diff options
Diffstat (limited to 'sound')
128 files changed, 4323 insertions, 1441 deletions
diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6d12ca9bcb80..9749f9e8b45c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT Say Y here to use the HR-timer backend as the default sequencer timer. -config SND_RTCTIMER - tristate "RTC Timer support" - depends on RTC - select SND_TIMER - help - Say Y here to enable RTC timer support for ALSA. ALSA uses - the RTC timer as a precise timing source and maps the RTC - timer to ALSA's timer interface. The ALSA sequencer code also - can use this timing source. - - To compile this driver as a module, choose M here: the module - will be called snd-rtctimer. - - Note that this option is exclusive with the new RTC drivers - (CONFIG_RTC_CLASS) since this requires the old API. - -config SND_SEQ_RTCTIMER_DEFAULT - bool "Use RTC as default sequencer timer" - depends on SND_RTCTIMER && SND_SEQUENCER - depends on !SND_SEQ_HRTIMER_DEFAULT - default y - help - Say Y here to use the RTC timer as the default sequencer - timer. This is strongly recommended because it ensures - precise MIDI timing even when the system timer runs at less - than 1000 Hz. - - If in doubt, say Y. - config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" help diff --git a/sound/core/Makefile b/sound/core/Makefile index 48ab4b8f8279..e85d9dd12c2d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o -obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a9933c07a6bf..9b3334be9df2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); /* write is allowed when stream is running or has been steup */ - if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_PREPARED && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_RUNNING: + break; + default: mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -391,14 +394,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) int retval = 0; if (snd_BUG_ON(!data)) - return -EFAULT; + return POLLERR; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; mutex_lock(&stream->device->lock); if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { - retval = -EBADFD; + retval = snd_compr_get_poll(stream) | POLLERR; goto out; } poll_wait(f, &stream->runtime->sleep, wait); @@ -421,10 +423,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) retval = snd_compr_get_poll(stream); break; default: - if (stream->direction == SND_COMPRESS_PLAYBACK) - retval = POLLOUT | POLLWRNORM | POLLERR; - else - retval = POLLIN | POLLRDNORM | POLLERR; + retval = snd_compr_get_poll(stream) | POLLERR; break; } out: @@ -802,9 +801,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!data)) return -EFAULT; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; + mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 656d9a9032dc..e2f27022b363 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -38,37 +38,53 @@ static unsigned int resolution; struct snd_hrtimer { struct snd_timer *timer; struct hrtimer hrt; - atomic_t running; + bool in_callback; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; - unsigned long oruns; - - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - - oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks * oruns); + ktime_t delta; + unsigned long ticks; + enum hrtimer_restart ret = HRTIMER_NORESTART; + + spin_lock(&t->lock); + if (!t->running) + goto out; /* fast path */ + stime->in_callback = true; + ticks = t->sticks; + spin_unlock(&t->lock); + + /* calculate the drift */ + delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt)); + if (delta.tv64 > 0) + ticks += ktime_divns(delta, ticks * resolution); + + snd_timer_interrupt(stime->timer, ticks); + + spin_lock(&t->lock); + if (t->running) { + hrtimer_add_expires_ns(hrt, t->sticks * resolution); + ret = HRTIMER_RESTART; + } - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - return HRTIMER_RESTART; + stime->in_callback = false; + out: + spin_unlock(&t->lock); + return ret; } static int snd_hrtimer_open(struct snd_timer *t) { struct snd_hrtimer *stime; - stime = kmalloc(sizeof(*stime), GFP_KERNEL); + stime = kzalloc(sizeof(*stime), GFP_KERNEL); if (!stime) return -ENOMEM; hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; stime->hrt.function = snd_hrtimer_callback; - atomic_set(&stime->running, 0); t->private_data = stime; return 0; } @@ -78,6 +94,11 @@ static int snd_hrtimer_close(struct snd_timer *t) struct snd_hrtimer *stime = t->private_data; if (stime) { + spin_lock_irq(&t->lock); + t->running = 0; /* just to be sure */ + stime->in_callback = 1; /* skip start/stop */ + spin_unlock_irq(&t->lock); + hrtimer_cancel(&stime->hrt); kfree(stime); t->private_data = NULL; @@ -89,18 +110,19 @@ static int snd_hrtimer_start(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); - hrtimer_try_to_cancel(&stime->hrt); + if (stime->in_callback) + return 0; hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); - atomic_set(&stime->running, 1); return 0; } static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); + + if (stime->in_callback) + return 0; hrtimer_try_to_cancel(&stime->hrt); return 0; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3a9b66c6e09c..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_timer_interrupt(substream->timer, 1); #endif _end: - snd_pcm_stream_unlock_irqrestore(substream, flags); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9106d8e2300e..c61fd50f771f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLOUT | POLLWRNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); @@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLIN | POLLRDNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d11a12f..000000000000 --- a/sound/core/rtctimer.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * RTC based high-frequency timer - * - * Copyright (C) 2000 Takashi Iwai - * based on rtctimer.c by Steve Ratcliffe - * - * 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 <linux/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/log2.h> -#include <sound/core.h> -#include <sound/timer.h> - -#if IS_ENABLED(CONFIG_RTC) - -#include <linux/mc146818rtc.h> - -#define RTC_FREQ 1024 /* default frequency */ -#define NANO_SEC 1000000000L /* 10^9 in sec */ - -/* - * prototypes - */ -static int rtctimer_open(struct snd_timer *t); -static int rtctimer_close(struct snd_timer *t); -static int rtctimer_start(struct snd_timer *t); -static int rtctimer_stop(struct snd_timer *t); - - -/* - * The hardware dependent description for this timer. - */ -static struct snd_timer_hardware rtc_hw = { - .flags = SNDRV_TIMER_HW_AUTO | - SNDRV_TIMER_HW_FIRST | - SNDRV_TIMER_HW_TASKLET, - .ticks = 100000000L, /* FIXME: XXX */ - .open = rtctimer_open, - .close = rtctimer_close, - .start = rtctimer_start, - .stop = rtctimer_stop, -}; - -static int rtctimer_freq = RTC_FREQ; /* frequency */ -static struct snd_timer *rtctimer; -static struct tasklet_struct rtc_tasklet; -static rtc_task_t rtc_task; - - -static int -rtctimer_open(struct snd_timer *t) -{ - int err; - - err = rtc_register(&rtc_task); - if (err < 0) - return err; - t->private_data = &rtc_task; - return 0; -} - -static int -rtctimer_close(struct snd_timer *t) -{ - rtc_task_t *rtc = t->private_data; - if (rtc) { - rtc_unregister(rtc); - tasklet_kill(&rtc_tasklet); - t->private_data = NULL; - } - return 0; -} - -static int -rtctimer_start(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq); - rtc_control(rtc, RTC_PIE_ON, 0); - return 0; -} - -static int -rtctimer_stop(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_PIE_OFF, 0); - return 0; -} - -static void rtctimer_tasklet(unsigned long data) -{ - snd_timer_interrupt((struct snd_timer *)data, 1); -} - -/* - * interrupt - */ -static void rtctimer_interrupt(void *private_data) -{ - tasklet_schedule(private_data); -} - - -/* - * ENTRY functions - */ -static int __init rtctimer_init(void) -{ - int err; - struct snd_timer *timer; - - if (rtctimer_freq < 2 || rtctimer_freq > 8192 || - !is_power_of_2(rtctimer_freq)) { - pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq); - return -EINVAL; - } - - /* Create a new timer and set up the fields */ - err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer); - if (err < 0) - return err; - - timer->module = THIS_MODULE; - strcpy(timer->name, "RTC timer"); - timer->hw = rtc_hw; - timer->hw.resolution = NANO_SEC / rtctimer_freq; - - tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer); - - /* set up RTC callback */ - rtc_task.func = rtctimer_interrupt; - rtc_task.private_data = &rtc_tasklet; - - err = snd_timer_global_register(timer); - if (err < 0) { - snd_timer_global_free(timer); - return err; - } - rtctimer = timer; /* remember this */ - - return 0; -} - -static void __exit rtctimer_exit(void) -{ - if (rtctimer) { - snd_timer_global_free(rtctimer); - rtctimer = NULL; - } -} - - -/* - * exported stuff - */ -module_init(rtctimer_init) -module_exit(rtctimer_exit) - -module_param(rtctimer_freq, int, 0444); -MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz"); - -MODULE_LICENSE("GPL"); - -MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); - -#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7e0aabb808a6..639544b4fb04 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; int