNU General Public - * License. See the file "COPYING" in the main directory of this archive - * for more details. - */ -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#define MODNAME "sh_dac_audio" - -#define BUFFER_SIZE 48000 - -static DEFINE_MUTEX(sh_dac_audio_mutex); -static int rate; -static int empty; -static char *data_buffer, *buffer_begin, *buffer_end; -static int in_use, device_major; -static struct hrtimer hrtimer; -static ktime_t wakeups_per_second; - -static void dac_audio_start_timer(void) -{ - hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); -} - -static void dac_audio_stop_timer(void) -{ - hrtimer_cancel(&hrtimer); -} - -static void dac_audio_reset(void) -{ - dac_audio_stop_timer(); - buffer_begin = buffer_end = data_buffer; - empty = 1; -} - -static void dac_audio_sync(void) -{ - while (!empty) - schedule(); -} - -static void dac_audio_start(void) -{ - if (mach_is_hp6xx()) { - u16 v = __raw_readw(HD64461_GPADR); - v &= ~HD64461_GPADR_SPEAKER; - __raw_writew(v, HD64461_GPADR); - } - - sh_dac_enable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); -} -static void dac_audio_stop(void) -{ - dac_audio_stop_timer(); - - if (mach_is_hp6xx()) { - u16 v = __raw_readw(HD64461_GPADR); - v |= HD64461_GPADR_SPEAKER; - __raw_writew(v, HD64461_GPADR); - } - - sh_dac_output(0, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); - sh_dac_disable(CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); -} - -static void dac_audio_set_rate(void) -{ - wakeups_per_second = ktime_set(0, 1000000000 / rate); -} - -static int dac_audio_ioctl(struct file *file, - unsigned int cmd, unsigned long arg) -{ - int val; - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *)arg); - - case SNDCTL_DSP_SYNC: - dac_audio_sync(); - return 0; - - case SNDCTL_DSP_RESET: - dac_audio_reset(); - return 0; - - case SNDCTL_DSP_GETFMTS: - return put_user(AFMT_U8, (int *)arg); - - case SNDCTL_DSP_SETFMT: - return put_user(AFMT_U8, (int *)arg); - - case SNDCTL_DSP_NONBLOCK: - spin_lock(&file->f_lock); - file->f_flags |= O_NONBLOCK; - spin_unlock(&file->f_lock); - return 0; - - case SNDCTL_DSP_GETCAPS: - return 0; - - case SOUND_PCM_WRITE_RATE: - val = *(int *)arg; - if (val > 0) { - rate = val; - dac_audio_set_rate(); - } - return put_user(rate, (int *)arg); - - case SNDCTL_DSP_STEREO: - return put_user(0, (int *)arg); - - case SOUND_PCM_WRITE_CHANNELS: - return put_user(1, (int *)arg); - - case SNDCTL_DSP_SETDUPLEX: - return -EINVAL; - - case SNDCTL_DSP_PROFILE: - return -EINVAL; - - case SNDCTL_DSP_GETBLKSIZE: - return put_user(BUFFER_SIZE, (int *)arg); - - case SNDCTL_DSP_SETFRAGMENT: - return 0; - - default: - printk(KERN_ERR "sh_dac_audio: unimplemented ioctl=0x%x\n", - cmd); - return -EINVAL; - } - return -EINVAL; -} - -static long dac_audio_unlocked_ioctl(struct file *file, u_int cmd, u_long arg) -{ - int ret; - - mutex_lock(&sh_dac_audio_mutex); - ret = dac_audio_ioctl(file, cmd, arg); - mutex_unlock(&sh_dac_audio_mutex); - - return ret; -} - -static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count, - loff_t * ppos) -{ - int free; - int nbytes; - - if (!count) { - dac_audio_sync(); - return 0; - } - - free = buffer_begin - buffer_end; - - if (free < 0) - free += BUFFER_SIZE; - if ((free == 0) && (empty)) - free = BUFFER_SIZE; - if (count > free) - count = free; - if (buffer_begin > buffer_end) { - if (copy_from_user((void *)buffer_end, buf, count)) - return -EFAULT; - - buffer_end += count; - } else { - nbytes = data_buffer + BUFFER_SIZE - buffer_end; - if (nbytes > count) { - if (copy_from_user((void *)buffer_end, buf, count)) - return -EFAULT; - buffer_end += count; - } else { - if (copy_from_user((void *)buffer_end, buf, nbytes)) - return -EFAULT; - if (copy_from_user - ((void *)data_buffer, buf + nbytes, count - nbytes)) - return -EFAULT; - buffer_end = data_buffer + count - nbytes; - } - } - - if (empty) { - empty = 0; - dac_audio_start_timer(); - } - - return count; -} - -static ssize_t dac_audio_read(struct file *file, char *buf, size_t count, - loff_t * ppos) -{ - return -EINVAL; -} - -static int dac_audio_open(struct inode *inode, struct file *file) -{ - if (file->f_mode & FMODE_READ) - return -ENODEV; - - mutex_lock(&sh_dac_audio_mutex); - if (in_use) { - mutex_unlock(&sh_dac_audio_mutex); - return -EBUSY; - } - - in_use = 1; - - dac_audio_start(); - mutex_unlock(&sh_dac_audio_mutex); - return 0; -} - -static int dac_audio_release(struct inode *inode, struct file *file) -{ - dac_audio_sync(); - dac_audio_stop(); - in_use = 0; - - return 0; -} - -const struct file_operations dac_audio_fops = { - .read = dac_audio_read, - .write = dac_audio_write, - .unlocked_ioctl = dac_audio_unlocked_ioctl, - .open = dac_audio_open, - .release = dac_audio_release, -}; - -static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) -{ - if (!empty) { - sh_dac_output(*buffer_begin, CONFIG_SOUND_SH_DAC_AUDIO_CHANNEL); - buffer_begin++; - - if (buffer_begin == data_buffer + BUFFER_SIZE) - buffer_begin = data_buffer; - if (buffer_begin == buffer_end) - empty = 1; - } - - if (!empty) - hrtimer_start(&hrtimer, wakeups_per_second, HRTIMER_MODE_REL); - - return HRTIMER_NORESTART; -} - -static int __init dac_audio_init(void) -{ - if ((device_major = register_sound_dsp(&dac_audio_fops, -1)) < 0) { - printk(KERN_ERR "Cannot register dsp device"); - return device_major; - } - - in_use = 0; - - data_buffer = kmalloc(BUFFER_SIZE, GFP_KERNEL); - if (data_buffer == NULL) - return -ENOMEM; - - dac_audio_reset(); - rate = 8000; - dac_audio_set_rate(); - - /* Today: High Resolution Timer driven DAC playback. - * The timer callback gets called once per sample. Ouch. - * - * Future: A much better approach would be to use the - * SH7720 CMT+DMAC+DAC hardware combination like this: - * - Program sample rate using CMT0 or CMT1 - * - Program DMAC to use CMT for timing and output to DAC - * - Play sound using DMAC, let CPU sleep. - * - While at it, rewrite this driver to use ALSA. - */ - - hrtimer_init(&hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - hrtimer.function = sh_dac_audio_timer; - - return 0; -} - -static void __exit dac_audio_exit(void) -{ - unregister_sound_dsp(device_major); - kfree((void *)data_buffer); -} - -module_init(dac_audio_init); -module_exit(dac_audio_exit); - -MODULE_AUTHOR("Andriy Skulysh, askulysh@image.kiev.ua"); -MODULE_DESCRIPTION("SH DAC sound driver"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 4e7d7c6018567fa03f387d06602d4145c75ebbe0 Mon Sep 17 00:00:00 2001 From: Vince Weaver Date: Wed, 22 Sep 2010 17:31:37 -0400 Subject: ALSA: hda - MacBookPro 5,3 line-in support I've found the following patch is necessary to enable line-in on my MacBookPro 5,3 machine. With the patch applied I've successfully recorded audio from the line-in jack. This is based on the existing 5,5 support. Signed-off-by: Vince Weaver Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 483c3f2d8d39..5c00106cbc2e 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -65,6 +65,7 @@ struct cs_spec { /* available models */ enum { + CS420X_MBP53, CS420X_MBP55, CS420X_IMAC27, CS420X_AUTO, @@ -839,7 +840,8 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); } - if (spec->board_config == CS420X_MBP55 || + if (spec->board_config == CS420X_MBP53 || + spec->board_config == CS420X_MBP55 || spec->board_config == CS420X_IMAC27) { unsigned int gpio = hp_present ? 0x02 : 0x08; snd_hda_codec_write(codec, 0x01, 0, @@ -1128,6 +1130,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) } static const char *cs420x_models[CS420X_MODELS] = { + [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", [CS420X_AUTO] = "auto", @@ -1135,6 +1138,7 @@ static const char *cs420x_models[CS420X_MODELS] = { static struct snd_pci_quirk cs420x_cfg_tbl[] = { + SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), {} /* terminator */ @@ -1145,6 +1149,20 @@ struct cs_pincfg { u32 val; }; +static struct cs_pincfg mbp53_pincfgs[] = { + { 0x09, 0x012b4050 }, + { 0x0a, 0x90100141 }, + { 0x0b, 0x90100140 }, + { 0x0c, 0x018b3020 }, + { 0x0d, 0x90a00110 }, + { 0x0e, 0x400000f0 }, + { 0x0f, 0x01cbe030 }, + { 0x10, 0x014be060 }, + { 0x12, 0x400000f0 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, @@ -1174,6 +1192,7 @@ static struct cs_pincfg imac27_pincfgs[] = { }; static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { + [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; @@ -1206,6 +1225,7 @@ static int patch_cs420x(struct hda_codec *codec) switch (spec->board_config) { case CS420X_IMAC27: + case CS420X_MBP53: case CS420X_MBP55: /* GPIO1 = headphones */ /* GPIO3 = speakers */ -- cgit v1.2.3 From 01fdf1801e349302fce5d9865470a7100a2d9b74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Sep 2010 09:09:42 +0200 Subject: ALSA: hda - Fix auto-parse of SPDIF input of Realtek codecs The SPDIF in audio widget must be searched through the list as the widget that contains the given pin as the connection source. The current code was implemented in a reverse way. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a1312a6c8af2..a432e6efd19b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1594,12 +1594,22 @@ static void alc_auto_parse_digital(struct hda_codec *codec) } if (spec->autocfg.dig_in_pin) { - hda_nid_t dig_nid; - err = snd_hda_get_connections(codec, - spec->autocfg.dig_in_pin, - &dig_nid, 1); - if (err > 0) - spec->dig_in_nid = dig_nid; + dig_nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, dig_nid++) { + unsigned int wcaps = get_wcaps(codec, dig_nid); + if (get_wcaps_type(wcaps) != AC_WID_AUD_IN) + continue; + if (!(wcaps & AC_WCAP_DIGITAL)) + continue; + if (!(wcaps & AC_WCAP_CONN_LIST)) + continue; + err = get_connection_index(codec, dig_nid, + spec->autocfg.dig_in_pin); + if (err >= 0) { + spec->dig_in_nid = dig_nid; + break; + } + } } } -- cgit v1.2.3 From f41cc2a85d52ac6971299922084ac5ac59dc339d Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Fri, 24 Sep 2010 16:21:53 -0500 Subject: ALSA: hda - Fix switching between dmic and mic using the same mux on IDT/STAC Fix bug in switching between dmic and mic when both use the same mux. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a90327b0cc3e..d8dfafeab80e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3481,8 +3481,10 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, return err; } - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) + if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) { snd_hda_add_imux_item(imux, label, index, NULL); + spec->num_analog_muxes++; + } } return 0; -- cgit v1.2.3 From e68d3b316ab7b02a074edc4f770e6a746390cb7d Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Sat, 25 Sep 2010 11:07:27 -0400 Subject: ALSA: sound/pci/rme9652: prevent reading uninitialized stack memory The SNDRV_HDSP_IOCTL_GET_CONFIG_INFO and SNDRV_HDSP_IOCTL_GET_CONFIG_INFO ioctls in hdspm.c and hdsp.c allow unprivileged users to read uninitialized kernel stack memory, because several fields of the hdsp{m}_config_info structs declared on the stack are not altered or zeroed before being copied back to the user. This patch takes care of it. Signed-off-by: Dan Rosenberg Cc: Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 1 + sound/pci/rme9652/hdspm.c | 1 + 2 files changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b92adef8e81e..d6fa7bfd9aa1 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4609,6 +4609,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (err < 0) return err; + memset(&info, 0, sizeof(info)); spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 547b713d7204..0c98ef9156d8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4127,6 +4127,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, case SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO: + memset(&info, 0, sizeof(info)); spin_lock_irq(&hdspm->lock); info.pref_sync_ref = hdspm_pref_sync_ref(hdspm); info.wordclock_sync_check = hdspm_wc_sync_check(hdspm); -- cgit v1.2.3 From e35d4b119578a054515ccb4ed5dddc4e8a81ec15 Mon Sep 17 00:00:00 2001 From: Otavio Salvador Date: Sun, 26 Sep 2010 23:35:06 -0300 Subject: ALSA: hda: add Vortex86MX PCI ids Signed-off-by: Otavio Salvador Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5f6f9039a41a..ec07e4700e3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2791,6 +2791,8 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* this entry seems still valid -- i.e. without emu20kx chip */ { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, #endif + /* Vortex86MX */ + { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- cgit v1.2.3 From 5591bf07225523600450edd9e6ad258bb877b779 Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Tue, 28 Sep 2010 14:18:20 -0400 Subject: ALSA: prevent heap corruption in snd_ctl_new() The snd_ctl_new() function in sound/core/control.c allocates space for a snd_kcontrol struct by performing arithmetic operations on a user-provided size without checking for integer overflow. If a user provides a large enough size, an overflow will occur, the allocated chunk will be too small, and a second user-influenced value will be written repeatedly past the bounds of this chunk. This code is reachable by unprivileged users who have permission to open a /dev/snd/controlC* device (on many distros, this is group "audio") via the SNDRV_CTL_IOCTL_ELEM_ADD and SNDRV_CTL_IOCTL_ELEM_REPLACE ioctls. Signed-off-by: Dan Rosenberg Cc: Signed-off-by: Takashi Iwai --- sound/core/control.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 070aab490191..45a818002d99 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -31,6 +31,7 @@ /* max number of user-defined controls */ #define MAX_USER_CONTROLS 32 +#define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { struct list_head list; /* list of all ioctls */ @@ -195,6 +196,10 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, if (snd_BUG_ON(!control || !control->count)) return NULL; + + if (control->count > MAX_CONTROL_COUNT) + return NULL; + kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); if (kctl == NULL) { snd_printk(KERN_ERR "Cannot allocate control instance\n"); -- cgit v1.2.3 From c123e5e437a0e61e364c1cbad3ef9a7384975fb2 Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Tue, 28 Sep 2010 12:04:06 +1000 Subject: ALSA: hda - Add quirk for another Acer laptop with a CX20585 codec BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/647374 Set another Acer laptop (SSID 1025:043d) to use CXT5066_IDEAPAD Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a6c68cb06ddb..80cc74bf77a0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3094,6 +3094,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x1025, 0x043d, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), -- cgit v1.2.3 From 20d9a26dbbbec32aa7c9da49b979f201bd7104b9 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 30 Sep 2010 00:16:50 +0200 Subject: ALSA: snd-aloop - fix capture buffer silence In a special case, some old samples are left in the capture ring buffer. Fix it. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index f2b8f868d97a..2748fee8d405 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -347,7 +347,7 @@ static void copy_play_buf(struct loopback_pcm *play, unsigned int bytes) { struct snd_pcm_runtime *runtime = play->substream->runtime; - char *src = play->substream->runtime->dma_area; + char *src = runtime->dma_area; char *dst = capt->substream->runtime->dma_area; unsigned int src_off = play->buf_pos; unsigned int dst_off = capt->buf_pos; @@ -385,8 +385,10 @@ static void copy_play_buf(struct loopback_pcm *play, dst_off = (dst_off + size) % capt->pcm_buffer_size; } - if (clear_bytes > 0) + if (clear_bytes > 0) { clear_capture_buf(capt, clear_bytes); + capt->silent_size = 0; + } } #define BYTEPOS_UPDATE_POSONLY 0 -- cgit v1.2.3 From 4cb36310848fd17766aa72afd1f2873f54b4e055 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 30 Sep 2010 10:12:50 +0200 Subject: ALSA: HDA: Add position_fix=3 module option, and refactor related code What was previously known as via_dmapos_patch, and hard-coded to be used for VIA and ATI controllers, is now configurable through a module option. The background is that some VIA controllers seem to prefer via_dmapos_patch to be turned off. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 8 ++++--- sound/pci/hda/hda_intel.c | 41 ++++++++++++++++------------------- 2 files changed, 24 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 278cc2122ea0..c82beb007634 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -57,9 +57,11 @@ dead. However, this detection isn't perfect on some devices. In such a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. -`position_fix=2` means to use the position-buffer. 0 is the default -value, the automatic check and fallback to LPIB as described in the -above. If you get a problem of repeated sounds, this option might +`position_fix=2` means to use the position-buffer. +`position_fix=3` means to use a combination of both methods, needed +for some VIA and ATI controllers. 0 is the default value for all other +controllers, the automatic check and fallback to LPIB as described in +the above. If you get a problem of repeated sounds, this option might help. In addition to that, every controller is known to be broken regarding diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec07e4700e3b..38b063eb80e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -78,8 +78,8 @@ MODULE_PARM_DESC(enable, "Enable Intel HD audio interface."); module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); -MODULE_PARM_DESC(position_fix, "Fix DMA pointer " - "(0 = auto, 1 = none, 2 = POSBUF)."); +MODULE_PARM_DESC(position_fix, "DMA pointer read method." + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -305,6 +305,7 @@ enum { POS_FIX_AUTO, POS_FIX_LPIB, POS_FIX_POSBUF, + POS_FIX_VIACOMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -433,7 +434,6 @@ struct azx { unsigned int polling_mode :1; unsigned int msi :1; unsigned int irq_pending_warned :1; - unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */ unsigned int probing :1; /* codec probing phase */ /* for debugging */ @@ -1309,11 +1309,8 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) azx_sd_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ - if (chip->position_fix[0] == POS_FIX_POSBUF || - chip->position_fix[0] == POS_FIX_AUTO || - chip->position_fix[1] == POS_FIX_POSBUF || - chip->position_fix[1] == POS_FIX_AUTO || - chip->via_dmapos_patch) { + if (chip->position_fix[0] != POS_FIX_LPIB || + chip->position_fix[1] != POS_FIX_LPIB) { if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); @@ -1852,20 +1849,21 @@ static unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev) { unsigned int pos; + int stream = azx_dev->substream->stream; - if (chip->via_dmapos_patch) + switch (chip->position_fix[stream]) { + case POS_FIX_LPIB: + /* read LPIB */ + pos = azx_sd_readl(azx_dev, SD_LPIB); + break; + case POS_FIX_VIACOMBO: pos = azx_via_get_position(chip, azx_dev); - else { - int stream = azx_dev->substream->stream; - if (chip->position_fix[stream] == POS_FIX_POSBUF || - chip->position_fix[stream] == POS_FIX_AUTO) { - /* use the position buffer */ - pos = le32_to_cpu(*azx_dev->posbuf); - } else { - /* read LPIB */ - pos = azx_sd_readl(azx_dev, SD_LPIB); - } + break; + default: + /* use the position buffer */ + pos = le32_to_cpu(*azx_dev->posbuf); } + if (pos >= azx_dev->bufsize) pos = 0; return pos; @@ -2313,6 +2311,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) switch (fix) { case POS_FIX_LPIB: case POS_FIX_POSBUF: + case POS_FIX_VIACOMBO: return fix; } @@ -2320,11 +2319,9 @@ static int __devinit check_position_fix(struct azx *chip, int fix) switch (chip->driver_type) { case AZX_DRIVER_VIA: case AZX_DRIVER_ATI: - chip->via_dmapos_patch = 1; /* Use link position directly, avoid any transfer problem. */ - return POS_FIX_LPIB; + return POS_FIX_VIACOMBO; } - chip->via_dmapos_patch = 0; q = snd_pci_quirk_lookup(chip->pci, position_fix_list); if (q) { -- cgit v1.2.3 From e913b146493993c8ac33561655c590e58b500c6f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Sep 2010 22:59:12 +0200 Subject: ALSA: i2c/other/ak4xx-adda: Fix a compile warning with CONFIG_PROCFS=n Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4xxx-adda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 1adb8a3c2b62..42d7844ecd0b 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -900,7 +900,7 @@ static int proc_init(struct snd_akm4xxx *ak) return 0; } #else /* !CONFIG_PROC_FS */ -static int proc_init(struct snd_akm4xxx *ak) {} +static int proc_init(struct snd_akm4xxx *ak) { return 0; } #endif int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) -- cgit v1.2.3 From ac446fb7e690b317050ed158ba5dfd9273dc9e74 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sat, 2 Oct 2010 16:00:53 +0200 Subject: ALSA: snd-aloop - fix "PCM Slave Active" element read value Simple coding fix. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 2748fee8d405..040030aa9d8e 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -739,7 +739,7 @@ static int loopback_active_get(struct snd_kcontrol *kcontrol, { struct loopback *loopback = snd_kcontrol_chip(kcontrol); struct loopback_cable *cable = loopback->cables - [kcontrol->id.subdevice][kcontrol->id.device]; + [kcontrol->id.subdevice][kcontrol->id.device ^ 1]; unsigned int val = 0; if (cable != NULL) -- cgit v1.2.3 From d41185882b828896ccecac319c9f65f708baaf0d Mon Sep 17 00:00:00 2001 From: Valentine Sinitsyn Date: Fri, 1 Oct 2010 22:24:08 +0600 Subject: ALSA: hda - Added fixup for Lenovo Y550P Signed-off-by: Valentine Sinitsyn Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6427cc12a28..b4e0959b1f9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19246,6 +19246,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), {} }; -- cgit v1.2.3 From 422fdc318efd7d34d8b79decde0f8cb90a336c11 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:09:12 +0200 Subject: ALSA: usb-audio: add more Yamaha USB MIDI devices Add quirks for more devices (according to driver V.3.0.4-2). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c86c613e0b96..682e3e06b07c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -240,9 +240,21 @@ YAMAHA_DEVICE(0x104f, NULL), YAMAHA_DEVICE(0x1050, NULL), YAMAHA_DEVICE(0x1051, NULL), YAMAHA_DEVICE(0x1052, NULL), +YAMAHA_INTERFACE(0x1053, 0, NULL), +YAMAHA_INTERFACE(0x1054, 0, NULL), +YAMAHA_DEVICE(0x1055, NULL), +YAMAHA_DEVICE(0x1056, NULL), +YAMAHA_DEVICE(0x1057, NULL), +YAMAHA_DEVICE(0x1058, NULL), +YAMAHA_DEVICE(0x1059, NULL), +YAMAHA_DEVICE(0x105a, NULL), +YAMAHA_DEVICE(0x105b, NULL), +YAMAHA_DEVICE(0x105c, NULL), +YAMAHA_DEVICE(0x105d, NULL), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), +YAMAHA_DEVICE(0x2003, NULL), YAMAHA_DEVICE(0x5000, "CS1D"), YAMAHA_DEVICE(0x5001, "DSP1D"), YAMAHA_DEVICE(0x5002, "DME32"), -- cgit v1.2.3 From 45bc307f328c044e69cad2a18a9ae972bb15f254 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:17:26 +0200 Subject: ALSA: virtuoso: fix Xonar DS chip name The controller on the Xonar DS is labeled "AV66", not "AV200". Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 4 ++-- sound/pci/oxygen/xonar_wm87x6.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 06c863e86e3d..599bb9ab97ee 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -25,9 +25,9 @@ #include "xonar.h" MODULE_AUTHOR("Clemens Ladisch "); -MODULE_DESCRIPTION("Asus AVx00 driver"); +MODULE_DESCRIPTION("Asus Virtuoso driver"); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{Asus,AV100},{Asus,AV200}}"); +MODULE_SUPPORTED_DEVICE("{{Asus,AV66},{Asus,AV100},{Asus,AV200}}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index aceaaa036da6..5f9f59c10198 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1071,7 +1071,7 @@ static int xonar_ds_mixer_init(struct oxygen *chip) static const struct oxygen_model model_xonar_ds = { .shortname = "Xonar DS", - .longname = "Asus Virtuoso 200", + .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, .control_filter = xonar_ds_control_filter, -- cgit v1.2.3 From b6ca8ab399d913eed0d89d65d6b768337a3d20d7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:21:52 +0200 Subject: ALSA: oxygen: handle CD input configuration with a flag There are more models without a CD input than with one, so handle this explicitly with a device_config flag to avoid having to define a control filter callback to filter it out. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 3 ++- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_mixer.c | 3 +++ sound/pci/oxygen/xonar_cs43xx.c | 8 -------- sound/pci/oxygen/xonar_pcm179x.c | 11 ++--------- sound/pci/oxygen/xonar_wm87x6.c | 8 -------- 6 files changed, 8 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 289cb4dacfc7..f4fdf6dac800 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -505,7 +505,8 @@ static const struct oxygen_model model_generic = { PLAYBACK_2_TO_AC97_1 | CAPTURE_0_FROM_I2S_1 | CAPTURE_1_FROM_SPDIF | - CAPTURE_2_FROM_AC97_1, + CAPTURE_2_FROM_AC97_1 | + AC97_CD_INPUT, .dac_channels = 8, .dac_volume_min = 0, .dac_volume_max = 255, diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index a3409edcfb50..7d5222caa0a9 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -34,6 +34,7 @@ /* CAPTURE_3_FROM_I2S_3 not implemented */ #define MIDI_OUTPUT 0x0800 #define MIDI_INPUT 0x1000 +#define AC97_CD_INPUT 0x2000 enum { CONTROL_SPDIF_PCM, diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index f375b8a27862..7d40ba8db9fc 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -972,6 +972,9 @@ static int add_controls(struct oxygen *chip, if (!strcmp(template.name, "Stereo Upmixing") && chip->model.dac_channels == 2) continue; + if (!strncmp(template.name, "CD Capture ", 11) && + !(chip->model.device_config & AC97_CD_INPUT)) + continue; if (!strcmp(template.name, "Master Playback Volume") && chip->model.dac_tlv) { template.tlv.p = chip->model.dac_tlv; diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c index 7c4986b27f2b..aa27c31049af 100644 --- a/sound/pci/oxygen/xonar_cs43xx.c +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -367,13 +367,6 @@ static void xonar_d1_line_mic_ac97_switch(struct oxygen *chip, static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); -static int xonar_d1_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int xonar_d1_mixer_init(struct oxygen *chip) { int err; @@ -391,7 +384,6 @@ static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", .init = xonar_d1_init, - .control_filter = xonar_d1_control_filter, .mixer_init = xonar_d1_mixer_init, .cleanup = xonar_d1_cleanup, .suspend = xonar_d1_suspend, diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index ba18fb546b4f..338f88567f57 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -915,13 +915,6 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } -static int xonar_st_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int add_pcm1796_controls(struct oxygen *chip) { int err; @@ -991,7 +984,8 @@ static const struct oxygen_model model_xonar_d2 = { CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF | MIDI_OUTPUT | - MIDI_INPUT, + MIDI_INPUT | + AC97_CD_INPUT, .dac_channels = 8, .dac_volume_min = 255 - 2*60, .dac_volume_max = 255, @@ -1037,7 +1031,6 @@ static const struct oxygen_model model_xonar_st = { .longname = "Asus Virtuoso 100", .chip = "AV200", .init = xonar_st_init, - .control_filter = xonar_st_control_filter, .mixer_init = xonar_st_mixer_init, .cleanup = xonar_st_cleanup, .suspend = xonar_st_suspend, diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 5f9f59c10198..200f7601276f 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -1028,13 +1028,6 @@ static const struct snd_kcontrol_new lc_controls[] = { LC_CONTROL_ALC, wm8776_ngth_db_scale), }; -static int xonar_ds_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - static int xonar_ds_mixer_init(struct oxygen *chip) { struct xonar_wm87x6 *data = chip->model_data; @@ -1074,7 +1067,6 @@ static const struct oxygen_model model_xonar_ds = { .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, - .control_filter = xonar_ds_control_filter, .mixer_init = xonar_ds_mixer_init, .cleanup = xonar_ds_cleanup, .suspend = xonar_ds_suspend, -- cgit v1.2.3 From 2b830bae1fc2a27b3b0ab86091013bdec3c12427 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:22:51 +0200 Subject: ALSA: virtuoso: add HDAV1.3 Slim PCI ID Add a PCI ID for the Xonar HDAV1.3 Slim. There is no actual support, but the presence of the ID allows the EEPROM repair code to work for this card. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_pcm179x.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 599bb9ab97ee..469010a8b849 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -49,6 +49,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 338f88567f57..571d0ae42afb 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -132,6 +132,18 @@ * GPIO 5 <- 0 */ +/* + * Xonar HDAV1.3 Slim + * ------------------ + * + * CMI8788: + * + * GPIO 1 -> enable output + * + * TXD -> HDMI controller + * RXD <- HDMI controller + */ + #include #include #include @@ -1101,6 +1113,9 @@ int __devinit get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x835e: + snd_printk(KERN_ERR "the HDAV1.3 Slim is not supported\n"); + return -ENODEV; default: return -EINVAL; } -- cgit v1.2.3 From d737f3eedef0717c8b8233bb6455ff13637ff243 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:23:26 +0200 Subject: ALSA: virtuoso: fix Xonar STX anti-pop delay The anti-pop delay for the STX should be 800 ms, not 100 ms like the ST. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_pcm179x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index 571d0ae42afb..d491fd6c0be2 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -374,7 +374,6 @@ static void xonar_st_init_common(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; - data->generic.anti_pop_delay = 100; data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; data->dacs = chip->model.private_data ? 4 : 1; data->hp_gain_offset = 2*-18; @@ -420,6 +419,7 @@ static void xonar_st_init(struct oxygen *chip) { struct xonar_pcm179x *data = chip->model_data; + data->generic.anti_pop_delay = 100; data->has_cs2000 = 1; data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; @@ -440,6 +440,7 @@ static void xonar_stx_init(struct oxygen *chip) struct xonar_pcm179x *data = chip->model_data; xonar_st_init_i2c(chip); + data->generic.anti_pop_delay = 800; data->generic.ext_power_reg = OXYGEN_GPI_DATA; data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; data->generic.ext_power_bit = GPI_EXT_POWER; -- cgit v1.2.3 From de0074ee7ae7d61da40567afa53912d7e3e16b25 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:24:43 +0200 Subject: ALSA: oxygen: fix chip ID register symbols Rename the symbol for the XCID pins, fix up a decimal/hex confusion for the CMI8787 package ID, and add the other known package IDs. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_regs.h | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_regs.h b/sound/pci/oxygen/oxygen_regs.h index 72de159d4567..4dcd41b78258 100644 --- a/sound/pci/oxygen/oxygen_regs.h +++ b/sound/pci/oxygen/oxygen_regs.h @@ -436,13 +436,15 @@ /* OXYGEN_CHANNEL_* */ #define OXYGEN_CODEC_VERSION 0xe4 -#define OXYGEN_XCID_MASK 0x07 +#define OXYGEN_CODEC_ID_MASK 0x07 #define OXYGEN_REVISION 0xe6 -#define OXYGEN_REVISION_XPKGID_MASK 0x0007 +#define OXYGEN_PACKAGE_ID_MASK 0x0007 +#define OXYGEN_PACKAGE_ID_8786 0x0004 +#define OXYGEN_PACKAGE_ID_8787 0x0006 +#define OXYGEN_PACKAGE_ID_8788 0x0007 #define OXYGEN_REVISION_MASK 0xfff8 -#define OXYGEN_REVISION_2 0x0008 /* bit flag */ -#define OXYGEN_REVISION_8787 0x0014 /* 8 bits */ +#define OXYGEN_REVISION_2 0x0008 #define OXYGEN_OFFSIN_48K 0xe8 #define OXYGEN_OFFSBASE_48K 0xe9 -- cgit v1.2.3 From 9a0b37926595b57c4b5fc56aa6fd243bed4ee4eb Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 4 Oct 2010 13:25:13 +0200 Subject: ALSA: oxygen: fix input monitor dB scale The input monitor half volume bit results in a factor of 0.5, so the minimum scale value should be -6 dB. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 7d40ba8db9fc..2849b36f5f7e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -708,7 +708,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, .private_value = ((codec) << 24) | ((stereo) << 16) | (index), \ } -static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0); +static DECLARE_TLV_DB_SCALE(monitor_db_scale, -600, 600, 0); static DECLARE_TLV_DB_SCALE(ac97_db_scale, -3450, 150, 0); static DECLARE_TLV_DB_SCALE(ac97_rec_db_scale, 0, 150, 0); -- cgit v1.2.3 From 93943beb29be7084afb61556e96bc454079bfb0e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 6 Oct 2010 10:57:11 +0200 Subject: ALSA: oxygen: reduce minimum period count The interrupt counter is independent of the buffer counter, so there are no restrictions on the period size. Having fewer periods also makes PulseAudio happy. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 9dff6954c397..814667442eb0 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -56,8 +56,8 @@ static const struct snd_pcm_hardware oxygen_stereo_hardware = { .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_multichannel_hardware = { @@ -82,8 +82,8 @@ static const struct snd_pcm_hardware oxygen_multichannel_hardware = { .channels_max = 8, .buffer_bytes_max = BUFFER_BYTES_MAX_MULTICH, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX_MULTICH / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX_MULTICH, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX_MULTICH / PERIOD_BYTES_MIN, }; static const struct snd_pcm_hardware oxygen_ac97_hardware = { @@ -100,8 +100,8 @@ static const struct snd_pcm_hardware oxygen_ac97_hardware = { .channels_max = 2, .buffer_bytes_max = BUFFER_BYTES_MAX, .period_bytes_min = PERIOD_BYTES_MIN, - .period_bytes_max = BUFFER_BYTES_MAX / 2, - .periods_min = 2, + .period_bytes_max = BUFFER_BYTES_MAX, + .periods_min = 1, .periods_max = BUFFER_BYTES_MAX / PERIOD_BYTES_MIN, }; -- cgit v1.2.3 From 7cb4ced5aa83b681c76b004c8960b4f2a6471fef Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 6 Oct 2010 10:57:50 +0200 Subject: ALSA: oxygen: rewrite PCIe bridge initialization Change the PCIe/PCI bridge initialization code to configure only the bridge that is actually connected to the sound chip, instead of any bridge found in the system. The new code also makes it easier to add other bridges. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 40 +++++++++++++++++++++++----------------- 1 file changed, 23 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 7e93cf884437..d10cc6ee1a68 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -308,25 +308,31 @@ static void oxygen_restore_eeprom(struct oxygen *chip, } } -static void pci_bridge_magic(void) +static void configure_pcie_bridge(struct pci_dev *pci) { - struct pci_dev *pci = NULL; + enum { PI7C9X110 }; + static const struct pci_device_id bridge_ids[] = { + { PCI_DEVICE(0x12d8, 0xe110), .driver_data = PI7C9X110 }, + { } + }; + struct pci_dev *bridge; + const struct pci_device_id *id; u32 tmp; - for (;;) { - /* If there is any Pericom PI7C9X110 PCI-E/PCI bridge ... */ - pci = pci_get_device(0x12d8, 0xe110, pci); - if (!pci) - break; - /* - * ... configure its secondary internal arbiter to park to - * the secondary port, instead of to the last master. - */ - if (!pci_read_config_dword(pci, 0x40, &tmp)) { - tmp |= 1; - pci_write_config_dword(pci, 0x40, tmp); - } - /* Why? Try asking C-Media. */ + if (!pci->bus || !pci->bus->self) + return; + bridge = pci->bus->self; + + id = pci_match_id(bridge_ids, bridge); + if (!id) + return; + + switch (id->driver_data) { + case PI7C9X110: /* Pericom PI7C9X110 PCIe/PCI bridge */ + pci_read_config_dword(bridge, 0x40, &tmp); + tmp |= 1; /* park the PCI arbiter to the sound chip */ + pci_write_config_dword(bridge, 0x40, tmp); + break; } } @@ -613,7 +619,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; - pci_bridge_magic(); + configure_pcie_bridge(pci); oxygen_init(chip); chip->model.init(chip); -- cgit v1.2.3 From ebebeece4ba596973c0c181a8cce5fd77bae427c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 6 Oct 2010 10:58:50 +0200 Subject: ALSA: oxygen: add PEX8111 initialization Configure the PEX8111 bridge on the PCI Express cards so that the audio DMA controller can do proper burst reads and is less likely to lose data. This is usually done automatically, but is required on older cards where the user has not applied the PLX firmware update. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index d10cc6ee1a68..e5ebe56fb0c5 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -310,8 +310,10 @@ static void oxygen_restore_eeprom(struct oxygen *chip, static void configure_pcie_bridge(struct pci_dev *pci) { - enum { PI7C9X110 }; + enum { PEX811X, PI7C9X110 }; static const struct pci_device_id bridge_ids[] = { + { PCI_VDEVICE(PLX, 0x8111), .driver_data = PEX811X }, + { PCI_VDEVICE(PLX, 0x8112), .driver_data = PEX811X }, { PCI_DEVICE(0x12d8, 0xe110), .driver_data = PI7C9X110 }, { } }; @@ -328,6 +330,19 @@ static void configure_pcie_bridge(struct pci_dev *pci) return; switch (id->driver_data) { + case PEX811X: /* PLX PEX8111/PEX8112 PCIe/PCI bridge */ + pci_read_config_dword(bridge, 0x48, &tmp); + tmp |= 1; /* enable blind prefetching */ + tmp |= 1 << 11; /* enable beacon generation */ + pci_write_config_dword(bridge, 0x48, tmp); + + pci_write_config_dword(bridge, 0x84, 0x0c); + pci_read_config_dword(bridge, 0x88, &tmp); + tmp &= ~(7 << 27); + tmp |= 2 << 27; /* set prefetch size to 128 bytes */ + pci_write_config_dword(bridge, 0x88, tmp); + break; + case PI7C9X110: /* Pericom PI7C9X110 PCIe/PCI bridge */ pci_read_config_dword(bridge, 0x40, &tmp); tmp |= 1; /* park the PCI arbiter to the sound chip */ -- cgit v1.2.3 From dd1d3a49db4ae5c6afffadaff526b96c7993c7dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 6 Oct 2010 17:28:56 +0200 Subject: ALSA: oxygen - Add a SSID for CMI8787-HG2PCI This board has a strange PCI SSID 13f6:ffff. Works as compabile as MODEL_CMEDIA_REF. Reported-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index f4fdf6dac800..1d915efb2695 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -79,6 +79,7 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x13f6, 0x0001), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x0010), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x13f6, 0x8788), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x13f6, 0xffff), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, -- cgit v1.2.3 From ec588ae6c21ae20a22ce13a287728a220935b8ee Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 6 Oct 2010 16:47:26 +0300 Subject: ASoC: omap: Remove needless prints from machine drivers It is currently completely normal to execute these machine drivers code on different boards if the kernel includes support for multiple boards so no error message should be printed if the machine_is_xxx does not match with the machine driver. Therefore remove these pr_err and pr_debug prints in those cases. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/am3517evm.c | 4 +--- sound/soc/omap/igep0020.c | 4 +--- sound/soc/omap/omap2evm.c | 4 +--- sound/soc/omap/omap3beagle.c | 4 +--- sound/soc/omap/omap3evm.c | 4 +--- sound/soc/omap/sdp3430.c | 4 +--- sound/soc/omap/sdp4430.c | 4 +--- sound/soc/omap/zoom2.c | 4 +--- 8 files changed, 8 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 68bd902ccd4e..979dd508305f 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -157,10 +157,8 @@ static int __init am3517evm_soc_init(void) { int ret; - if (!machine_is_omap3517evm()) { - pr_err("Not OMAP3517 / AM3517 EVM!\n"); + if (!machine_is_omap3517evm()) return -ENODEV; - } pr_info("OMAP3517 / AM3517 EVM SoC init\n"); am3517evm_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index d296cfcc672e..fd3a40f309c8 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -101,10 +101,8 @@ static int __init igep2_soc_init(void) { int ret; - if (!machine_is_igep0020()) { - pr_debug("Not IGEP v2!\n"); + if (!machine_is_igep0020()) return -ENODEV; - } printk(KERN_INFO "IGEP v2 SoC init\n"); igep2_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 38cd1894623e..cf3fc8a675b5 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -103,10 +103,8 @@ static int __init omap2evm_soc_init(void) { int ret; - if (!machine_is_omap2evm()) { - pr_debug("Not omap2evm!\n"); + if (!machine_is_omap2evm()) return -ENODEV; - } printk(KERN_INFO "omap2evm SoC init\n"); omap2evm_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 7c11e1afe9e6..e56832b0c444 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -112,10 +112,8 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { - pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) return -ENODEV; - } pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 1ac5babef00d..810f1e36da21 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -99,10 +99,8 @@ static int __init omap3evm_soc_init(void) { int ret; - if (!machine_is_omap3evm()) { - pr_err("Not OMAP3 EVM!\n"); + if (!machine_is_omap3evm()) return -ENODEV; - } pr_info("OMAP3 EVM SoC init\n"); omap3evm_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 76ce77b91844..07fbcf7d2411 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -296,10 +296,8 @@ static int __init sdp3430_soc_init(void) int ret; u8 pin_mux; - if (!machine_is_omap_3430sdp()) { - pr_debug("Not SDP3430!\n"); + if (!machine_is_omap_3430sdp()) return -ENODEV; - } printk(KERN_INFO "SDP3430 SoC init\n"); sdp3430_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 62f6a622d791..4b4463db6ba0 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -186,10 +186,8 @@ static int __init sdp4430_soc_init(void) { int ret; - if (!machine_is_omap_4430sdp()) { - pr_debug("Not SDP4430!\n"); + if (!machine_is_omap_4430sdp()) return -ENODEV; - } printk(KERN_INFO "SDP4430 SoC init\n"); sdp4430_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 338dc9552bd6..718031eeac34 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -245,10 +245,8 @@ static int __init zoom2_soc_init(void) { int ret; - if (!machine_is_omap_zoom2()) { - pr_debug("Not Zoom2!\n"); + if (!machine_is_omap_zoom2()) return -ENODEV; - } printk(KERN_INFO "Zoom2 SoC init\n"); zoom2_snd_device = platform_device_alloc("soc-audio", -1); -- cgit v1.2.3 From b1c73fc8e697eb73e23603e465e9af2711ed4183 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 11 Oct 2010 10:45:00 +0200 Subject: ALSA: snd-aloop: Fix hw_params restrictions and checking This patch fixes the hw_params restrictions when first (or playback) stream sets the final hardware parameters. Also, fix the hw_params checking in the trigger callback. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 124 ++++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 106 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 040030aa9d8e..3c0088272095 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -188,7 +188,7 @@ static inline void loopback_timer_stop(struct loopback_pcm *dpcm) static int loopback_check_format(struct loopback_cable *cable, int stream) { - struct snd_pcm_runtime *runtime; + struct snd_pcm_runtime *runtime, *cruntime; struct loopback_setup *setup; struct snd_card *card; int check; @@ -200,11 +200,11 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) } runtime = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]-> substream->runtime; - check = cable->hw.formats != (1ULL << runtime->format) || - cable->hw.rate_min != runtime->rate || - cable->hw.rate_max != runtime->rate || - cable->hw.channels_min != runtime->channels || - cable->hw.channels_max != runtime->channels; + cruntime = cable->streams[SNDRV_PCM_STREAM_CAPTURE]-> + substream->runtime; + check = runtime->format != cruntime->format || + runtime->rate != cruntime->rate || + runtime->channels != cruntime->channels; if (!check) return 0; if (stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -274,12 +274,42 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static void params_change_substream(struct loopback_pcm *dpcm, + struct snd_pcm_runtime *runtime) +{ + struct snd_pcm_runtime *dst_runtime; + + if (dpcm == NULL || dpcm->substream == NULL) + return; + dst_runtime = dpcm->substream->runtime; + if (dst_runtime == NULL) + return; + dst_runtime->hw = dpcm->cable->hw; +} + +static void params_change(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct loopback_pcm *dpcm = runtime->private_data; + struct loopback_cable *cable = dpcm->cable; + + cable->hw.formats = (1ULL << runtime->format); + cable->hw.rate_min = runtime->rate; + cable->hw.rate_max = runtime->rate; + cable->hw.channels_min = runtime->channels; + cable->hw.channels_max = runtime->channels; + params_change_substream(cable->streams[SNDRV_PCM_STREAM_PLAYBACK], + runtime); + params_change_substream(cable->streams[SNDRV_PCM_STREAM_CAPTURE], + runtime); +} + static int loopback_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; - unsigned int bps, salign; + int bps, salign; salign = (snd_pcm_format_width(runtime->format) * runtime->channels) / 8; @@ -303,13 +333,10 @@ static int loopback_prepare(struct snd_pcm_substream *substream) dpcm->pcm_period_size = frames_to_bytes(runtime, runtime->period_size); mutex_lock(&dpcm->loopback->cable_lock); - if (!(cable->valid & ~(1 << substream->stream))) { - cable->hw.formats = (1ULL << runtime->format); - cable->hw.rate_min = runtime->rate; - cable->hw.rate_max = runtime->rate; - cable->hw.channels_min = runtime->channels; - cable->hw.channels_max = runtime->channels; - } + if (!(cable->valid & ~(1 << substream->stream)) || + (get_setup(dpcm)->notify && + substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + params_change(substream); cable->valid |= 1 << substream->stream; mutex_unlock(&dpcm->loopback->cable_lock); @@ -542,6 +569,47 @@ static unsigned int get_cable_index(struct snd_pcm_substream *substream) return !substream->stream; } +static int rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + + struct snd_pcm_hardware *hw = rule->private; + struct snd_mask *maskp = hw_param_mask(params, rule->var); + + maskp->bits[0] &= (u_int32_t)hw->formats; + maskp->bits[1] &= (u_int32_t)(hw->formats >> 32); + memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX-64) / 8); /* clear rest */ + if (! maskp->bits[0] && ! maskp->bits[1]) + return -EINVAL; + return 0; +} + +static int rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hardware *hw = rule->private; + struct snd_interval t; + + t.min = hw->rate_min; + t.max = hw->rate_max; + t.openmin = t.openmax = 0; + t.integer = 0; + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +static int rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_pcm_hardware *hw = rule->private; + struct snd_interval t; + + t.min = hw->channels_min; + t.max = hw->channels_max; + t.openmin = t.openmax = 0; + t.integer = 0; + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + static int loopback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -579,14 +647,34 @@ static int loopback_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + /* use dynamic rules based on actual runtime->hw values */ + /* note that the default rules created in the PCM midlevel code */ + /* are cached -> they do not reflect the actual state */ + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + rule_format, &runtime->hw, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (err < 0) + goto unlock; + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + rule_rate, &runtime->hw, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (err < 0) + goto unlock; + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + rule_channels, &runtime->hw, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (err < 0) + goto unlock; + runtime->private_data = dpcm; runtime->private_free = loopback_runtime_free; - if (get_notify(dpcm) && - substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (get_notify(dpcm)) runtime->hw = loopback_pcm_hardware; - } else { + else runtime->hw = cable->hw; - } unlock: mutex_unlock(&loopback->cable_lock); return err; -- cgit v1.2.3 From 838c364ff05c143fd1810e8ad1469935d6c23a7a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 8 Oct 2010 10:48:50 +0200 Subject: ALSA: OSS mixer emulation - fix locking Fix mutex release and cleanup some locking code. Cc: Signed-off-by: Jaroslav Kysela --- sound/core/oss/mixer_oss.