seq_default_timer_device = #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT SNDRV_TIMER_GLOBAL_HRTIMER -#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) - SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6469bedda2f3..e722022d325d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -37,8 +37,6 @@ #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif IS_ENABLED(CONFIG_SND_RTCTIMER) -#define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 #endif @@ -1225,6 +1223,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, tu->tstamp = *tstamp; if ((tu->filter & (1 << event)) == 0 || !tu->tread) return; + memset(&r1, 0, sizeof(r1)); r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; @@ -1267,6 +1266,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { + memset(&r1, 0, sizeof(r1)); r1.event = SNDRV_TIMER_EVENT_RESOLUTION; r1.tstamp = tstamp; r1.val = resolution; @@ -1739,6 +1739,7 @@ static int snd_timer_user_params(struct file *file, if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { if (tu->tread) { struct snd_timer_tread tread; + memset(&tread, 0, sizeof(tread)); tread.event = SNDRV_TIMER_EVENT_EARLY; tread.tstamp.tv_sec = 0; tread.tstamp.tv_nsec = 0; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index c0f8f613f1f1..172dacd925f5 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -420,6 +420,7 @@ static int dummy_hrtimer_stop(struct snd_pcm_substream *substream) static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) { + hrtimer_cancel(&dpcm->timer); tasklet_kill(&dpcm->tasklet); } diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 2a779c2f63ab..ab894ed1ff67 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -134,6 +134,7 @@ config SND_FIREWIRE_TASCAM Say Y here to include support for TASCAM. * FW-1884 * FW-1082 + * FW-1804 To compile this driver as a module, choose M here: the module will be called snd-firewire-tascam. diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 003c09029786..0ee1fb115d88 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 000000000000..9c04faf206b2 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,110 @@ +/* + * amdtp-stream-trace.h - tracepoint definitions to dump a part of packet data + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include <linux/tracepoint.h> + +TRACE_EVENT(in_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(unsigned int, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->node_id; + __entry->dest = fw_parent_device(s->unit)->card->node_id; + __entry->cip_header0 = cip_header[0]; + __entry->cip_header1 = cip_header[1]; + __entry->payload_quadlets = payload_quadlets; + __entry->packet_index = s->packet_index; + __entry->irq = !!in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(unsigned int, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->card->node_id; + __entry->dest = fw_parent_device(s->unit)->node_id; + __entry->cip_header0 = be32_to_cpu(cip_header[0]); + __entry->cip_header1 = be32_to_cpu(cip_header[1]); + __entry->payload_quadlets = payload_length / 4; + __entry->packet_index = s->packet_index; + __entry->irq = !!in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include <trace/define_trace.h> diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed2902609a4c..00060c4a9deb 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -19,6 +19,10 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* Always support Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ @@ -87,7 +91,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, init_waitqueue_head(&s->callback_wait); s->callbacked = false; - s->sync_slave = NULL; s->fmt = fmt; s->process_data_blocks = process_data_blocks; @@ -102,6 +105,10 @@ EXPORT_SYMBOL(amdtp_stream_init); */ void amdtp_stream_destroy(struct amdtp_stream *s) { + /* Not initialized. */ + if (s->protocol == NULL) + return; + WARN_ON(amdtp_stream_running(s)); kfree(s->protocol); mutex_destroy(&s->mutex); @@ -244,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -349,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -363,9 +368,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -376,8 +380,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -392,27 +398,30 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, - unsigned int syt) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; + unsigned int syt; + unsigned int data_blocks; unsigned int payload_length; unsigned int pcm_frames; struct snd_pcm_substream *pcm; buffer = s->buffer.packets[s->packet_index].buffer; + syt = calculate_syt(s, cycle); + data_blocks = calculate_data_blocks(s, syt); pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | @@ -424,9 +433,11 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, (syt & CIP_SYT_MASK)); s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; - payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - if (queue_out_packet(s, payload_length, false) < 0) + + trace_out_packet(s, cycle, buffer, payload_length, index); + + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -438,19 +449,24 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks, unsigned int syt) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { + __be32 *buffer; u32 cip_header[2]; - unsigned int fmt, fdf; + unsigned int fmt, fdf, syt; unsigned int data_block_quadlets, data_block_counter, dbc_interval; + unsigned int data_blocks; struct snd_pcm_substream *pcm; unsigned int pcm_frames; bool lost; + buffer = s->buffer.packets[s->packet_index].buffer; cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -460,7 +476,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -471,7 +487,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Detect unexpected protocol: %08x %08x\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -480,7 +496,7 @@ static int handle_in_packet(struct amdtp_stream *s, fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT; if (payload_quadlets < 3 || (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) { - *data_blocks = 0; + data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -494,12 +510,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - *data_blocks = (payload_quadlets - 2) / data_block_quadlets; + data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -510,10 +526,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) + if (data_blocks > 0 && s->tx_dbc_interval > 0) dbc_interval = s->tx_dbc_interval; else - dbc_interval = *data_blocks; + dbc_interval = data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -526,13 +542,14 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); + syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; + pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + *data_blocks) & 0xff; + (data_block_counter + data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; @@ -544,29 +561,50 @@ end: return 0; } -static void out_stream_callback(struct fw_iso_context *context, u32 cycle, +/* + * In CYCLE_TIMER register of IEEE 1394, 7 bits are used to represent second. On + * the other hand, in DMA descriptors of 1394 OHCI, 3 bits are used to represent + * it. Thus, via Linux firewire subsystem, we can get the 3 bits for second. + */ +static inline u32 compute_cycle_count(u32 tstamp) +{ + return (((tstamp >> 13) & 0x07) * 8000) + (tstamp & 0x1fff); +} + +static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) +{ + cycle += addend; + if (cycle >= 8 * CYCLES_PER_SECOND) + cycle -= 8 * CYCLES_PER_SECOND; + return cycle; +} + +static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) +{ + if (cycle < subtrahend) + cycle += 8 * CYCLES_PER_SECOND; + return cycle - subtrahend; +} + +static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int i, syt, packets = header_length / 4; - unsigned int data_blocks; + unsigned int i, packets = header_length / 4; + u32 cycle; if (s->packet_index < 0) return; - /* - * Compute the cycle of the last queued packet. - * (We need only the four lowest bits for the SYT, so we can ignore - * that bits 0-11 must wrap around at 3072.) - */ - cycle += QUEUE_LENGTH - packets; + cycle = compute_cycle_count(tstamp); - for (i = 0; i < packets; ++i) { - syt = calculate_syt(s, ++cycle); - data_blocks = calculate_data_blocks(s, syt); + /* Align to actual cycle count for the last packet. */ + cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); - if (handle_out_packet(s, data_blocks, syt) < 0) { + for (i = 0; i < packets; ++i) { + cycle = increment_cycle_count(cycle, 1); + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -576,15 +614,15 @@ static void out_stream_callback(struct fw_iso_context *context, u32 cycle, fw_iso_context_queue_flush(s->context); } -static void in_stream_callback(struct fw_iso_context *context, u32 cycle, +static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int p, syt, packets; + unsigned int i, packets; unsigned int payload_quadlets, max_payload_quadlets; - unsigned int data_blocks; - __be32 *buffer, *headers = header; + __be32 *headers = header; + u32 cycle; if (s->packet_index < 0) return; @@ -592,70 +630,44 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, /* The number of packets in buffer */ packets = header_length / IN_PACKET_HEADER_SIZE; + cycle = compute_cycle_count(tstamp); + + /* Align to actual cycle count for the last packet. */ + cycle = decrement_cycle_count(cycle, packets); + /* For buffer-over-run prevention. */ max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4; - for (p = 0; p < packets; p++) { - buffer = s->buffer.packets[s->packet_index].buffer; + for (i = 0; i < packets; i++) { + cycle = increment_cycle_count(cycle, 1); /* The number of quadlets in this packet */ payload_quadlets = - (be32_to_cpu(headers[p]) >> ISO_DATA_LENGTH_SHIFT) / 4; + (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT) / 4; if (payload_quadlets > max_payload_quadlets) { dev_err(&s->unit->device, "Detect jumbo payload: %02x %02x\n", payload_quadlets, max_payload_quadlets); - s->packet_index = -1; break; } - syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; - if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, syt) < 0) { - s->packet_index = -1; + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; - } - - /* Process sync slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) { - if (handle_out_packet(s->sync_slave, - data_blocks, syt) < 0) { - s->packet_index = -1; - break; - } - } } - /* Queueing error or detecting discontinuity */ - if (s->packet_index < 0) { + /* Queueing error or detecting invalid payload. */ + if (i < packets) { + s->packet_index = -1; amdtp_stream_pcm_abort(s); - - /* Abort sync slave. */ - if (s->sync_slave) { - s->sync_slave->packet_index = -1; - amdtp_stream_pcm_abort(s->sync_slave); - } return; } - /* when sync to device, flush the packets for slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) - fw_iso_context_queue_flush(s->sync_slave->context); - fw_iso_context_queue_flush(s->context); } -/* processing is done by master callback */ -static void slave_stream_callback(struct fw_iso_context *context, u32 cycle, - size_t header_length, void *header, - void *private_data) -{ - return; -} - /* this is executed one time */ static void amdtp_stream_first_callback(struct fw_iso_context *context, - u32 cycle, size_t header_length, + u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; @@ -669,12 +681,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, if (s->direction == AMDTP_IN_STREAM) context->callback.sc = in_stream_callback; - else if (s->flags & CIP_SYNC_TO_DEVICE) - context->callback.sc = slave_stream_callback; else context->callback.sc = out_stream_callback; - context->callback.sc(context, cycle, header_length, header, s); + context->callback.sc(context, tstamp, header_length, header, s); } /** @@ -713,8 +723,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) goto err_unlock; } - if (s->direction == AMDTP_IN_STREAM && - s->flags & CIP_SKIP_INIT_DBC_CHECK) + if (s->direction == AMDTP_IN_STREAM) s->data_block_counter = UINT_MAX; else s->data_block_counter = 0; @@ -755,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); @@ -794,11 +803,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets will be already + * queued in the buffer. These packets should be handled immediately + * to keep better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704a3665..c1bc7fad056e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -17,8 +17,6 @@ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or * SYT_INTERVAL samples, with these two types alternating so that * the overall sample rate comes out right. - * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is - * generated by in packets. Defaultly this driver generates timestamp. * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0. * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet * corresponds to the end of event in the packet. Out of IEC 61883. @@ -26,8 +24,6 @@ * The value of data_block_quadlets is used instead of reported value. * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. - * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first - * packet is not continuous from an initial value. * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty * packet is wrong but the others are correct. * @CIP_JUMBO_PAYLOAD: Only for in-stream. The number of data blocks in an @@ -37,14 +33,12 @@ enum cip_flags { CIP_NONBLOCKING = 0x00, CIP_BLOCKING = 0x01, - CIP_SYNC_TO_DEVICE = 0x02, - CIP_EMPTY_WITH_TAG0 = 0x04, - CIP_DBC_IS_END_EVENT = 0x08, - CIP_WRONG_DBS = 0x10, - CIP_SKIP_DBC_ZERO_CHECK = 0x20, - CIP_SKIP_INIT_DBC_CHECK = 0x40, - CIP_EMPTY_HAS_WRONG_DBC = 0x80, - CIP_JUMBO_PAYLOAD = 0x100, + CIP_EMPTY_WITH_TAG0 = 0x02, + CIP_DBC_IS_END_EVENT = 0x04, + CIP_WRONG_DBS = 0x08, + CIP_SKIP_DBC_ZERO_CHECK = 0x10, + CIP_EMPTY_HAS_WRONG_DBC = 0x20, + CIP_JUMBO_PAYLOAD = 0x40, }; /** @@ -132,12 +126,10 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; wait_queue_head_t callback_wait; - struct amdtp_stream *sync_slave; /* For backends to process data blocks. */ void *protocol; @@ -223,23 +215,6 @@ static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) return sfc & 1; } -static inline void amdtp_stream_set_sync(enum cip_flags sync_mode, - struct amdtp_stream *master, - struct amdtp_stream *slave) -{ - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master->flags |= CIP_SYNC_TO_DEVICE; - slave->flags |= CIP_SYNC_TO_DEVICE; - master->sync_slave = slave; - } else { - master->flags &= ~CIP_SYNC_TO_DEVICE; - slave->flags &= ~CIP_SYNC_TO_DEVICE; - master->sync_slave = NULL; - } - - slave->sync_slave = NULL; -} - /** * amdtp_stream_wait_callback - sleep till callbacked or timeout * @s: the AMDTP stream diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 3e4e0756e3fe..f7e2cbd2a313 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -67,7 +67,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define MODEL_MAUDIO_PROJECTMIX 0x00010091 static int -name_device(struct snd_bebob *bebob, unsigned int vendor_id) +name_device(struct snd_bebob *bebob) { struct fw_device *fw_dev = fw_parent_device(bebob->unit); char vendor[24] = {0}; @@ -126,6 +126,17 @@ end: return err; } +static void bebob_free(struct snd_bebob *bebob) +{ + snd_bebob_stream_destroy_duplex(bebob); + fw_unit_put(bebob->unit); + + kfree(bebob->maudio_special_quirk); + + mutex_destroy(&bebob->mutex); + kfree(bebob); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -137,18 +148,11 @@ bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; - snd_bebob_stream_destroy_duplex(bebob); - fw_unit_put(bebob->unit); - - kfree(bebob->maudio_special_quirk); - - if (bebob->card_index >= 0) { - mutex_lock(&devices_mutex); - clear_bit(bebob->card_index, devices_used); - mutex_unlock(&devices_mutex); - } + mutex_lock(&devices_mutex); + clear_bit(bebob->card_index, devices_used); + mutex_unlock(&devices_mutex); - mutex_destroy(&bebob->mutex); + bebob_free(card->private_data); } static const struct snd_bebob_spec * @@ -176,16 +180,17 @@ check_audiophile_booted(struct fw_unit *unit) return strncmp(name, "FW Audiophile Bootloader", 15) != 0; } -static int -bebob_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_bebob *bebob; - const struct snd_bebob_spec *spec; + struct snd_bebob *bebob = + container_of(work, struct snd_bebob, dwork.work); unsigned int card_index; int err; + if (bebob->registered) + return; + mutex_lock(&devices_mutex); for (card_index = 0; card_index < SNDRV_CARDS; card_index++) { @@ -193,64 +198,39 @@ bebob_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - if ((entry->vendor_id == VEN_FOCUSRITE) && - (entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH)) - spec = get_saffire_spec(unit); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH) && - !check_audiophile_booted(unit)) - spec = NULL; - else - spec = (const struct snd_bebob_spec *)entry->driver_data; - - if (spec == NULL) { - if ((entry->vendor_id == VEN_MAUDIO1) || - (entry->vendor_id == VEN_MAUDIO2)) - err = snd_bebob_maudio_load_firmware(unit); - else - err = -ENOSYS; - goto end; + err = snd_card_new(&bebob->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &bebob->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_bebob), &card); + err = name_device(bebob); if (err < 0) - goto end; - bebob = card->private_data; - bebob->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = bebob_card_free; - - bebob->card = card; - bebob->unit = fw_unit_get(unit); - bebob->spec = spec; - mutex_init(&bebob->mutex); - spin_lock_init(&bebob->lock); - init_waitqueue_head(&bebob->hwdep_wait); + goto error; - err = name_device(bebob, entry->vendor_id); + if (bebob->spec == &maudio_special_spec) { + if (bebob->entry->model_id == MODEL_MAUDIO_FW1814) + err = snd_bebob_maudio_special_discover(bebob, true); + else + err = snd_bebob_maudio_special_discover(bebob, false); + } else { + err = snd_bebob_stream_discover(bebob); + } if (err < 0) goto error; - if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_FW1814)) - err = snd_bebob_maudio_special_discover(bebob, true); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_PROJECTMIX)) - err = snd_bebob_maudio_special_discover(bebob, false); - else - err = snd_bebob_stream_discover(bebob); + err = snd_bebob_stream_init_duplex(bebob); if (err < 0) goto error; snd_bebob_proc_init(bebob); - if ((bebob->midi_input_ports > 0) || - (bebob->midi_output_ports > 0)) { + if (bebob->midi_input_ports > 0 || bebob->midi_output_ports > 0) { err = snd_bebob_create_midi_devices(bebob); if (err < 