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f50ebf20df96..8442a088677d 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -618,8 +618,10 @@ static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; down_read(&card->controls_rwsem); - if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) + if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) { + up_read(&card->controls_rwsem); return; + } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) @@ -658,7 +660,7 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, return; down_read(&card->controls_rwsem); if ((kctl = snd_ctl_find_numid(card, numid)) == NULL) { - up_read(&fmixer->card->controls_rwsem); + up_read(&card->controls_rwsem); return; } uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); @@ -797,7 +799,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) { err = -ENOMEM; - goto __unlock; + goto __free_only; } down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); @@ -826,6 +828,7 @@ static int snd_mixer_oss_get_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned err = 0; __unlock: up_read(&card->controls_rwsem); + __free_only: kfree(uctl); kfree(uinfo); return err; @@ -847,7 +850,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); if (uinfo == NULL || uctl == NULL) { err = -ENOMEM; - goto __unlock; + goto __free_only; } down_read(&card->controls_rwsem); kctl = snd_mixer_oss_test_id(mixer, "Capture Source", 0); @@ -880,6 +883,7 @@ static int snd_mixer_oss_put_recsrc2(struct snd_mixer_oss_file *fmixer, unsigned err = 0; __unlock: up_read(&card->controls_rwsem); + __free_only: kfree(uctl); kfree(uinfo); return err; -- cgit v1.2.3 From d4cfa4d12f46e2520f4c1d1a92e891ce068b7464 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 10 Oct 2010 19:33:52 +0200 Subject: OSS: soundcard: locking bug in sound_ioctl() We shouldn't return directly here because we're still holding the &soundcard_mutex. This bug goes all the way back to the start of git. It's strange that no one has complained about it as a runtime bug. CC: stable@kernel.org Signed-off-by: Dan Carpenter Acked-by: Arnd Bergmann Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 92aa762ffb7e..07f803e6d203 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -391,11 +391,11 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return audio_ioctl(dev, file, cmd, p); + ret = audio_ioctl(dev, file, cmd, p); break; case SND_DEV_MIDIN: - return MIDIbuf_ioctl(dev, file, cmd, p); + ret = MIDIbuf_ioctl(dev, file, cmd, p); break; } -- cgit v1.2.3 From b7d22ccf08d67d13f77a9580d07e7f72e6241213 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 10 Oct 2010 19:34:22 +0200 Subject: OSS: soundcard: fix return value of sound_open() Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index a5ab61ed0a97..46c0d03dbecc 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -249,7 +249,7 @@ static int sound_open(struct inode *inode, struct file *file) } mutex_unlock(&soundcard_mutex); - return 0; + return retval; } static int sound_release(struct inode *inode, struct file *file) -- cgit v1.2.3 From 9b2167d59f38691b86430ce559c7fa9d4f973b1f Mon Sep 17 00:00:00 2001 From: Luke Yelavich Date: Wed, 6 Oct 2010 15:45:46 +1100 Subject: ALSA: hda - Add another HP DV6 quirk BugLink: https://bugs.launchpad.net/bugs/653420 Add another HP DV6 notebook (103c:363e) to use STAC_HP_DV5. Signed-off-by: Luke Yelavich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 95148e58026c..c16c5ba0fda0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1747,6 +1747,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv6", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x363e, + "HP DV6", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- cgit v1.2.3 From 1d2019fb6be2f318f0aa85be5f224f47a5f006fe Mon Sep 17 00:00:00 2001 From: Nicolas Kaiser Date: Tue, 5 Oct 2010 17:38:12 +0200 Subject: ALSA: sound/usb/usx2y: simplify conditional Simplify conditional: (a || (!a && b)) => (a || b) Signed-off-by: Nicolas Kaiser Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 2a528e56afd5..3146a816068b 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -54,7 +54,7 @@ #include #include "usbusx2yaudio.c" -#if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1) +#if defined(USX2Y_NRPACKS_VARIABLE) || USX2Y_NRPACKS == 1 #include -- cgit v1.2.3 From f46119b73425df9d1e05c5d5e909a993d95b0218 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Oct 2010 14:46:35 +0200 Subject: ALSA: hda - Add model=mbp55 entry for MacBookPro 7,1 Reference: Novell bnc#645066 https://bugzilla.novell.com/show_bug.cgi?id=645066 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 5c00106cbc2e..d9a3dbcc9ba0 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1140,6 +1140,7 @@ static const char *cs420x_models[CS420X_MODELS] = { static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), + SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), {} /* terminator */ }; -- cgit v1.2.3 From 6a92934d9e987b6363db3e6a08e17bc0f2078c5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 11 Oct 2010 15:16:20 +0200 Subject: ALSA: hda - Add input volume control for each mic/line-in pin The input pins on cirrus codecs have also input amps. Let's make control elemetns for them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d9a3dbcc9ba0..460fb2ef7e39 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -742,6 +742,27 @@ static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, return bind; } +/* add a (input-boost) volume control to the given input pin */ +static int add_input_volume_control(struct hda_codec *codec, + struct auto_pin_cfg *cfg, + int item) +{ + hda_nid_t pin = cfg->inputs[item].pin; + u32 caps; + const char *label; + struct snd_kcontrol *kctl; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + label = hda_get_autocfg_input_label(codec, cfg, item); + return add_volume(codec, label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + static int build_input(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; @@ -781,6 +802,12 @@ static int build_input(struct hda_codec *codec) return err; } + for (i = 0; i < spec->num_inputs; i++) { + err = add_input_volume_control(codec, &spec->autocfg, i); + if (err < 0) + return err; + } + return 0; } -- cgit v1.2.3 From bdd9ef24cd343c508ed93f1e08f30d4db595b754 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 4 Oct 2010 12:02:14 +0200 Subject: ALSA: HDA: Correctly apply position_fix quirks for ATI and VIA controllers Position_fix quirks for specific machines now override the default position_fix behavior for all HDA controllers. BugLink: http://launchpad.net/bugs/465942 BugLink: http://launchpad.net/bugs/580749 BugLink: http://launchpad.net/bugs/587546 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 38b063eb80e9..5cbea85a6453 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2315,14 +2315,6 @@ static int __devinit check_position_fix(struct azx *chip, int fix) return fix; } - /* Check VIA/ATI HD Audio Controller exist */ - switch (chip->driver_type) { - case AZX_DRIVER_VIA: - case AZX_DRIVER_ATI: - /* Use link position directly, avoid any transfer problem. */ - return POS_FIX_VIACOMBO; - } - q = snd_pci_quirk_lookup(chip->pci, position_fix_list); if (q) { printk(KERN_INFO @@ -2331,6 +2323,15 @@ static int __devinit check_position_fix(struct azx *chip, int fix) q->value, q->subvendor, q->subdevice); return q->value; } + + /* Check VIA/ATI HD Audio Controller exist */ + switch (chip->driver_type) { + case AZX_DRIVER_VIA: + case AZX_DRIVER_ATI: + /* Use link position directly, avoid any transfer problem. */ + return POS_FIX_VIACOMBO; + } + return POS_FIX_AUTO; } -- cgit v1.2.3 From cf4bb69884c8f6a5791e0e251f0b9dc5d32fc256 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 13 Oct 2010 11:56:28 +0300 Subject: ASoC: tlv320dac33: Control for line output gain New control to select the line output gain. This gain control affects the linein-to-lineout and dac-to-loneout gain differently. Use enum type to select the desired gain combination. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index bf6d01f4a45a..58349dcd1a6e 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -524,6 +524,22 @@ static const struct soc_enum dac33_fifo_mode_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), dac33_fifo_mode_texts); +/* L/R Line Output Gain */ +static const char *lr_lineout_gain_texts[] = { + "Line -12dB DAC 0dB", "Line -6dB DAC 6dB", + "Line 0dB DAC 12dB", "Line 6dB DAC 18dB", +}; + +static const struct soc_enum l_lineout_gain_enum = + SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0, + ARRAY_SIZE(lr_lineout_gain_texts), + lr_lineout_gain_texts); + +static const struct soc_enum r_lineout_gain_enum = + SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0, + ARRAY_SIZE(lr_lineout_gain_texts), + lr_lineout_gain_texts); + /* * DACL/R digital volume control: * from 0 dB to -63.5 in 0.5 dB steps @@ -541,6 +557,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), SOC_DOUBLE_R("Line to Line Out Volume", DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), + SOC_ENUM("Left Line Output Gain", l_lineout_gain_enum), + SOC_ENUM("Right Line Output Gain", r_lineout_gain_enum), }; static const struct snd_kcontrol_new dac33_mode_snd_controls[] = { -- cgit v1.2.3 From 2df03514de41f3bbb5623f2e7f2bf594e49cb2ec Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 10 Oct 2010 22:39:28 -0400 Subject: ALSA: hda: Add speaker pin to automute Acer Aspire 8943G BugLink: https://bugs.launchpad.net/bugs/656625 Add clause for handling Acer Aspire 8943G's subwoofer as additional speaker pin for automuting. Reported-by: RussianNeuroMancer Cc: stable@kernel.org Signed-off-by: Daniel T Chen Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4e0959b1f9f..c41ac30ffc7f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19233,10 +19233,17 @@ static void alc662_auto_init(struct hda_codec *codec) } enum { + ALC662_FIXUP_ASPIRE, ALC662_FIXUP_IDEAPAD, }; static const struct alc_fixup alc662_fixups[] = { + [ALC662_FIXUP_ASPIRE] = { + .pins = (const struct alc_pincfg[]) { + { 0x15, 0x99130112 }, /* subwoofer */ + { } + } + }, [ALC662_FIXUP_IDEAPAD] = { .pins = (const struct alc_pincfg[]) { { 0x17, 0x99130112 }, /* subwoofer */ @@ -19246,6 +19253,7 @@ static const struct alc_fixup alc662_fixups[] = { }; static struct snd_pci_quirk alc662_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), {} -- cgit v1.2.3 From 0db710230589b5571c23f59250eabc9504b17c98 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 14 Oct 2010 21:46:12 +0200 Subject: ALSA: snd-aloop - fix issue in the timer start function In some circumstances (the rate shift value was changed), the irq_pos value may be higher than the fraction value in the timer start function. Check for it. Also, to avoid value overflow, decrease maximum period size. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 3c0088272095..838ad86311b8 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -171,6 +171,10 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) dpcm->pcm_rate_shift = rate_shift; dpcm->period_size_frac = frac_pos(dpcm, dpcm->pcm_period_size); } + if (dpcm->period_size_frac <= dpcm->irq_pos) { + dpcm->irq_pos %= dpcm->period_size_frac; + dpcm->period_update_pending = 1; + } tick = dpcm->period_size_frac - dpcm->irq_pos; tick = (tick + dpcm->pcm_bps - 1) / dpcm->pcm_bps; dpcm->timer.expires = jiffies + tick; @@ -531,7 +535,9 @@ static struct snd_pcm_hardware loopback_pcm_hardware = .channels_max = 32, .buffer_bytes_max = 2 * 1024 * 1024, .period_bytes_min = 64, - .period_bytes_max = 2 * 1024 * 1024, + /* note check overflow in frac_pos() using pcm_rate_shift before + changing period_bytes_max value */ + .period_bytes_max = 1024 * 1024, .periods_min = 1, .periods_max = 1024, .fifo_size = 0, -- cgit v1.2.3 From fa2eb005ebcbac89745a1f9a9f0c8678ba63f61a Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 16 Oct 2010 15:19:20 +0200 Subject: sound: fixed typos Signed-off-by: Andrea Gelmini Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_mixer.c | 2 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/usb/usx2y/usx2yhwdeppcm.c | 4 ++-- 5 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c index c92f493d341e..557c782ae4fc 100644 --- a/sound/pci/au88x0/au88x0_mixer.c +++ b/sound/pci/au88x0/au88x0_mixer.c @@ -23,7 +23,7 @@ static int __devinit snd_vortex_mixer(vortex_t * vortex) if ((err = snd_ac97_bus(vortex->card, 0, &ops, NULL, &pbus)) < 0) return err; memset(&ac97, 0, sizeof(ac97)); - // Intialize AC97 codec stuff. + // Initialize AC97 codec stuff. ac97.private_data = vortex; ac97.scaps = AC97_SCAP_NO_SPDIF; err = snd_ac97_mixer(pbus, &ac97, &vortex->codec); diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 40eccfe9e358..4948a79f86a0 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -150,7 +150,7 @@ static int __init sffsdr_init(void) sffsdr_snd_resources, ARRAY_SIZE(sffsdr_snd_resources)); if (ret) { - printk(KERN_ERR "platform device add ressources failed\n"); + printk(KERN_ERR "platform device add resources failed\n"); goto error; } diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 209c25994c7e..4719558289d4 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -182,7 +182,7 @@ static int neo1973_gta02_voice_hw_params( if (ret < 0) return ret; - /* configue and enable PLL for 12.288MHz output */ + /* configure and enable PLL for 12.288MHz output */ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0cb4f86f6d1e..4ac620988e7c 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -201,7 +201,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* configue and enable PLL for 12.288MHz output */ + /* configure and enable PLL for 12.288MHz output */ ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 3146a816068b..287ef73b1237 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -36,9 +36,9 @@ plain usx2y alsa mode is able to achieve 64frames, 4periods, but only at the cost of easier triggered i.e. aeolus xruns (128 or 256frames, 2periods works but is useless cause of crackling). - + This is a first "proof of concept" implementation. - Later, funcionalities should migrate to more apropriate places: + Later, functionalities should migrate to more apropriate places: Userland: - The jackd could mmap its float-pcm buffers directly from alsa-lib. - alsa-lib could provide power of 2 period sized shaping combined with int/float -- cgit v1.2.3 From aa73aec6c385e2c797ac25cc7ccf0318031de7c8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 15 Oct 2010 12:06:18 +0200 Subject: ALSA: rawmidi: fix oops (use after free) when unloading a driver module When a driver module is unloaded and the last still open file is a raw MIDI device, the card and its devices will be actually freed in the snd_card_file_remove() call when that file is closed. Afterwards, rmidi and rmidi->card point into freed memory, so the module pointer is likely to be garbage. (This was introduced by commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a.) Signed-off-by: Clemens Ladisch Reported-by: Krzysztof Foltman Cc: 2.6.30-2.6.35 Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index a7868ad4d530..cbbed0db9e56 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -535,13 +535,15 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file) { struct snd_rawmidi_file *rfile; struct snd_rawmidi *rmidi; + struct module *module; rfile = file->private_data; rmidi = rfile->rmidi; rawmidi_release_priv(rfile); kfree(rfile); + module = rmidi->card->module; snd_card_file_remove(rmidi->card, file); - module_put(rmidi->card->module); + module_put(module); return 0; } -- cgit v1.2.3 From 906229174c20e3d5cbda8da070af8f30196316c3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 14 Oct 2010 14:50:18 +0200 Subject: ALSA: HDA: Enable SKU quirks for Realtek Realtek have ways of specifying external amps and more via a special nid or via the Codec's subsystem ID, this is called "SKU". The computer manufacturer sometimes gets this wrong, so we need to be able to override or ignore the SKU customization value. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 +++++++++++++++++++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c41ac30ffc7f..7b24a2d72936 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -301,6 +301,7 @@ struct alc_customize_define { unsigned int platform_type:1; unsigned int swap:1; unsigned int override:1; + unsigned int fixup:1; /* Means that this sku is set by driver, not read from hw */ }; struct alc_spec { @@ -1464,6 +1465,11 @@ static void alc_init_auto_mic(struct hda_codec *codec) spec->unsol_event = alc_sku_unsol_event; } +/* Could be any non-zero and even value. When used as fixup, tells + * the driver to ignore any present sku defines. + */ +#define ALC_FIXUP_SKU_IGNORE (2) + static int alc_auto_parse_customize_define(struct hda_codec *codec) { unsigned int ass, tmp, i; @@ -1472,6 +1478,13 @@ static int alc_auto_parse_customize_define(struct hda_codec *codec) spec->cdefine.enable_pcbeep = 1; /* assume always enabled */ + if (spec->cdefine.fixup) { + ass = spec->cdefine.sku_cfg; + if (ass == ALC_FIXUP_SKU_IGNORE) + return -1; + goto do_sku; + } + ass = codec->subsystem_id & 0xffff; if (ass != codec->bus->pci->subsystem_device && (ass & 1)) goto do_sku; @@ -1539,6 +1552,13 @@ static int alc_subsystem_id(struct hda_codec *codec, unsigned nid; struct alc_spec *spec = codec->spec; + if (spec->cdefine.fixup) { + ass = spec->cdefine.sku_cfg; + if (ass == ALC_FIXUP_SKU_IGNORE) + return 0; + goto do_sku; + } + ass = codec->subsystem_id & 0xffff; if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) goto do_sku; @@ -1658,6 +1678,7 @@ struct alc_pincfg { }; struct alc_fixup { + unsigned int sku; const struct alc_pincfg *pins; const struct hda_verb *verbs; }; @@ -1668,12 +1689,22 @@ static void alc_pick_fixup(struct hda_codec *codec, int pre_init) { const struct alc_pincfg *cfg; + struct alc_spec *spec; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (!quirk) return; fix += quirk->value; cfg = fix->pins; + if (pre_init && fix->sku) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply sku override for %s\n", + codec->chip_name, quirk->name); +#endif + spec = codec->spec; + spec->cdefine.sku_cfg = fix->sku; + spec->cdefine.fixup = 1; + } if (pre_init && cfg) { #ifdef CONFIG_SND_DEBUG_VERBOSE snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", @@ -10861,8 +10892,6 @@ static int patch_alc882(struct hda_codec *codec) codec->spec = spec; - alc_auto_parse_customize_define(codec); - switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: @@ -10890,6 +10919,8 @@ static int patch_alc882(struct hda_codec *codec) if (board_config == ALC882_AUTO) alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); + alc_auto_parse_customize_define(codec); + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); -- cgit v1.2.3 From c3d226ab8b44fe31e5e6d5739eb353597cea4029 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 14 Oct 2010 15:42:08 +0200 Subject: ALSA: HDA: Apply SKU override for Acer aspire 7736z BugLink: http://launchpad.net/bugs/617647 The current SKU value disables playback, so ignore the SKU value. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b24a2d72936..788ac4bbbba9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10628,6 +10628,7 @@ static struct alc_config_preset alc882_presets[] = { enum { PINFIX_ABIT_AW9D_MAX, PINFIX_PB_M5210, + PINFIX_ACER_ASPIRE_7736, }; static const struct alc_fixup alc882_fixups[] = { @@ -10645,11 +10646,15 @@ static const struct alc_fixup alc882_fixups[] = { {} } }, + [PINFIX_ACER_ASPIRE_7736] = { + .sku = ALC_FIXUP_SKU_IGNORE, + }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), {} }; -- cgit v1.2.3 From de8c85f7840e5e29629de95f5af24297fb325e0b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 15 Oct 2010 10:32:50 +0200 Subject: ALSA: HDA: Sigmatel: work around incorrect master muting The HDA specification does not allow for a codec to mute itself just because the volume is reduced, so _of course_ somebody had to go and do it. This wouldn'\''t hurt too much when the volume is adjusted by hand, but programs like PA that try to set the volume automatically could inadvertently mute the output. To work around this, change the TLV dB information for the Master volume on all Sigmatel HDA codecs to indicate the the minimal volume setting actually mutes. Reported-by: Colin Guthrie Reported-by: "Alexander E. Patrakov" Tested-by: Colin Guthrie Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ sound/pci/hda/hda_local.h | 14 ++++++++++---- sound/pci/hda/patch_sigmatel.c | 6 ++++-- 3 files changed, 17 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 053f827d2c2c..8c933c8006f4 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1831,6 +1831,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, hda_nid_t nid = get_amp_nid(kcontrol); int dir = get_amp_direction(kcontrol); unsigned int ofs = get_amp_offset(kcontrol); + bool min_mute = get_amp_min_mute(kcontrol); u32 caps, val1, val2; if (size < 4 * sizeof(unsigned int)) @@ -1841,6 +1842,8 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); val1 += ofs; val1 = ((int)val1) * ((int)val2); + if (min_mute) + val2 |= 0x10000; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; if (put_user(2 * sizeof(unsigned int), _tlv + 1)) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d7dfa547e2d8..46bbefe2e4a9 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -38,10 +38,11 @@ */ #define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs) \ ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23)) +#define HDA_AMP_VAL_MIN_MUTE (1<<29) #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \ HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0) /* mono volume with index (index=0,1,...) (channel=1,2) */ -#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ +#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, dir, flags) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ .subdevice = HDA_SUBDEV_AMP_FLAG, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ @@ -51,16 +52,20 @@ .get = snd_hda_mixer_amp_volume_get, \ .put = snd_hda_mixer_amp_volume_put, \ .tlv = { .c = snd_hda_mixer_amp_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, dir) | flags } /* stereo volume with index */ #define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \ - HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction) + HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction, 0) /* mono volume */ #define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \ - HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction) + HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction, 0) /* stereo volume */ #define HDA_CODEC_VOLUME(xname, nid, xindex, direction) \ HDA_CODEC_VOLUME_MONO(xname, nid, 3, xindex, direction) +/* stereo volume with min=mute */ +#define HDA_CODEC_VOLUME_MIN_MUTE(xname, nid, xindex, direction) \ + HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, 3, xindex, direction, \ + HDA_AMP_VAL_MIN_MUTE) /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -581,6 +586,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) #define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) +#define get_amp_min_mute(kc) (((kc)->private_value >> 29) & 0x1) /* * CEA Short Audio Descriptor data diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d8dfafeab80e..1a563a2fbbec 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -992,7 +992,7 @@ static struct hda_verb stac9205_core_init[] = { } static struct snd_kcontrol_new stac9200_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), @@ -1020,7 +1020,7 @@ static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), { } /* end */ }; @@ -1144,6 +1144,8 @@ static int stac92xx_build_controls(struct hda_codec *codec) HDA_OUTPUT, vmaster_tlv); /* correct volume offset */ vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; + /* minimum value is actually mute */ + vmaster_tlv[3] |= 0x1000; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) -- cgit v1.2.3 From c08d91695b2a3349254a62b60f03f7971bd90fa0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Oct 2010 10:40:53 +0200 Subject: ALSA: tlv - Define numbers in sound/tlv.h Signed-off-by: Takashi Iwai --- include/sound/tlv.h | 4 +++- sound/pci/hda/hda_codec.c | 2 +- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 9fd5b19ccf5c..7067e2dfb0b9 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -38,9 +38,11 @@ #define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ #define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ +#define TLV_DB_SCALE_MASK 0xffff +#define TLV_DB_SCALE_MUTE 0x10000 #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ - (min), ((step) & 0xffff) | ((mute) ? 0x10000 : 0) + (min), ((step) & TLV_DB_SCALE_MASK) | ((mute) ? TLV_DB_SCALE_MUTE : 0) #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8c933c8006f4..ee134a25092c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1843,7 +1843,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val1 += ofs; val1 = ((int)val1) * ((int)val2); if (min_mute) - val2 |= 0x10000; + val2 |= TLV_DB_SCALE_MUTE; if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; if (put_user(2 * sizeof(unsigned int), _tlv + 1)) -- cgit v1.2.3 From e74670b6fdc37b15ebee11825849d8983e52a74a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 18 Oct 2010 09:43:10 +0200 Subject: ALSA: snd-aloop: add cable#0 and cable#1 files to proc card tree Show some useful runtime information using procfs. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 84 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 84 insertions(+) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 838ad86311b8..66786ea6f480 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -39,6 +39,7 @@ #include #include #include +#include #include MODULE_AUTHOR("Jaroslav Kysela "); @@ -184,6 +185,7 @@ static void loopback_timer_start(struct loopback_pcm *dpcm) static inline void loopback_timer_stop(struct loopback_pcm *dpcm) { del_timer(&dpcm->timer); + dpcm->timer.expires = 0; } #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) @@ -1011,6 +1013,86 @@ static int __devinit loopback_mixer_new(struct loopback *loopback, int notify) return 0; } +#ifdef CONFIG_PROC_FS + +static void print_dpcm_info(struct snd_info_buffer *buffer, + struct loopback_pcm *dpcm, + const char *id) +{ + snd_iprintf(buffer, " %s\n", id); + if (dpcm == NULL) { + snd_iprintf(buffer, " inactive\n"); + return; + } + snd_iprintf(buffer, " buffer_size:\t%u\n", dpcm->pcm_buffer_size); + snd_iprintf(buffer, " buffer_pos:\t\t%u\n", dpcm->buf_pos); + snd_iprintf(buffer, " silent_size:\t%u\n", dpcm->silent_size); + snd_iprintf(buffer, " period_size:\t%u\n", dpcm->pcm_period_size); + snd_iprintf(buffer, " bytes_per_sec:\t%u\n", dpcm->pcm_bps); + snd_iprintf(buffer, " sample_align:\t%u\n", dpcm->pcm_salign); + snd_iprintf(buffer, " rate_shift:\t\t%u\n", dpcm->pcm_rate_shift); + snd_iprintf(buffer, " update_pending:\t%u\n", + dpcm->period_update_pending); + snd_iprintf(buffer, " irq_pos:\t\t%u\n", dpcm->irq_pos); + snd_iprintf(buffer, " period_frac:\t%u\n", dpcm->period_size_frac); + snd_iprintf(buffer, " last_jiffies:\t%lu (%lu)\n", + dpcm->last_jiffies, jiffies); + snd_iprintf(buffer, " timer_expires:\t%lu\n", dpcm->timer.expires); +} + +static void print_substream_info(struct snd_info_buffer *buffer, + struct loopback *loopback, + int sub, + int num) +{ + struct loopback_cable *cable = loopback->cables[sub][num]; + + snd_iprintf(buffer, "Cable %i substream %i:\n", num, sub); + if (cable == NULL) { + snd_iprintf(buffer, " inactive\n"); + return; + } + snd_iprintf(buffer, " valid: %u\n", cable->valid); + snd_iprintf(buffer, " running: %u\n", cable->running); + print_dpcm_info(buffer, cable->streams[0], "Playback"); + print_dpcm_info(buffer, cable->streams[1], "Capture"); +} + +static void print_cable_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct loopback *loopback = entry->private_data; + int sub, num; + + mutex_lock(&loopback->cable_lock); + num = entry->name[strlen(entry->name)-1]; + num = num == '0' ? 0 : 1; + for (sub = 0; sub < MAX_PCM_SUBSTREAMS; sub++) + print_substream_info(buffer, loopback, sub, num); + mutex_unlock(&loopback->cable_lock); +} + +static int __devinit loopback_proc_new(struct loopback *loopback, int cidx) +{ + char name[32]; + struct snd_info_entry *entry; + int err; + + snprintf(name, sizeof(name), "cable#%d", cidx); + err = snd_card_proc_new(loopback->card, name, &entry); + if (err < 0) + return err; + + snd_info_set_text_ops(entry, loopback, print_cable_info); + return 0; +} + +#else /* !CONFIG_PROC_FS */ + +#define loopback_proc_new(loopback, cidx) do { } while (0) + +#endif + static int __devinit loopback_probe(struct platform_device *devptr) { struct snd_card *card; @@ -1041,6 +1123,8 @@ static int __devinit loopback_probe(struct platform_device *devptr) err = loopback_mixer_new(loopback, pcm_notify[dev] ? 1 : 0); if (err < 0) goto __nodev; + loopback_proc_new(loopback, 0); + loopback_proc_new(loopback, 1); strcpy(card->driver, "Loopback"); strcpy(card->shortname, "Loopback"); sprintf(card->longname, "Loopback %i", dev + 1); -- cgit v1.2.3 From 1cc9e8f4c45999e6069f41521d9d391eeeccc3b3 Mon Sep 17 00:00:00 2001 From: Charles Date: Mon, 18 Oct 2010 16:22:00 +0800 Subject: ALSA: hda - Fix codec muted after rebooting from Windows Windows may leave pin power-down registers set after reboot, and this resulted in muted output on Linux. Reset these registers at initialization properly. Signed-off-by: Charles Chin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1a563a2fbbec..7d70f8ca3742 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5334,6 +5334,9 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; + /* reset pin power-down; Windows may leave these bits after reboot */ + snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7EC, 0); + snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0); codec->no_trigger_sense = 1; codec->spec = spec; spec->linear_tone_beep = 1; -- cgit v1.2.3 From dd04bb12d047a4d4461772093472a40dbe171e5f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 20 Oct 2010 08:27:02 +0200 Subject: ALSA: snd-aloop - fix locking issues (running flag updates) On SMP machines, the cable->running update must be atomic, otherwise stream is not started correctly sometimes. Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 32 +++++++++++++++++++------------- 1 file changed, 19 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 66786ea6f480..38e8351e935d 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -263,13 +263,17 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) return err; dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; - loopback_timer_start(dpcm); + spin_lock(&cable->lock); cable->running |= (1 << substream->stream); + spin_unlock(&cable->lock); + loopback_timer_start(dpcm); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_STOP: + spin_lock(&cable->lock); cable->running &= ~(1 << substream->stream); + spin_unlock(&cable->lock); loopback_timer_stop(dpcm); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); @@ -454,28 +458,30 @@ static void loopback_bytepos_update(struct loopback_pcm *dpcm, } } -static void loopback_pos_update(struct loopback_cable *cable) +static unsigned int loopback_pos_update(struct loopback_cable *cable) { struct loopback_pcm *dpcm_play = cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; unsigned long delta_play = 0, delta_capt = 0; + unsigned int running; spin_lock(&cable->lock); - if (cable->running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { + running = cable->running; + if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { delta_play = jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } - if (cable->running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { + if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { delta_capt = jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } if (delta_play == 0 && delta_capt == 0) { spin_unlock(&cable->lock); - return; + return running; } if (delta_play > delta_capt) { @@ -490,27 +496,27 @@ static void loopback_pos_update(struct loopback_cable *cable) if (delta_play == 0 && delta_capt == 0) { spin_unlock(&cable->lock); - return; + return running; } /* note delta_capt == delta_play at this moment */ loopback_bytepos_update(dpcm_capt, delta_capt, BYTEPOS_UPDATE_COPY); loopback_bytepos_update(dpcm_play, delta_play, BYTEPOS_UPDATE_POSONLY); spin_unlock(&cable->lock); + return running; } static void loopback_timer_function(unsigned long data) { struct loopback_pcm *dpcm = (struct loopback_pcm *)data; - int stream; + unsigned int running; - loopback_pos_update(dpcm->cable); - stream = dpcm->substream->stream; - if (dpcm->cable->running & (1 << stream)) + running = loopback_pos_update(dpcm->cable); + if (running & (1 << dpcm->substream->stream)) { loopback_timer_start(dpcm); - if (dpcm->period_update_pending) { - dpcm->period_update_pending = 0; - if (dpcm->cable->running & (1 << stream)) + if (dpcm->period_update_pending) { + dpcm->period_update_pending = 0; snd_pcm_period_elapsed(dpcm->substream); + } } } -- cgit v1.2.3 From 5de9e45fcfccdf8151a82fc1a521e7042cbe482a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 20 Oct 2010 09:33:03 +0200 Subject: ALSA: snd-aloop - add pause support Signed-off-by: Jaroslav Kysela --- sound/drivers/aloop.c | 27 ++++++++++++++++++++++----- 1 file changed, 22 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 38e8351e935d..12b44b0b6777 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -77,6 +77,7 @@ struct loopback_cable { /* flags */ unsigned int valid; unsigned int running; + unsigned int pause; }; struct loopback_setup { @@ -254,7 +255,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; - int err; + int err, stream = 1 << substream->stream; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -264,7 +265,8 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; spin_lock(&cable->lock); - cable->running |= (1 << substream->stream); + cable->running |= stream; + cable->pause &= ~stream; spin_unlock(&cable->lock); loopback_timer_start(dpcm); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -272,12 +274,26 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: spin_lock(&cable->lock); - cable->running &= ~(1 << substream->stream); + cable->running &= ~stream; + cable->pause &= ~stream; spin_unlock(&cable->lock); loopback_timer_stop(dpcm); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock(&cable->lock); + cable->pause |= stream; + spin_unlock(&cable->lock); + loopback_timer_stop(dpcm); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock(&cable->lock); + dpcm->last_jiffies = jiffies; + cable->pause &= ~stream; + spin_unlock(&cable->lock); + loopback_timer_start(dpcm); + break; default: return -EINVAL; } @@ -468,7 +484,7 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable) unsigned int running; spin_lock(&cable->lock); - running = cable->running; + running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { delta_play = jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; @@ -532,7 +548,7 @@ static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) static struct snd_pcm_hardware loopback_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID), + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), @@ -1060,6 +1076,7 @@ static void print_substream_info(struct snd_info_buffer *buffer, } snd_iprintf(buffer, " valid: %u\n", cable->valid); snd_iprintf(buffer, " running: %u\n", cable->running); + snd_iprintf(buffer, " pause: %u\n", cable->pause); print_dpcm_info(buffer, cable->streams[0], "Playback"); print_dpcm_info(buffer, cable->streams[1], "Capture"); } -- cgit v1.2.3 From bf1b022588eba78c990fd58fd2471cd92c2c5683 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 21 Oct 2010 08:49:56 +0200 Subject: ALSA: hda - Add alc_init_jacks() call to other codecs Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 788ac4bbbba9..2363f1893e88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11006,6 +11006,8 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC882_AUTO) spec->init_hook = alc882_auto_init; + + alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc882_loopbacks; @@ -12914,6 +12916,8 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) spec->init_hook = alc262_auto_init; + + alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; @@ -13993,6 +13997,8 @@ static int patch_alc268(struct hda_codec *codec) if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; + alc_init_jacks(codec); + return 0; } @@ -14359,6 +14365,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, HDA_AMP_MUTE, bits); + alc_report_jack(codec, nid); } /* unsolicited event for HP jack sensing */ @@ -14695,7 +14702,6 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); alc_auto_init_digital(codec); - alc_init_jacks(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -15128,6 +15134,8 @@ static int patch_alc269(struct hda_codec *codec) #endif if (board_config == ALC269_AUTO) spec->init_hook = alc269_auto_init; + + alc_init_jacks(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc269_loopbacks; @@ -19393,6 +19401,8 @@ static int patch_alc662(struct hda_codec *codec) alc_pick_fixup(codec, alc662_fixup_tbl, alc662_fixups, 0); } + alc_init_jacks(codec); + #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc662_loopbacks; -- cgit v1.2.3 From 693194f3b8af349a510604dffad9bdbbcf1c7db8 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 21 Oct 2010 08:51:48 +0200 Subject: ALSA: hda - Fix codec rename rules for ALC662-compatible codecs Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2363f1893e88..5f00589cb791 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19309,6 +19309,7 @@ static int patch_alc662(struct hda_codec *codec) { struct alc_spec *spec; int err, board_config; + int coef; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) @@ -19320,12 +19321,15 @@ static int patch_alc662(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); - if (alc_read_coef_idx(codec, 0) == 0x8020) + coef = alc_read_coef_idx(codec, 0); + if (coef == 0x8020 || coef == 0x8011) alc_codec_rename(codec, "ALC661"); - else if ((alc_read_coef_idx(codec, 0) & (1 << 14)) && - codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) + else if (coef & (1 << 14) && + codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) alc_codec_rename(codec, "ALC272X"); + else if (coef == 0x4011) + alc_codec_rename(codec, "ALC656"); board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, -- cgit v1.2.3 From 24b55c69b66eb2a122842820ec14ab215fc8572f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Oct 2010 08:55:13 +0200 Subject: ALSA: hda - Fix wrong SPDIF NID assignment for CA0110 The dig_out_nid field must take a digital-converter widget, but the current ca0110 parser passed the pin wrongly instead. Reported-by: Wai Yew CHAY Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index cca11fdd3d79..46c8bf48c31f 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -489,7 +489,7 @@ static void parse_digital(struct hda_codec *codec) if (cfg->dig_outs && snd_hda_get_connections(codec, cfg->dig_out_pins[0], &spec->dig_out, 1) == 1) - spec->multiout.dig_out_nid = cfg->dig_out_pins[0]; + spec->multiout.dig_out_nid = spec->dig_out; } static int ca0110_parse_auto_config(struct hda_codec *codec) -- cgit v1.2.3 From 14d34f166c57e77e3d7f9bc8b43d349186d922c1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Oct 2010 09:03:25 +0200 Subject: ALSA: hda - Add some workarounds for Creative IBG Creative HD-audio controller chips require some workarounds: - Additional delay before RIRB response - Set the initial RIRB counter to 0xc0 The latter seems to be done in general in Windows driver, so we may use this value later for all types if it's confirmed to work better. Reported-by: Wai Yew CHAY Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5cbea85a6453..ee445bc6e810 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -458,6 +458,7 @@ enum { AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, + AZX_DRIVER_CTX, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -473,6 +474,7 @@ static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -563,7 +565,10 @@ static void azx_init_cmd_io(struct azx *chip) /* reset the rirb hw write pointer */ azx_writew(chip, RIRBWP, ICH6_RIRBWP_RST); /* set N=1, get RIRB response interrupt for new entry */ - azx_writew(chip, RINTCNT, 1); + if (chip->driver_type == AZX_DRIVER_CTX) + azx_writew(chip, RINTCNT, 0xc0); + else + azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); spin_unlock_irq(&chip->reg_lock); @@ -1136,8 +1141,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) + if (status & RIRB_INT_RESPONSE) { + if (chip->driver_type == AZX_DRIVER_CTX) + udelay(80); azx_update_rirb(chip); + } azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } @@ -2784,10 +2792,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, - .driver_data = AZX_DRIVER_GENERIC }, + .driver_data = AZX_DRIVER_CTX }, #else /* this entry seems still valid -- i.e. without emu20kx chip */ - { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, + { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_CTX }, #endif /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, -- cgit v1.2.3 From 00cd0bb7307970b745aefd2b3e1db929792b43bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Oct 2010 09:57:40 +0200 Subject: ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066 Multiple Acer laptops with the SSID 1025:04xx require the quirk mode=ideapad, so let's use mask to apply to all these. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 80cc74bf77a0..6361f752b5f3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3093,8 +3093,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x040a, "Acer", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1025, 0x043d, "Acer", CXT5066_IDEAPAD), + SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), -- cgit v1.2.3 From 872a64d7e73f0e4dc435295da393db25332172b8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 21 Oct 2010 15:03:03 +0300 Subject: ASoC: tpa6130a2: Error handling for broken chip Correct/Implement handling of broken chip. Fail the i2c_prope if the communication with the chip fails. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 99b70e5978a2..329acc1a2074 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -98,16 +98,21 @@ static u8 tpa6130a2_read(int reg) return data->regs[reg]; } -static void tpa6130a2_initialize(void) +static int tpa6130a2_initialize(void) { struct tpa6130a2_data *data; - int i; + int i, ret = 0; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); - for (i = 1; i < TPA6130A2_REG_VERSION; i++) - tpa6130a2_i2c_write(i, data->regs[i]); + for (i = 1; i < TPA6130A2_REG_VERSION; i++) { + ret = tpa6130a2_i2c_write(i, data->regs[i]); + if (ret < 0) + break; + } + + return ret; } static int tpa6130a2_power(int power) @@ -133,7 +138,16 @@ static int tpa6130a2_power(int power) } data->power_state = 1; - tpa6130a2_initialize(); + ret = tpa6130a2_initialize(); + if (ret < 0) { + dev_err(&tpa6130a2_client->dev, + "Failed to initialize chip\n"); + if (data->power_gpio >= 0) + gpio_set_value(data->power_gpio, 0); + regulator_disable(data->supply); + data->power_state = 0; + goto exit; + } /* Clear SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); @@ -375,7 +389,9 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) { struct tpa6130a2_data *data; - BUG_ON(tpa6130a2_client == NULL); + if (tpa6130a2_client == NULL) + return -ENODEV; + data = i2c_get_clientdata(tpa6130a2_client); snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, -- cgit v1.2.3 From a74ccea51d4314632a81d568d59bf885e5b09d93 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 22 Oct 2010 15:52:34 +0200 Subject: ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs The bit value set for TLV mute was wrong in commit de8c85f7840e5e29629de95f5af24297fb325e0b, which resulted in bogus dB ranges that screw up PulseAudio. Corrected with the right constant. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7d70f8ca3742..7f487ab4dad4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -32,6 +32,7 @@ #include #include #include +#include #include "hda_codec.h" #include "hda_local.h" #include "hda_beep.h" @@ -1145,7 +1146,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) /* correct volume offset */ vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; /* minimum value is actually mute */ - vmaster_tlv[3] |= 0x1000; + vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) -- cgit v1.2.3 From 62b7e5e09bcb854ff05e6ee1aa161f8283dc36ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Oct 2010 17:15:47 +0200 Subject: ALSA: hda - Add workarounds for CT-IBG controllers Creative IBG controllers require the playback stream-tags to be started from 1, instead of capture+1. Otherwise the stream stalls. Reported-by: Wai Yew CHAY Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 ++++- sound/pci/hda/hda_intel.c | 9 +++++++-- 2 files changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ee134a25092c..13c1e7703c49 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1216,6 +1216,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, struct hda_codec *c; struct hda_cvt_setup *p; unsigned int oldval, newval; + int type; int i; if (!nid) @@ -1254,10 +1255,12 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, p->dirty = 0; /* make other inactive cvts with the same stream-tag dirty */ + type = get_wcaps_type(get_wcaps(codec, nid)); list_for_each_entry(c, &codec->bus->codec_list, list) { for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); - if (!p->active && p->stream_tag == stream_tag) + if (!p->active && p->stream_tag == stream_tag && + get_wcaps_type(get_wcaps(codec, p->nid)) == type) p->dirty = 1; } } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ee445bc6e810..21aa9b0e28f6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1652,7 +1652,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int bufsize, period_bytes, format_val; + unsigned int bufsize, period_bytes, format_val, stream_tag; int err; azx_stream_reset(chip, azx_dev); @@ -1694,7 +1694,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) else azx_dev->fifo_size = 0; - return snd_hda_codec_prepare(apcm->codec, hinfo, azx_dev->stream_tag, + stream_tag = azx_dev->stream_tag; + /* CA-IBG chips need the playback stream starting from 1 */ + if (chip->driver_type == AZX_DRIVER_CTX && + stream_tag > chip->capture_streams) + stream_tag -= chip->capture_streams; + return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, azx_dev->format_val, substream); } -- cgit v1.2.3 From c7f572168fc4840727c9bda955b7f103922209cd Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 22 Oct 2010 18:20:48 +0200 Subject: ALSA: usb-audio: add Novation Launchpad support Add a quirk entry for the Novation Launchpad USB MIDI controller. QUIRK_MIDI_FASTLANE gets renamed to *_RAW_BYTES because this quirk type is now shared by different devices. Signed-off-by: Clemens Ladisch Tested-by: Jakob Flierl Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 7 ++++--- sound/usb/quirks-table.h | 11 ++++++++++- sound/usb/quirks.c | 2 +- sound/usb/usbaudio.h | 2 +- 4 files changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 156cd0716c42..25bce7e5b1af 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -784,7 +784,7 @@ static struct usb_protocol_ops snd_usbmidi_novation_ops = { }; /* - * "raw" protocol: used by the MOTU FastLane. + * "raw" protocol: just move raw MIDI bytes from/to the endpoint */ static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep, @@ -2122,7 +2122,7 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_novation_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; - case QUIRK_MIDI_FASTLANE: + case QUIRK_MIDI_RAW_BYTES: umidi->usb_protocol_ops = &snd_usbmidi_raw_ops; /* * Interface 1 contains isochronous endpoints, but with the same @@ -2133,7 +2133,8 @@ int snd_usbmidi_create(struct snd_card *card, * interface 0, so we have to make sure that the USB core looks * again at interface 0 by calling usb_set_interface() on it. */ - usb_set_interface(umidi->dev, 0, 0); + if (umidi->usb_id == USB_ID(0x07fd, 0x0001)) /* MOTU Fastlane */ + usb_set_interface(umidi->dev, 0, 0); err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 682e3e06b07c..ad7079d1676c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2078,7 +2078,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = & (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_MIDI_FASTLANE + .type = QUIRK_MIDI_RAW_BYTES }, { .ifnum = 1, @@ -2226,6 +2226,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_NOVATION } }, +{ + USB_DEVICE(0x1235, 0x000e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Novation", */ + /* .product_name = "Launchpad", */ + .ifnum = 0, + .type = QUIRK_MIDI_RAW_BYTES + } +}, { USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 9a9da09586a5..cf8bf088394b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -287,7 +287,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, - [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, + [QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_MIDI_AKAI] = create_any_midi_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 24d3319cc34d..db3eb21627ee 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -70,7 +70,7 @@ enum quirk_type { QUIRK_MIDI_YAMAHA, QUIRK_MIDI_MIDIMAN, QUIRK_MIDI_NOVATION, - QUIRK_MIDI_FASTLANE, + QUIRK_MIDI_RAW_BYTES, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, QUIRK_MIDI_AKAI, -- cgit v1.2.3 From 84eae18c867fcb7ce43d5830e23377ed33e45df9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 22 Oct 2010 15:11:20 +0300 Subject: ASoC: tlv320dac33: Use usleep_range for delays Switch to use the more precise usleep_range instead of msleep(). Replace the udelay with usleep_range to remove the busy loop waiting. Signed-off-by: Peter Ujfalusi Acked-by: Mark Borwn Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 58349dcd1a6e..d251ff54a2d3 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -669,6 +669,7 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) { struct snd_soc_codec *codec = dac33->codec; + unsigned int delay; switch (dac33->fifo_mode) { case DAC33_FIFO_MODE1: @@ -684,8 +685,9 @@ static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33) dac33_write16(codec, DAC33_PREFILL_MSB, DAC33_THRREG(dac33->alarm_threshold)); /* Enable Alarm Threshold IRQ with a delay */ - udelay(SAMPLES_TO_US(dac33->burst_rate, - dac33->alarm_threshold)); + delay = SAMPLES_TO_US(dac33->burst_rate, + dac33->alarm_threshold) + 1000; + usleep_range(delay, delay + 500); dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); break; case DAC33_FIFO_MODE7: @@ -785,11 +787,11 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) static void dac33_oscwait(struct snd_soc_codec *codec) { - int timeout = 20; + int timeout = 60; u8 reg; do { - msleep(1); + usleep_range(1000, 2000); dac33_read(codec, DAC33_INT_OSC_STATUS, ®); } while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--); if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) -- cgit v1.2.3 From 23156e8faed5df60364976bffea0711a4f38d88a Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:28 +1100 Subject: ALSA: ca0106 - add Sound Blaster 5.1vx info. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 8e69620da20b..6dc9a5d01af5 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -239,6 +239,16 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .gpio_type = 1, .i2c_adc = 1, .spi_dac = 1 } , + /* Sound Blaster 5.1vx + * Tested: Playback on front, rear, center/lfe speakers + * Not-Tested: Capture + */ + { .serial = 0x10041102, + .name = "Sound Blaster 5.1vx [SB1070]", + .gpio_type = 1, + .i2c_adc = 0, + .spi_dac = 1 + } , /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ /* SB0438 * CTRL:CA0106-DAT -- cgit v1.2.3 From 51630142ed7da31618c0aca8f2767824834e18a8 Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:29 +1100 Subject: ALSA: ca0106: Pull out dac powering routine into separate function. This is ground work for a future commit where cards (such as the Sound Blaster 5.1vx) have different mappings between dacs and channels. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 39 +++++++++++++++++++++++++++------------ 1 file changed, 27 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 6dc9a5d01af5..22d2f6b6a05f 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -514,6 +514,23 @@ static void restore_spdif_bits(struct snd_ca0106 *chip, int idx) } } +static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, + int power) +{ + if (chip->details->spi_dac) { + const int reg = spi_dacd_reg[channel_id]; + + if (power) + /* Power up */ + chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; + else + /* Power down */ + chip->spi_dac_reg[reg] |= spi_dacd_bit[channel_id]; + return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + } + return 0; +} + /* open_playback callback */ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, int channel_id) @@ -553,12 +570,9 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr return err; snd_pcm_set_sync(substream); - if (chip->details->spi_dac && channel_id != PCM_FRONT_CHANNEL) { - const int reg = spi_dacd_reg[channel_id]; - - /* Power up dac */ - chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; - err = snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + /* Front channel dac should already be on */ + if (channel_id != PCM_FRONT_CHANNEL) { + err = snd_ca0106_pcm_power_dac(chip, channel_id, 1); if (err < 0) return err; } @@ -578,13 +592,14 @@ static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) restore_spdif_bits(chip, epcm->channel_id); - if (chip->details->spi_dac && epcm->channel_id != PCM_FRONT_CHANNEL) { - const int reg = spi_dacd_reg[epcm->channel_id]; - - /* Power down DAC */ - chip->spi_dac_reg[reg] |= spi_dacd_bit[epcm->channel_id]; - snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + /* Front channel dac should stay on */ + if (epcm->channel_id != PCM_FRONT_CHANNEL) { + int err; + err = snd_ca0106_pcm_power_dac(chip, epcm->channel_id, 0); + if (err < 0) + return err; } + /* FIXME: maybe zero others */ return 0; } -- cgit v1.2.3 From 9bfd94132dd97b76af41024eb7e980a5cb41afee Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:30 +1100 Subject: ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 22d2f6b6a05f..46ae98d9cb49 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1385,7 +1385,7 @@ static unsigned int spi_dac_init[] = { SPI_REG(12, 0x00), SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB), SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE), - SPI_REG(SPI_DACD4_REG, 0x00), + SPI_REG(SPI_DACD4_REG, SPI_DACD4_BIT), }; static unsigned int i2c_adc_init[][2] = { @@ -1576,6 +1576,9 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) if (reg < ARRAY_SIZE(chip->spi_dac_reg)) chip->spi_dac_reg[reg] = spi_dac_init[n]; } + + /* Enable front dac only */ + snd_ca0106_pcm_power_dac(chip, PCM_FRONT_CHANNEL, 1); } } -- cgit v1.2.3 From 861391d3a037fab38020c741baffdb147e1c732a Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:31 +1100 Subject: ALSA: ca0106: Create a nice spot for mapping channels to dacs. This is to allow a future patch to have card specific mappings between dacs, which is required since the Sound Blaster 5.1vx seems to have a different mapping to what was previously used. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 40 +++++++++++++++++++++++++++++----------- 1 file changed, 29 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 46ae98d9cb49..da910031edfa 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -493,16 +493,18 @@ static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) } static const int spi_dacd_reg[] = { - [PCM_FRONT_CHANNEL] = SPI_DACD4_REG, - [PCM_REAR_CHANNEL] = SPI_DACD0_REG, - [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_REG, - [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_REG, + SPI_DACD0_REG, + SPI_DACD1_REG, + SPI_DACD2_REG, + 0, + SPI_DACD4_REG, }; static const int spi_dacd_bit[] = { - [PCM_FRONT_CHANNEL] = SPI_DACD4_BIT, - [PCM_REAR_CHANNEL] = SPI_DACD0_BIT, - [PCM_CENTER_LFE_CHANNEL]= SPI_DACD2_BIT, - [PCM_UNKNOWN_CHANNEL] = SPI_DACD1_BIT, + SPI_DACD0_BIT, + SPI_DACD1_BIT, + SPI_DACD2_BIT, + 0, + SPI_DACD4_BIT, }; static void restore_spdif_bits(struct snd_ca0106 *chip, int idx) @@ -514,18 +516,34 @@ static void restore_spdif_bits(struct snd_ca0106 *chip, int idx) } } +static int snd_ca0106_channel_dac(struct snd_ca0106_details *details, + int channel_id) +{ + switch (channel_id) { + case PCM_FRONT_CHANNEL: return 4; + case PCM_REAR_CHANNEL: return 0; + case PCM_CENTER_LFE_CHANNEL: return 2; + case PCM_UNKNOWN_CHANNEL: return 1; + } + snd_printk(KERN_DEBUG "ca0106: unknown channel_id %d\n", channel_id); + return 0; +} + static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, int power) { if (chip->details->spi_dac) { - const int reg = spi_dacd_reg[channel_id]; + const int dac = snd_ca0106_channel_dac(chip->details, + channel_id); + const int reg = spi_dacd_reg[dac]; + const int bit = spi_dacd_bit[dac]; if (power) /* Power up */ - chip->spi_dac_reg[reg] &= ~spi_dacd_bit[channel_id]; + chip->spi_dac_reg[reg] &= ~bit; else /* Power down */ - chip->spi_dac_reg[reg] |= spi_dacd_bit[channel_id]; + chip->spi_dac_reg[reg] |= bit; return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); } return 0; -- cgit v1.2.3 From 6fef153afa8b25f81417488150e04db7c6b0b229 Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:32 +1100 Subject: ALSA: ca0106: Allow different sound cards to use different SPI channel mappings. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106.h | 5 +++-- sound/pci/ca0106/ca0106_main.c | 26 ++++++++++++++++---------- sound/pci/ca0106/ca0106_mixer.c | 4 ++-- 3 files changed, 21 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index 14b8d9a91aae..f19c11077255 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -670,8 +670,9 @@ struct snd_ca0106_details { gpio_type = 2 -> shared side-out/line-in. */ int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume controls, phone, mic, line-in and aux. */ - int spi_dac; /* spi_dac=1 adds the mute switch for each analog - output, front, rear, etc. */ + u16 spi_dac; /* spi_dac = 0 -> no spi interface for DACs + spi_dac = 0x + -> specifies DAC id for each channel pair. */ }; // definition of the chip-specific record diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index da910031edfa..d2d12c08f937 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -227,7 +227,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "Audigy SE [SB0570]", .gpio_type = 1, .i2c_adc = 1, - .spi_dac = 1 } , + .spi_dac = 0x4021 } , /* New Audigy LS. Has a different DAC. */ /* SB0570: * CTRL:CA0106-DAT @@ -238,7 +238,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "Audigy SE OEM [SB0570a]", .gpio_type = 1, .i2c_adc = 1, - .spi_dac = 1 } , + .spi_dac = 0x4021 } , /* Sound Blaster 5.1vx * Tested: Playback on front, rear, center/lfe speakers * Not-Tested: Capture @@ -247,7 +247,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "Sound Blaster 5.1vx [SB1070]", .gpio_type = 1, .i2c_adc = 0, - .spi_dac = 1 + .spi_dac = 0x0124 } , /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ /* SB0438 @@ -264,7 +264,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 1 } , + .spi_dac = 0x4021 } , /* Giga-byte GA-G1975X mobo * Novell bnc#395807 */ @@ -520,12 +520,18 @@ static int snd_ca0106_channel_dac(struct snd_ca0106_details *details, int channel_id) { switch (channel_id) { - case PCM_FRONT_CHANNEL: return 4; - case PCM_REAR_CHANNEL: return 0; - case PCM_CENTER_LFE_CHANNEL: return 2; - case PCM_UNKNOWN_CHANNEL: return 1; + case PCM_FRONT_CHANNEL: + return (details->spi_dac & 0xf000) >> (4 * 3); + case PCM_REAR_CHANNEL: + return (details->spi_dac & 0x0f00) >> (4 * 2); + case PCM_CENTER_LFE_CHANNEL: + return (details->spi_dac & 0x00f0) >> (4 * 1); + case PCM_UNKNOWN_CHANNEL: + return (details->spi_dac & 0x000f) >> (4 * 0); + default: + snd_printk(KERN_DEBUG "ca0106: unknown channel_id %d\n", + channel_id); } - snd_printk(KERN_DEBUG "ca0106: unknown channel_id %d\n", channel_id); return 0; } @@ -1582,7 +1588,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) /* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */ } - if (chip->details->spi_dac == 1) { + if (chip->details->spi_dac) { /* The SB0570 use SPI to control DAC. */ int size, n; diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 85fd315d9999..b522401b0318 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -832,7 +832,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) if (err < 0) return err; } - if (emu->details->spi_dac == 1) + if (emu->details->spi_dac) ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); /* Create virtual master controls */ @@ -845,7 +845,7 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) return err; add_slaves(card, vmaster, slave_vols); - if (emu->details->spi_dac == 1) { + if (emu->details->spi_dac) { vmaster = snd_ctl_make_virtual_master("Master Playback Switch", NULL); if (!vmaster) -- cgit v1.2.3 From 64e5310a249ba641ab6a00c6c1d61146d51b7984 Mon Sep 17 00:00:00 2001 From: Andy Owen Date: Sat, 23 Oct 2010 22:12:33 +1100 Subject: ALSA: ca0106: Use card specific dac id for mute controls. Signed-off-by: Andy Owen Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_mixer.c | 91 +++++++++++++++++++++++++++++++---------- 1 file changed, 69 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index b522401b0318..630aa4998189 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -676,28 +676,65 @@ static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] __devinitdata = I2C_VOLUME("Aux Capture Volume", 3), }; -#define SPI_SWITCH(xname,reg,bit) \ -{ \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ - .info = spi_mute_info, \ - .get = spi_mute_get, \ - .put = spi_mute_put, \ - .private_value = (reg<spi_dac & 0xf000) >> (4 * 3); + break; + case PCM_REAR_CHANNEL: + spi_switch.name = "Analog Rear Playback Switch"; + dac_id = (details->spi_dac & 0x0f00) >> (4 * 2); + break; + case PCM_CENTER_LFE_CHANNEL: + spi_switch.name = "Analog Center/LFE Playback Switch"; + dac_id = (details->spi_dac & 0x00f0) >> (4 * 1); + break; + case PCM_UNKNOWN_CHANNEL: + spi_switch.name = "Analog Side Playback Switch"; + dac_id = (details->spi_dac & 0x000f) >> (4 * 0); + break; + default: + /* Unused channel */ + spi_switch.name = NULL; + dac_id = 0; + } + reg = spi_dmute_reg[dac_id]; + bit = spi_dmute_bit[dac_id]; + + spi_switch.private_value = (reg << SPI_REG_SHIFT) | bit; + + return spi_switch; +} + static int __devinit remove_ctl(struct snd_card *card, const char *name) { struct snd_ctl_elem_id id; @@ -832,8 +869,18 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) if (err < 0) return err; } - if (emu->details->spi_dac) - ADD_CTLS(emu, snd_ca0106_volume_spi_dac_ctls); + if (emu->details->spi_dac) { + int i; + for (i = 0;; i++) { + struct snd_kcontrol_new ctl; + ctl = snd_ca0106_volume_spi_dac_ctl(emu->details, i); + if (!ctl.name) + break; + err = snd_ctl_add(card, snd_ctl_new1(&ctl, emu)); + if (err < 0) + return err; + } + } /* Create virtual master controls */ vmaster = snd_ctl_make_virtual_master("Master Playback Volume", -- cgit v1.2.3 From 97c44b2dbd0060e2e6bd56236eb638ab02ec7f30 Mon Sep 17 00:00:00 2001 From: Mandar Joshi Date: Sun, 24 Oct 2010 04:07:00 +0000 Subject: ALSA: usb - Creative USB X-Fi volume knob support Adds an entry for Creative USB X-Fi to the rc_config array in mixer_quirks.c to allow use of volume knob on the device. The action of the volume knob is received by lirc when its using the alsa_usb driver. Signed-off-by: Mandar Joshi Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e7df1e5e3f2e..7dae05d8783e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -60,6 +60,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; -- cgit v1.2.3 From 0e7adbe263f89ea2ef15b5af5e80a812b2a85025 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Oct 2010 10:37:11 +0200 Subject: ALSA: hda - Disable sticky PCM stream assignment for AD codecs The sticky PCM stream assignment introduced in 2.6.36 kernel seems causing problems on AD codecs. At some time later, the streaming no longer works by unknown reason. A simple workaround is to disable sticky-assignment for these codecs. Tested-by: Vasily Khoruzhick Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_analog.c | 7 +++++++ 3 files changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 13c1e7703c49..644e3f14f8ca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1284,6 +1284,9 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, if (!nid) return; + if (codec->no_sticky_stream) + do_now = 1; + snd_printdd("hda_codec_cleanup_stream: NID=0x%x\n", nid); p = get_hda_cvt_setup(codec, nid); if (p) { diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index ebf8eb02e3c2..fdf8d44f8b6b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -850,6 +850,7 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 507523d5ed42..f7ff3f7ccb8e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1276,6 +1276,7 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.no_share_stream = 1; codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -1463,6 +1464,7 @@ static int patch_ad1983(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -1917,6 +1919,7 @@ static int patch_ad1981(struct hda_codec *codec) } codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -3236,6 +3239,7 @@ static int patch_ad1988(struct hda_codec *codec) spec->vmaster_nid = 0x04; codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -3450,6 +3454,7 @@ static int patch_ad1884(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -4423,6 +4428,7 @@ static int patch_ad1884a(struct hda_codec *codec) } codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } @@ -4762,6 +4768,7 @@ static int patch_ad1882(struct hda_codec *codec) } codec->no_trigger_sense = 1; + codec->no_sticky_stream = 1; return 0; } -- cgit v1.2.3 From 335e3b8687fa6832bd6a033f2c705786e4bfb92c Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Fri, 22 Oct 2010 18:38:31 -0500 Subject: ALSA: hda - Change BTL amp level on some HP notebooks Some HP laptops have lower amplifier levels for speakers in comparison with headphone outputs. This patch changes the BTL amp level for these machines to balance both the speaker and headphone output levels. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 78 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 82ebeb9544fe..93fa59cc60ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5326,6 +5326,82 @@ again: return 0; } +static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) +{ + if (codec->vendor_id != 0x111d7605 && + codec->vendor_id != 0x111d76d1) + return 0; + + switch (codec->subsystem_id) { + case 0x103c1618: + case 0x103c1619: + case 0x103c161a: + case 0x103c161b: + case 0x103c161c: + case 0x103c161d: + case 0x103c161e: + case 0x103c161f: + case 0x103c1620: + case 0x103c1621: + case 0x103c1622: + case 0x103c1623: + + case 0x103c162a: + case 0x103c162b: + + case 0x103c1630: + case 0x103c1631: + + case 0x103c1633: + + case 0x103c1635: + + case 0x103c164f: + + case 0x103c1676: + case 0x103c1677: + case 0x103c1678: + case 0x103c1679: + case 0x103c167a: + case 0x103c167b: + case 0x103c167c: + case 0x103c167d: + case 0x103c167e: + case 0x103c167f: + case 0x103c1680: + case 0x103c1681: + case 0x103c1682: + case 0x103c1683: + case 0x103c1684: + case 0x103c1685: + case 0x103c1686: + case 0x103c1687: + case 0x103c1688: + case 0x103c1689: + case 0x103c168a: + case 0x103c168b: + case 0x103c168c: + case 0x103c168d: + case 0x103c168e: + case 0x103c168f: + case 0x103c1690: + case 0x103c1691: + case 0x103c1692: + + case 0x103c3587: + case 0x103c3588: + case 0x103c3589: + case 0x103c358a: + + case 0x103c3667: + case 0x103c3668: + /* set BTL amp level to 13.43dB for louder speaker output */ + return snd_hda_codec_write_cache(codec, codec->afg, 0, + 0x7F4, 0x14); + } + return 0; +} + static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5452,6 +5528,8 @@ again: AC_VERB_SET_CONNECT_SEL, num_dacs); } + stac92hd83xxx_set_system_btl_amp(codec); + codec->proc_widget_hook = stac92hd_proc_hook; return 0; -- cgit v1.2.3 From 3342b9680fbd6535f65148f5e95e581bc75be9cc Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Tue, 26 Oct 2010 12:25:38 +0200 Subject: sound/oss/sb_ess.c: delete double assignment Delete successive assignments to the same location. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression i; @@ *i = ...; i = ...; // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/sb_ess.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 51a3d381a59e..9890cf2066ff 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1721,7 +1721,6 @@ printk (KERN_INFO "FKS: es_rec_set_recmask mask = %x\n", mask); left = value & 0x000000ff; right = (value & 0x0000ff00) >> 8; } else { /* Turn it off (3) */ - left = 0; left = 0; right = 0; } -- cgit v1.2.3 From e94be5f3628565d0968d668341d9dd468700516d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 26 Oct 2010 20:07:43 +0100 Subject: ASoC: fsl - fix build error in pcm030-audio-fabric.c Fix build error:- sound/soc/fsl/pcm030-audio-fabric.c:27:33: fatal error: sound/soc-of-simple.h: No such file or directory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/soc/fsl/pcm030-audio-fabric.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index fe15bb26e484..25f27ec1dd6e 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -24,7 +24,6 @@ #include #include #include -#include #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" @@ -49,7 +48,7 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", .platform_name = "mpc5200-pcm-audio", - ..codec_name = "wm9712-codec", + .codec_name = "wm9712-codec", }, }; -- cgit v1.2.3 From 836f5394061830a9d1ece4aafc437c098774522f Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:34 -0400 Subject: ASoC: sound/ad73311: add missing __devexit marker This fixes the following warning: sound/soc/codecs/ad73311.c:50:12: warning: 'ad73311_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/ad73311.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index c53955fe17b6..de799cd1ba72 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -47,7 +47,7 @@ static int ad73311_probe(struct platform_device *pdev) &soc_codec_dev_ad73311, &ad73311_dai, 1); } -static int ad73311_remove(struct platform_device *pdev) +static int __devexit ad73311_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; -- cgit v1.2.3 From f3607aef0d3370ae9edbfecfc7182233397cb0aa Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:35 -0400 Subject: ASoC: sound/max98088: add missing __devexit marker This fixes the following warning: sound/soc/codecs/max98088.c:2054:12: warning: 'max98088_i2c_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index e7a40d16df90..bc22ee93a75d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2051,7 +2051,7 @@ static int max98088_i2c_probe(struct i2c_client *i2c, return ret; } -static int max98088_i2c_remove(struct i2c_client *client) +static int __devexit max98088_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); -- cgit v1.2.3 From 0d040df9984c8fcb6a777a8f6d5dc513eaefd2de Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:36 -0400 Subject: ASoC: sound/wm9090: add missing __devexit marker This fixes the following warning: sound/soc/codecs/wm9090.c:668:12: warning: 'wm9090_i2c_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 7a1825418ee4..99c046ba46bb 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -665,7 +665,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return ret; } -static int wm9090_i2c_remove(struct i2c_client *i2c) +static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); -- cgit v1.2.3 From 89e1e66d6be8a520cdcd26043cda2cc870a34015 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 26 Oct 2010 17:14:41 +0200 Subject: ALSA: usb-audio: automatically detect feedback format There are two USB Audio Class specifications (v1 and v2), but neither of them clearly defines the feedback format for high-speed UAC v1 devices. Add to this whatever the Creative and M-Audio firmware writers have been smoking, and it becomes impossible to predict the exact feedback format used by a particular device. Therefore, automatically detect the feedback format by looking at the magnitude of the first received feedback value. Also, this allows us to get rid of some special cases for E-Mu devices. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 + sound/usb/pcm.c | 2 + sound/usb/proc.c | 5 ++ sound/usb/urb.c | 170 +++++++++++++++++++------------------------------------ 4 files changed, 67 insertions(+), 112 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index 1febf2f23754..ae4251d5abf7 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -62,12 +62,14 @@ struct snd_usb_substream { unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ unsigned int freqmax; /* maximum sampling rate, used for buffer management */ unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f49756c1b837..cff3a3c465d7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -237,6 +237,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; + subs->syncmaxsize = 0; subs->fill_max = 0; /* we need a sync pipe in async OUT or adaptive IN mode */ @@ -283,6 +284,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; else subs->syncinterval = 3; + subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); } /* always fill max packet size */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 3c650ab3c91d..961c9a250686 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -132,6 +132,11 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn ? get_full_speed_hz(subs->freqm) : get_high_speed_hz(subs->freqm), subs->freqm >> 16, subs->freqm & 0xffff); + if (subs->freqshift != INT_MIN) + snd_iprintf(buffer, " Feedback Format = %d.%d\n", + (subs->syncmaxsize > 3 ? 32 : 24) + - (16 - subs->freqshift), + 16 - subs->freqshift); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/urb.c b/sound/usb/urb.c index 8deeaad10f10..e184349aee83 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -225,6 +225,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->freqn = get_usb_high_speed_rate(rate); subs->freqm = subs->freqn; + subs->freqshift = INT_MIN; /* calculate max. frequency */ if (subs->maxpacksize) { /* whatever fits into a max. size packet */ @@ -513,11 +514,10 @@ static int retire_paused_capture_urb(struct snd_usb_substream *subs, /* - * prepare urb for full speed playback sync pipe + * prepare urb for playback sync pipe * * set up the offset and length to receive the current frequency. */ - static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) @@ -525,103 +525,78 @@ static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_urb_ctx *ctx = urb->context; urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); urb->iso_frame_desc[0].offset = 0; return 0; } /* - * prepare urb for high speed playback sync pipe + * process after playback sync complete * - * set up the offset and length to receive the current frequency. - */ - -static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after full speed playback sync complete - * - * retrieve the current 10.14 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. */ static int retire_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { unsigned int f; + int shift; unsigned long flags; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 3) { - f = combine_triple((u8*)urb->transfer_buffer) << 2; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return 0; -/* - * process after high speed playback sync complete - * - * retrieve the current 12.13 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). - */ -static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + if (f == 0) + return 0; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); + if (unlikely(subs->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < subs->freqn - subs->freqn / 4) { + f <<= 1; + shift++; + } + while (f > subs->freqn + subs->freqn / 2) { + f >>= 1; + shift--; } + subs->freqshift = shift; } + else if (subs->freqshift >= 0) + f <<= subs->freqshift; + else + f >>= -subs->freqshift; - return 0; -} - -/* - * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete - * - * These devices return the number of samples per packet instead of the number - * of samples per microframe. - */ -static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - f >>= subs->datainterval; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } + if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + subs->freqshift = INT_MIN; } return 0; @@ -878,21 +853,6 @@ static struct snd_urb_ops audio_urb_ops[2] = { }, }; -static struct snd_urb_ops audio_urb_ops_high_speed[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb_hs, - .retire_sync = retire_playback_sync_urb_hs, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb_hs, - .retire_sync = retire_capture_sync_urb, - }, -}; - /* * initialize the substream instance. */ @@ -909,23 +869,9 @@ void snd_usb_init_substream(struct snd_usb_stream *as, subs->direction = stream; subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { - subs->ops = audio_urb_ops[stream]; - } else { - subs->ops = audio_urb_ops_high_speed[stream]; - switch (as->chip->usb_id) { - case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ - case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ - case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ - subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; - break; - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ - subs->ops.prepare_sync = prepare_playback_sync_urb; - subs->ops.retire_sync = retire_playback_sync_urb; - break; - } - } + subs->ops = audio_urb_ops[stream]; + if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) + subs->ops.prepare_sync = prepare_capture_sync_urb_hs; snd_usb_set_pcm_ops(as->pcm, stream); -- cgit v1.2.3 From c593b520cf70b0672680da04cc1e8c5f93bd739d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Oct 2010 20:11:17 -0700 Subject: ASoC: Check return value of struct_strtoul() in pmdown_time_set() strict_strtoul() has just been made must check so do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 70d9a7394b2b..805343fe903b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); + int ret; - strict_strtol(buf, 10, &rtd->pmdown_time); + ret = strict_strtol(buf, 10, &rtd->pmdown_time); + if (ret) + return ret; return count; } -- cgit v1.2.3 From 6d212d8e86fb4221bd91b9266b7567ee2b83bd01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Oct 2010 15:41:17 -0700 Subject: ASoC: Remove volatility from WM8900 POWER1 register Not all bits can be read back from POWER1 so avoid corruption when using a read/modify/write cycle by marking it non-volatile - the only thing we read back from it is the chip revision which has diagnostic value only. We can re-add later but that's a more invasive change than is suitable for a bugfix. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8900.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b4f11724a63f..aca4b1ea10bb 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - case WM8900_REG_POWER1: return 1; default: return 0; @@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) return -ENODEV; } - /* Read back from the chip */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - reg = (reg >> 12) & 0xf; - dev_info(codec->dev, "WM8900 revision %d\n", reg); - wm8900_reset(codec); /* Turn the chip on */ -- cgit v1.2.3 From 703dde6219346bc3b7d41d4fa2c36846d728e52c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:44 +0300 Subject: ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740 Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b9027..02a9751bf149 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,7 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_DA7210 if I2C - select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- cgit v1.2.3 From 76a6106f124e375df0ea6ba6bcf204b8caff786a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:45 +0300 Subject: ASoC: Include cx20442 to SND_SOC_ALL_CODECS Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 02a9751bf149..3b5690d28b8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C -- cgit v1.2.3