0) goto error; @@ -264,16 +244,75 @@ bebob_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_bebob_stream_init_duplex(bebob); + err = snd_card_register(bebob->card); if (err < 0) goto error; - if (!bebob->maudio_special_quirk) { - err = snd_card_register(card); - if (err < 0) { - snd_bebob_stream_destroy_duplex(bebob); - goto error; - } + set_bit(card_index, devices_used); + mutex_unlock(&devices_mutex); + + /* + * After registered, bebob instance can be released corresponding to + * releasing the sound card instance. + */ + bebob->card->private_free = bebob_card_free; + bebob->card->private_data = bebob; + bebob->registered = true; + + return; +error: + mutex_unlock(&devices_mutex); + snd_bebob_stream_destroy_duplex(bebob); + snd_card_free(bebob->card); + dev_info(&bebob->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_bebob *bebob; + const struct snd_bebob_spec *spec; + + if (entry->vendor_id == VEN_FOCUSRITE && + entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH) + spec = get_saffire_spec(unit); + else if (entry->vendor_id == VEN_MAUDIO1 && + entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH && + !check_audiophile_booted(unit)) + spec = NULL; + else + spec = (const struct snd_bebob_spec *)entry->driver_data; + + if (spec == NULL) { + if (entry->vendor_id == VEN_MAUDIO1 || + entry->vendor_id == VEN_MAUDIO2) + return snd_bebob_maudio_load_firmware(unit); + else + return -ENODEV; + } + + /* Allocate this independent of sound card instance. */ + bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL); + if (bebob == NULL) + return -ENOMEM; + + bebob->unit = fw_unit_get(unit); + bebob->entry = entry; + bebob->spec = spec; + dev_set_drvdata(&unit->device, bebob); + + mutex_init(&bebob->mutex); + spin_lock_init(&bebob->lock); + init_waitqueue_head(&bebob->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&bebob->dwork, do_registration); + + if (entry->vendor_id != VEN_MAUDIO1 || + (entry->model_id != MODEL_MAUDIO_FW1814 && + entry->model_id != MODEL_MAUDIO_PROJECTMIX)) { + snd_fw_schedule_registration(unit, &bebob->dwork); } else { /* * This is a workaround. This bus reset seems to have an effect @@ -285,19 +324,11 @@ bebob_probe(struct fw_unit *unit, * signals from dbus and starts I/Os. To avoid I/Os till the * future bus reset, registration is done in next update(). */ - bebob->deferred_registration = true; fw_schedule_bus_reset(fw_parent_device(bebob->unit)->card, false, true); } - dev_set_drvdata(&unit->device, bebob); -end: - mutex_unlock(&devices_mutex); - return err; -error: - mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + return 0; } /* @@ -324,15 +355,11 @@ bebob_update(struct fw_unit *unit) if (bebob == NULL) return; - fcp_bus_reset(bebob->unit); - - if (bebob->deferred_registration) { - if (snd_card_register(bebob->card) < 0) { - snd_bebob_stream_destroy_duplex(bebob); - snd_card_free(bebob->card); - } - bebob->deferred_registration = false; - } + /* Postpone a workqueue for deferred registration. */ + if (!bebob->registered) + snd_fw_schedule_registration(unit, &bebob->dwork); + else + fcp_bus_reset(bebob->unit); } static void bebob_remove(struct fw_unit *unit) @@ -342,8 +369,20 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(bebob->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&bebob->dwork); + + if (bebob->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(bebob->card); + } else { + /* Don't forget this case. */ + bebob_free(bebob); + } } static const struct snd_bebob_rate_spec normal_rate_spec = { diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index b50bb33d9d46..e7f1bb925b12 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -83,6 +83,10 @@ struct snd_bebob { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + + const struct ieee1394_device_id *entry; const struct snd_bebob_spec *spec; unsigned int midi_input_ports; @@ -90,7 +94,6 @@ struct snd_bebob { bool connected; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; @@ -111,7 +114,6 @@ struct snd_bebob { /* for M-Audio special devices */ void *maudio_special_quirk; - bool deferred_registration; /* For BeBoB version quirk. */ unsigned int version; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 77cbb02bff34..4d3034a68bdf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,30 +484,6 @@ destroy_both_connections(struct snd_bebob *bebob) } static int -get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode) -{ - enum snd_bebob_clock_type src; - int err; - - err = snd_bebob_stream_get_clock_src(bebob, &src); - if (err < 0) - return err; - - switch (src) { - case SND_BEBOB_CLOCK_TYPE_INTERNAL: - case SND_BEBOB_CLOCK_TYPE_EXTERNAL: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - default: - case SND_BEBOB_CLOCK_TYPE_SYT: - *sync_mode = 0; - break; - } - - return 0; -} - -static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate) { @@ -550,8 +526,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) goto end; } - bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; - /* * BeBoB v3 transfers packets with these qurks: * - In the beginning of streaming, the value of dbc is incremented @@ -584,8 +558,6 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; - struct amdtp_stream *master, *slave; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -593,22 +565,11 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (bebob->substreams_counter == 0) goto end; - err = get_sync_mode(bebob, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &bebob->tx_stream; - slave = &bebob->rx_stream; - } else { - master = &bebob->rx_stream; - slave = &bebob->tx_stream; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(bebob, master); + err = check_connection_used_by_others(bebob, &bebob->rx_stream); if (err < 0) goto end; @@ -618,11 +579,12 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) * At bus reset, connections should not be broken here. So streams need * to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag. */ - if (amdtp_streaming_error(master)) - amdtp_stream_stop(master); - if (amdtp_streaming_error(slave)) - amdtp_stream_stop(slave); - if (!amdtp_stream_running(master) && !amdtp_stream_running(slave)) + if (amdtp_streaming_error(&bebob->rx_stream)) + amdtp_stream_stop(&bebob->rx_stream); + if (amdtp_streaming_error(&bebob->tx_stream)) + amdtp_stream_stop(&bebob->tx_stream); + if (!amdtp_stream_running(&bebob->rx_stream) && + !amdtp_stream_running(&bebob->tx_stream)) break_both_connections(bebob); /* stop streams if rate is different */ @@ -635,16 +597,13 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - amdtp_stream_stop(master); - amdtp_stream_stop(slave); + amdtp_stream_stop(&bebob->rx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - bebob->master = master; - + if (!amdtp_stream_running(&bebob->rx_stream)) { /* * NOTE: * If establishing connections at first, Yamaha GO46 @@ -666,7 +625,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (err < 0) goto end; - err = start_stream(bebob, master, rate); + err = start_stream(bebob, &bebob->rx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP master stream:%d\n", err); @@ -685,15 +644,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) dev_err(&bebob->unit->device, "fail to ensure sampling rate: %d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } } /* wait first callback */ - if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->rx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; goto end; @@ -701,20 +661,21 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) } /* start slave if needed */ - if (!amdtp_stream_running(slave)) { - err = start_stream(bebob, slave, rate); + if (!amdtp_stream_running(&bebob->tx_stream)) { + err = start_stream(bebob, &bebob->tx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP slave stream:%d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } /* wait first callback */ - if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(slave); - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->tx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->tx_stream); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; } @@ -725,22 +686,12 @@ end: void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { - struct amdtp_stream *master, *slave; - - if (bebob->master == &bebob->rx_stream) { - slave = &bebob->tx_stream; - master = &bebob->rx_stream; - } else { - slave = &bebob->rx_stream; - master = &bebob->tx_stream; - } - if (bebob->substreams_counter == 0) { - amdtp_stream_pcm_abort(master); - amdtp_stream_stop(master); + amdtp_stream_pcm_abort(&bebob->rx_stream); + amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_pcm_abort(slave); - amdtp_stream_stop(slave); + amdtp_stream_pcm_abort(&bebob->tx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 8b64aef31a86..25e9f77275c4 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -20,8 +20,6 @@ MODULE_LICENSE("GPL v2"); #define WEISS_CATEGORY_ID 0x00 #define LOUD_CATEGORY_ID 0x10 -#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) - /* * Some models support several isochronous channels, while these streams are not * always available. In this case, add the model name to this list. @@ -201,6 +199,10 @@ static void do_registration(struct work_struct *work) dice_card_strings(dice); + err = snd_dice_stream_init_duplex(dice); + if (err < 0) + goto error; + snd_dice_create_proc(dice); err = snd_dice_create_pcm(dice); @@ -229,28 +231,14 @@ static void do_registration(struct work_struct *work) return; error: + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); + snd_dice_stream_destroy_duplex(dice); snd_card_free(dice->card); dev_info(&dice->unit->device, "Sound card registration failed: %d\n", err); } -static void schedule_registration(struct snd_dice *dice) -{ - struct fw_card *fw_card = fw_parent_device(dice->unit)->card; - u64 now, delay; - - now = get_jiffies_64(); - delay = fw_card->reset_jiffies + msecs_to_jiffies(PROBE_DELAY_MS); - - if (time_after64(delay, now)) - delay -= now; - else - delay = 0; - - mod_delayed_work(system_wq, &dice->dwork, delay); -} - static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_dice *dice; @@ -273,15 +261,9 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) init_completion(&dice->clock_accepted); init_waitqueue_head(&dice->hwdep_wait); - err = snd_dice_stream_init_duplex(dice); - if (err < 0) { - dice_free(dice); - return err; - } - /* Allocate and register this sound card later. */ INIT_DEFERRABLE_WORK(&dice->dwork, do_registration); - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); return 0; } @@ -312,7 +294,7 @@ static void dice_bus_reset(struct fw_unit *unit) /* Postpone a workqueue for deferred registration. */ if (!dice->registered) - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); /* The handler address register becomes initialized. */ snd_dice_transaction_reinit(dice); @@ -335,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index 0ac92aba5bc1..b3cffd01a19f 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -421,7 +421,7 @@ int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, /* Use different mode between incoming/outgoing. */ if (dir == AMDTP_IN_STREAM) { - flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK; + flags = CIP_NONBLOCKING; process_data_blocks = process_tx_data_blocks; } else { flags = CIP_BLOCKING; diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c index 554324d8c602..735d35640807 100644 --- a/sound/firewire/digi00x/digi00x-transaction.c +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -126,12 +126,17 @@ int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) return err; error: fw_core_remove_address_handler(&dg00x->async_handler); - dg00x->async_handler.address_callback = NULL; + dg00x->async_handler.callback_data = NULL; return err; } void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x) { + if (dg00x->async_handler.callback_data == NULL) + return; + snd_fw_async_midi_port_destroy(&dg00x->out_control); fw_core_remove_address_handler(&dg00x->async_handler); + + dg00x->async_handler.callback_data = NULL; } diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f33b7a1fca4..cc4776c6ded3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -40,10 +40,8 @@ static int name_card(struct snd_dg00x *dg00x) return 0; } -static void dg00x_card_free(struct snd_card *card) +static void dg00x_free(struct snd_dg00x *dg00x) { - struct snd_dg00x *dg00x = card->private_data; - snd_dg00x_stream_destroy_duplex(dg00x); snd_dg00x_transaction_unregister(dg00x); @@ -52,28 +50,24 @@ static void dg00x_card_free(struct snd_card *card) mutex_destroy(&dg00x->mutex); } -static int snd_dg00x_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void dg00x_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_dg00x *dg00x; - int err; + dg00x_free(card->private_data); +} - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_dg00x), &card); - if (err < 0) - return err; - card->private_free = dg00x_card_free; +static void do_registration(struct work_struct *work) +{ + struct snd_dg00x *dg00x = + container_of(work, struct snd_dg00x, dwork.work); + int err; - /* initialize myself */ - dg00x = card->private_data; - dg00x->card = card; - dg00x->unit = fw_unit_get(unit); + if (dg00x->registered) + return; - mutex_init(&dg00x->mutex); - spin_lock_init(&dg00x->lock); - init_waitqueue_head(&dg00x->hwdep_wait); + err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0, + &dg00x->card); + if (err < 0) + return; err = name_card(dg00x); if (err < 0) @@ -101,35 +95,86 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(dg00x->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, dg00x); + dg00x->card->private_free = dg00x_card_free; + dg00x->card->private_data = dg00x; + dg00x->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_dg00x_transaction_unregister(dg00x); + snd_dg00x_stream_destroy_duplex(dg00x); + snd_card_free(dg00x->card); + dev_info(&dg00x->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_dg00x_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_dg00x *dg00x; + + /* Allocate this independent of sound card instance. */ + dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL); + if (dg00x == NULL) + return -ENOMEM; + + dg00x->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, dg00x); + + mutex_init(&dg00x->mutex); + spin_lock_init(&dg00x->lock); + init_waitqueue_head(&dg00x->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&dg00x->dwork, do_registration); + snd_fw_schedule_registration(unit, &dg00x->dwork); + + return 0; } static void snd_dg00x_update(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!dg00x->registered) + snd_fw_schedule_registration(unit, &dg00x->dwork); + snd_dg00x_transaction_reregister(dg00x); - mutex_lock(&dg00x->mutex); - snd_dg00x_stream_update_duplex(dg00x); - mutex_unlock(&dg00x->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (dg00x->registered) { + mutex_lock(&dg00x->mutex); + snd_dg00x_stream_update_duplex(dg00x); + mutex_unlock(&dg00x->mutex); + } } static void snd_dg00x_remove(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dg00x->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&dg00x->dwork); + + if (dg00x->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(dg00x->card); + } else { + /* Don't forget this case. */ + dg00x_free(dg00x); + } } static const struct ieee1394_device_id snd_dg00x_id_table[] = { diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 907e73993677..2cd465c0caae 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -37,6 +37,9 @@ struct snd_dg00x { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + struct amdtp_stream tx_stream; struct fw_iso_resources tx_resources; diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8f27b67503c8..71a0613d3da0 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -168,11 +168,34 @@ get_hardware_info(struct snd_efw *efw) sizeof(struct snd_efw_phys_grp) * hwinfo->phys_in_grp_count); memcpy(&efw->phys_out_grps, hwinfo->phys_out_grps, sizeof(struct snd_efw_phys_grp) * hwinfo->phys_out_grp_count); + + /* AudioFire8 (since 2009) and AudioFirePre8 */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_9) + efw->is_af9 = true; + /* These models uses the same firmware. */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_2 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_4 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_9 || + hwinfo->type == MODEL_GIBSON_RIP || + hwinfo->type == MODEL_GIBSON_GOLDTOP) + efw->is_fireworks3 = true; end: kfree(hwinfo); return err; } +static void efw_free(struct snd_efw *efw) +{ + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); + fw_unit_put(efw->unit); + + kfree(efw->resp_buf); + + mutex_destroy(&efw->mutex); + kfree(efw); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -184,28 +207,24 @@ efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - fw_unit_put(efw->unit); - - kfree(efw->resp_buf); - if (efw->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(efw->card_index, devices_used); mutex_unlock(&devices_mutex); } - mutex_destroy(&efw->mutex); + efw_free(card->private_data); } -static int -efw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_efw *efw; - int card_index, err; + struct snd_efw *efw = container_of(work, struct snd_efw, dwork.work); + unsigned int card_index; + int err; + + if (efw->registered) + return; mutex_lock(&devices_mutex); @@ -215,24 +234,16 @@ efw_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_efw), &card); - if (err < 0) - goto end; - efw = card->private_data; - efw->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = efw_card_free; - - efw->card = card; - efw->unit = fw_unit_get(unit); - mutex_init(&efw->mutex); - spin_lock_init(&efw->lock); - init_waitqueue_head(&efw->hwdep_wait); + err = snd_card_new(&efw->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &efw->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; + } /* prepare response buffer */ snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size, @@ -248,16 +259,10 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; - /* AudioFire8 (since 2009) and AudioFirePre8 */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) - efw->is_af9 = true; - /* These models uses the same firmware. */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_4 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_9 || - entry->model_id == MODEL_GIBSON_RIP || - entry->model_id == MODEL_GIBSON_GOLDTOP) - efw->is_fireworks3 = true; + + err = snd_efw_stream_init_duplex(efw); + if (err < 0) + goto error; snd_efw_proc_init(efw); @@ -275,44 +280,93 @@ efw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_efw_stream_init_duplex(efw); + err = snd_card_register(efw->card); if (err < 0) goto error; - err = snd_card_register(card); - if (err < 0) { - snd_efw_stream_destroy_duplex(efw); - goto error; - } - - dev_set_drvdata(&unit->device, efw); -end: + set_bit(card_index, devices_used); mutex_unlock(&devices_mutex); - return err; + + /* + * After registered, efw instance can be released corresponding to + * releasing the sound card instance. + */ + efw->card->private_free = efw_card_free; + efw->card->private_data = efw; + efw->registered = true; + + return; error: - snd_efw_transaction_remove_instance(efw); mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + snd_efw_transaction_remove_instance(efw); + snd_efw_stream_destroy_duplex(efw); + snd_card_free(efw->card); + dev_info(&efw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_efw *efw; + + efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL); + if (efw == NULL) + return -ENOMEM; + + efw->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, efw); + + mutex_init(&efw->mutex); + spin_lock_init(&efw->lock); + init_waitqueue_head(&efw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&efw->dwork, do_registration); + snd_fw_schedule_registration(unit, &efw->dwork); + + return 0; } static void efw_update(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!efw->registered) + snd_fw_schedule_registration(unit, &efw->dwork); + snd_efw_transaction_bus_reset(efw->unit); - mutex_lock(&efw->mutex); - snd_efw_stream_update_duplex(efw); - mutex_unlock(&efw->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (efw->registered) { + mutex_lock(&efw->mutex); + snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); + } } static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(efw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&efw->dwork); + + if (efw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(efw->card); + } else { + /* Don't forget this case. */ + efw_free(efw); + } } static const struct ieee1394_device_id efw_id_table[] = { diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 96c4e0c6a9bd..03ed35237e2b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -65,6 +65,9 @@ struct snd_efw { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + /* for transaction */ u32 seqnum; bool resp_addr_changable; @@ -81,7 +84,6 @@ struct snd_efw { unsigned int pcm_capture_channels[SND_EFW_MULTIPLIER_MODES]; unsigned int pcm_playback_channels[SND_EFW_MULTIPLIER_MODES]; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 425db8d88235..ee47924aef0d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -121,23 +121,6 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) } static int -get_sync_mode(struct snd_efw *efw, enum cip_flags *sync_mode) -{ - enum snd_efw_clock_source clock_source; - int err; - - err = snd_efw_command_get_clock_source(efw, &clock_source); - if (err < 0) - return err; - - if (clock_source == SND_EFW_CLOCK_SOURCE_SYTMATCH) - return -ENOSYS; - - *sync_mode = CIP_SYNC_TO_DEVICE; - return 0; -} - -static int check_connection_used_by_others(struct snd_efw *efw, struct amdtp_stream *s) { struct cmp_connection *conn; @@ -208,9 +191,6 @@ end: int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) { - struct amdtp_stream *master, *slave; - unsigned int slave_substreams; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -218,32 +198,19 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (efw->playback_substreams == 0 && efw->capture_substreams == 0) goto end; - err = get_sync_mode(efw, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &efw->tx_stream; - slave = &efw->rx_stream; - slave_substreams = efw->playback_substreams; - } else { - master = &efw->rx_stream; - slave = &efw->tx_stream; - slave_substreams = efw->capture_substreams; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(efw, master); + err = check_connection_used_by_others(efw, &efw->rx_stream); if (err < 0) goto end; /* packet queueing error */ - if (amdtp_streaming_error(slave)) - stop_stream(efw, slave); - if (amdtp_streaming_error(master)) - stop_stream(efw, master); + if (amdtp_streaming_error(&efw->tx_stream)) + stop_stream(efw, &efw->tx_stream); + if (amdtp_streaming_error(&efw->rx_stream)) + stop_stream(efw, &efw->rx_stream); /* stop streams if rate is different */ err = snd_efw_command_get_sampling_rate(efw, &curr_rate); @@ -252,20 +219,17 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - stop_stream(efw, slave); - stop_stream(efw, master); + stop_stream(efw, &efw->tx_stream); + stop_stream(efw, &efw->rx_stream); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - efw->master = master; - + if (!amdtp_stream_running(&efw->rx_stream)) { err = snd_efw_command_set_sampling_rate(efw, rate); if (err < 0) goto end; - err = start_stream(efw, master, rate); + err = start_stream(efw, &efw->rx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP master stream:%d\n", err); @@ -274,12 +238,13 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) } /* start slave if needed */ - if (slave_substreams > 0 && !amdtp_stream_running(slave)) { - err = start_stream(efw, slave, rate); + if (efw->capture_substreams > 0 && + !amdtp_stream_running(&efw->tx_stream)) { + err = start_stream(efw, &efw->tx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP slave stream:%d\n", err); - stop_stream(efw, master); + stop_stream(efw, &efw->rx_stream); } } end: @@ -288,26 +253,11 @@ end: void snd_efw_stream_stop_duplex(struct snd_efw *efw) { - struct amdtp_stream *master, *slave; - unsigned int master_substreams, slave_substreams; - - if (efw->master == &efw->rx_stream) { - slave = &efw->tx_stream; - master = &efw->rx_stream; - slave_substreams = efw->capture_substreams; - master_substreams = efw->playback_substreams; - } else { - slave = &efw->rx_stream; - master = &efw->tx_stream; - slave_substreams = efw->playback_substreams; - master_substreams = efw->capture_substreams; - } - - if (slave_substreams == 0) { - stop_stream(efw, slave); + if (efw->capture_substreams == 0) { + stop_stream(efw, &efw->tx_stream); - if (master_substreams == 0) - stop_stream(efw, master); + if (efw->playback_substreams == 0) + stop_stream(efw, &efw->rx_stream); } } diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index f80aafa44c89..ca4dfcf43175 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -67,6 +67,38 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, } EXPORT_SYMBOL(snd_fw_transaction); +#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) + +/** + * snd_fw_schedule_registration - schedule work for sound card registration + * @unit: an instance for unit on IEEE 1394 bus + * @dwork: delayed work with callback function + * + * This function is not designed for general purposes. When new unit is + * connected to IEEE 1394 bus, the bus is under bus-reset state because of + * topological change. In this state, units tend to fail both of asynchronous + * and isochronous communication. To avoid this problem, this function is used + * to postpone sound card registration after the state. The callers must + * set up instance of delayed work in advance. + */ +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork) +{ + u64 now, delay; + + now = get_jiffies_64(); + delay = fw_parent_device(unit)->card->reset_jiffies + + msecs_to_jiffies(PROBE_DELAY_MS); + + if (time_after64(delay, now)) + delay -= now; + else + delay = 0; + + mod_delayed_work(system_wq, dwork, delay); +} +EXPORT_SYMBOL(snd_fw_schedule_registration); + static void async_midi_port_callback(struct fw_card *card, int rcode, void *data, size_t length, void *callback_data) diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index f3f6f84c48d6..f6769312ebfc 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -22,6 +22,9 @@ static inline bool rcode_is_permanent_error(int rcode) return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR; } +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork); + struct snd_fw_async_midi_port; typedef int (*snd_fw_async_midi_port_fill)( struct snd_rawmidi_substream *substream, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 7cb5743c073b..d9361f352133 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -242,8 +242,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * blocks than IEC 61883-6 defines. */ if (stream == &oxfw->tx_stream) { - oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | - CIP_JUMBO_PAYLOAD; + oxfw->tx_stream.flags |= CIP_JUMBO_PAYLOAD; if (oxfw->wrong_dbs) oxfw->tx_stream.flags |= CIP_WRONG_DBS; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index abedc2207261..e629b88f7d93 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -118,15 +118,8 @@ end: return err; } -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void oxfw_card_free(struct snd_card *card) +static void oxfw_free(struct snd_oxfw *oxfw) { - struct snd_oxfw *oxfw = card->private_data; unsigned int i; snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); @@ -144,6 +137,17 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ +static void oxfw_card_free(struct snd_card *card) +{ + oxfw_free(card->private_data); +} + static int detect_quirks(struct snd_oxfw *oxfw) { struct fw_device *fw_dev = fw_parent_device(oxfw->unit); @@ -205,41 +209,39 @@ static int detect_quirks(struct snd_oxfw *oxfw) return 0; } -static int oxfw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_oxfw *oxfw; + struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); int err; - if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) - return -ENODEV; + if (oxfw->registered) + return; - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*oxfw), &card); + err = snd_card_new(&oxfw->unit->device, -1, NULL, THIS_MODULE, 0, + &oxfw->card); if (err < 0) - return err; + return; - card->private_free = oxfw_card_free; - oxfw = card->private_data; - oxfw->card = card; - mutex_init(&oxfw->mutex); - oxfw->unit = fw_unit_get(unit); - oxfw->entry = entry; - spin_lock_init(&oxfw->lock); - init_waitqueue_head(&oxfw->hwdep_wait); + err = name_card(oxfw); + if (err < 0) + goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; - err = name_card(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; + if (oxfw->has_output) { + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); + if (err < 0) + goto error; + } err = snd_oxfw_create_pcm(oxfw); if (err < 0) @@ -255,54 +257,97 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); + err = snd_card_register(oxfw->card); if (err < 0) goto error; - if (oxfw->has_output) { - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); - if (err < 0) - goto error; - } - err = snd_card_register(card); - if (err < 0) { - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - goto error; - } + /* + * After registered, oxfw instance can be released corresponding to + * releasing the sound card instance. + */ + oxfw->card->private_free = oxfw_card_free; + oxfw->card->private_data = oxfw; + oxfw->registered = true; + + return; +error: + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + snd_card_free(oxfw->card); + dev_info(&oxfw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int oxfw_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_oxfw *oxfw; + + if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) + return -ENODEV; + + /* Allocate this independent of sound card instance. */ + oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL); + if (oxfw == NULL) + return -ENOMEM; + + oxfw->entry = entry; + oxfw->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, oxfw); + mutex_init(&oxfw->mutex); + spin_lock_init(&oxfw->lock); + init_waitqueue_head(&oxfw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&oxfw->dwork, do_registration); + snd_fw_schedule_registration(unit, &oxfw->dwork); + return 0; -error: - snd_card_free(card); - return err; } static void oxfw_bus_reset(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + if (!oxfw->registered) + snd_fw_schedule_registration(unit, &oxfw->dwork); + fcp_bus_reset(oxfw->unit); - mutex_lock(&oxfw->mutex); + if (oxfw->registered) { + mutex_lock(&oxfw->mutex); - snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); + snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); - mutex_unlock(&oxfw->mutex); + mutex_unlock(&oxfw->mutex); - if (oxfw->entry->vendor_id == OUI_STANTON) - snd_oxfw_scs1x_update(oxfw); + if (oxfw->entry->vendor_id == OUI_STANTON) + snd_oxfw_scs1x_update(oxfw); + } } static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(oxfw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&oxfw->dwork); + + if (oxfw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(oxfw->card); + } else { + /* Don't forget this case. */ + oxfw_free(oxfw); + } } static const struct compat_info griffin_firewave = { diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 9beecc214767..2047dcb27625 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -36,10 +36,12 @@ struct snd_oxfw { struct snd_card *card; struct fw_unit *unit; - const struct device_info *device_info; struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 0e6dd5c61f53..4ad3bd7fd445 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -381,19 +381,17 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) return err; if (curr_rate != rate || - amdtp_streaming_error(&tscm->tx_stream) || - amdtp_streaming_error(&tscm->rx_stream)) { + amdtp_streaming_error(&tscm->rx_stream) || + amdtp_streaming_error(&tscm->tx_stream)) { finish_session(tscm); - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); release_resources(tscm); } - if (!amdtp_stream_running(&tscm->tx_stream)) { - amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE, - &tscm->tx_stream, &tscm->rx_stream); + if (!amdtp_stream_running(&tscm->rx_stream)) { err = keep_resources(tscm, rate); if (err < 0) goto error; @@ -406,27 +404,27 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_stream_start(&tscm->tx_stream, - tscm->tx_resources.channel, + err = amdtp_stream_start(&tscm->rx_stream, + tscm->rx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->tx_stream, + if (!amdtp_stream_wait_callback(&tscm->rx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } } - if (!amdtp_stream_running(&tscm->rx_stream)) { - err = amdtp_stream_start(&tscm->rx_stream, - tscm->rx_resources.channel, + if (!amdtp_stream_running(&tscm->tx_stream)) { + err = amdtp_stream_start(&tscm->tx_stream, + tscm->tx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->rx_stream, + if (!amdtp_stream_wait_callback(&tscm->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -435,8 +433,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; error: - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); finish_session(tscm); release_resources(tscm); diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index e281c338e562..9dc93a7eb9da 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -85,10 +85,8 @@ static int identify_model(struct snd_tscm *tscm) return 0; } -static void tscm_card_free(struct snd_card *card) +static void tscm_free(struct snd_tscm *tscm) { - struct snd_tscm *tscm = card->private_data; - snd_tscm_transaction_unregister(tscm); snd_tscm_stream_destroy_duplex(tscm); @@ -97,44 +95,36 @@ static void tscm_card_free(struct snd_card *card) mutex_destroy(&tscm->mutex); } -static int snd_tscm_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void tscm_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_tscm *tscm; + tscm_free(card->private_data); +} + +static void do_registration(struct work_struct *work) +{ + struct snd_tscm *tscm = container_of(work, struct snd_tscm, dwork.work); int err; - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_tscm), &card); + err = snd_card_new(&tscm->unit->device, -1, NULL, THIS_MODULE, 0, + &tscm->card); if (err < 0) - return err; - card->private_free = tscm_card_free; - - /* initialize myself */ - tscm = card->private_data; - tscm->card = card; - tscm->unit = fw_unit_get(unit); - - mutex_init(&tscm->mutex); - spin_lock_init(&tscm->lock); - init_waitqueue_head(&tscm->hwdep_wait); + return; err = identify_model(tscm); if (err < 0) goto error; - snd_tscm_proc_init(tscm); - - err = snd_tscm_stream_init_duplex(tscm); + err = snd_tscm_transaction_register(tscm); if (err < 0) goto error; - err = snd_tscm_create_pcm_devices(tscm); + err = snd_tscm_stream_init_duplex(tscm); if (err < 0) goto error; - err = snd_tscm_transaction_register(tscm); + snd_tscm_proc_init(tscm); + + err = snd_tscm_create_pcm_devices(tscm); if (err < 0) goto error; @@ -146,35 +136,91 @@ static int snd_tscm_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(tscm->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, tscm); + /* + * After registered, tscm instance can be released corresponding to + * releasing the sound card instance. + */ + tscm->card->private_free = tscm_card_free; + tscm->card->private_data = tscm; + tscm->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_tscm_transaction_unregister(tscm); + snd_tscm_stream_destroy_duplex(tscm); + snd_card_free(tscm->card); + dev_info(&tscm->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_tscm_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_tscm *tscm; + + /* Allocate this independent of sound card instance. */ + tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL); + if (tscm == NULL) + return -ENOMEM; + + /* initialize myself */ + tscm->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, tscm); + + mutex_init(&tscm->mutex); + spin_lock_init(&tscm->lock); + init_waitqueue_head(&tscm->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&tscm->dwork, do_registration); + snd_fw_schedule_registration(unit, &tscm->dwork); + + return 0; } static void snd_tscm_update(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!tscm->registered) + snd_fw_schedule_registration(unit, &tscm->dwork); + snd_tscm_transaction_reregister(tscm); - mutex_lock(&tscm->mutex); - snd_tscm_stream_update_duplex(tscm); - mutex_unlock(&tscm->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (tscm->registered) { + mutex_lock(&tscm->mutex); + snd_tscm_stream_update_duplex(tscm); + mutex_unlock(&tscm->mutex); + } } static void snd_tscm_remove(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(tscm->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&tscm->dwork); + + if (tscm->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(tscm->card); + } else { + /* Don't forget this case. */ + tscm_free(tscm); + } } static const struct ieee1394_device_id snd_tscm_id_table[] = { diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 30ab77e924f7..1f61011579a7 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -51,6 +51,8 @@ struct snd_tscm { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; const struct snd_tscm_spec *spec; struct fw_iso_resources tx_resources; diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 3b7ae24900fd..31b510c5ca0b 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -147,6 +147,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 8c486235c905..9fee464e5d49 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -80,6 +80,22 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io); +/* wait for cmd dmas till they are stopped */ +static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus) +{ + unsigned long timeout; + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN) + && time_before(jiffies, timeout)) + udelay(10); + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN) + && time_before(jiffies, timeout)) + udelay(10); +} + /** * snd_hdac_bus_stop_cmd_io - clean up CORB/RIRB buffers * @bus: HD-audio core bus @@ -90,6 +106,7 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus) /* disable ringbuffer DMAs */ snd_hdac_chip_writeb(bus, RIRBCTL, 0); snd_hdac_chip_writeb(bus, CORBCTL, 0); + hdac_wait_for_cmd_dmas(bus); /* disable unsolicited responses */ snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0); spin_unlock_irq(&bus->reg_lock); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 607bbeaebddf..c9af022676c2 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -158,22 +158,40 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/* There is a fixed mapping between audio pin node and display port - * on current Intel platforms: +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: * Pin Widget 5 - PORT B (port = 1 in i915 driver) * Pin Widget 6 - PORT C (port = 2 in i915 driver) * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) */ -static int pin2port(hda_nid_t pin_nid) +static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { - if (WARN_ON(pin_nid < 5 || pin_nid > 7)) + int base_nid; + + switch (codec->vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + base_nid = 3; + break; + default: + base_nid = 4; + break; + } + + if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) return -1; - return pin_nid - 4; + return pin_nid - base_nid; } /** * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @rate: the sample rate to set * @@ -183,14 +201,15 @@ static int pin2port(hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->sync_audio_rate(acomp->dev, port, rate); @@ -199,7 +218,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); /** * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @audio_enabled: the pointer to store the current audio state * @buffer: the buffer pointer to store ELD bytes @@ -217,16 +236,17 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->get_eld(acomp->dev, port, audio_enabled, @@ -338,6 +358,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct i915_audio_component *acomp; int ret; + if (WARN_ON(hdac_acomp)) + return -EBUSY; + if (!i915_gfx_present()) return -ENODEV; @@ -371,6 +394,7 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; @@ -404,6 +428,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 87041ddd29cb..47a358fab132 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -444,7 +444,7 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_write(codec, reg, val); snd_hdac_power_down_pm(codec); } @@ -470,7 +470,7 @@ static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, err = reg_raw_read(codec, reg, val, uncached); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_read(codec, reg, val, uncached); snd_hdac_power_down_pm(codec); } diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec86263828..c6c75e7e0981 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static int spk_mask_from_spk_alloc(int spk_alloc) +{ + int i; + int spk_mask = eld_speaker_allocation_bits[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + return spk_mask; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; int chs, count = 0; + unsigned long max_chs; + int type; + int spk_alloc, spk_mask; if (size < 8) return -ENOMEM; @@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + spk_mask = spk_mask_from_spk_alloc(spk_alloc); + + max_chs = hweight_long(spk_mask); + + for (chs = 2; chs <= max_chs; chs++) { int i; struct hdac_cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (cap->channels != chs) + continue; + + if (!(cap->spk_mask == (spk_mask & cap->spk_mask))) continue; + + type = chmap->ops.chmap_cea_alloc_validate_get_type( + chmap, cap, chs); + if (type < 0) + return -ENODEV; if (size < 8) return -ENOMEM; + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; + dst += 2; size -= 8; count += 8; + if (size < chs_bytes) return -ENOMEM; + size -= chs_bytes; count += chs_bytes; chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) return -EFAULT; dst += chs; } } + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff5693d..718d5e3b7806 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index b36ea47527e8..0b8d0de87273 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - init_timer(&vnc_timer); - vnc_timer.function = vnc_slider_tick; - vnc_timer.expires = jiffies; - vnc_timer.data = nr_waveartist_devs; - add_timer(&vnc_timer); + setup_timer(&vnc_timer, vnc_slider_tick, + nr_waveartist_devs); + mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index a6d6d8d0867a..df5741a78fd2 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -432,7 +432,10 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr #endif //printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr); spin_unlock(&chip->lock); - return (bytes_to_frames(substream->runtime, current_ptr)); + current_ptr = bytes_to_frames(substream->runtime, current_ptr); + if (current_ptr >= substream->runtime->buffer_size) + current_ptr = 0; + return current_ptr; } /* operators */ diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d460453d7b..8f945341720b 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop, diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 0dc44ebb0032..626cd2167d29 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1548,7 +1548,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, int val = 0; spin_lock_irq(&ensoniq->reg_lock); - if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4) + if (ensoniq->ctrl & ES_1371_GPIO_OUT(4)) val = 1; ucontrol->value.integer.value[0] = val; spin_unlock_irq(&ensoniq->reg_lock); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb02c2d48fd5..7f3b5ed81995 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -50,9 +50,13 @@ config SND_HDA_RECONFIG bool "Allow dynamic codec reconfiguration" help Say Y here to enable the HD-audio codec re-configuration feature. - This adds the sysfs interfaces to allow user to clear the whole - codec configuration, change the codec setup, add extra verbs, - and re-configure the codec dynamically. + It allows user to clear the whole codec configuration, change the + codec setup, add extra verbs, and re-configure the codec dynamically. + + Note that this item alone doesn't provide the sysfs interface, but + enables the feature just for the patch loader below. + If you need the traditional sysfs entries for the manual interaction, + turn on CONFIG_SND_HDA_HWDEP as well. config SND_HDA_INPUT_BEEP bool "Support digital beep via input layer" diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dfaf1a93fb8a..320445f3bf73 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5434,6 +5434,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE); return 0; } @@ -5444,6 +5445,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_gen_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP); return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9a0d1445ca5c..94089fc71884 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -365,8 +365,11 @@ enum { #define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) #define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_KBL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa171) +#define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ + IS_KBL(pci) || IS_KBL_LP(pci) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2181,6 +2184,12 @@ static const struct pci_device_id azx_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake */ + { PCI_DEVICE(0x8086, 0xa171), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake-LP */ + { PCI_DEVICE(0x8086, 0x9d71), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 64e0d1d81ca5..9739fce9e032 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -141,14 +141,6 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_configure(codec); if (err < 0) goto error; - /* rebuild PCMs */ - err = snd_hda_codec_build_pcms(codec); - if (err < 0) - goto error; - /* rebuild mixers */ - err = snd_hda_codec_build_controls(codec); - if (err < 0) - goto error; err = snd_card_register(codec->card); error: snd_hda_power_down(codec); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 17fd81736d3d..0621920f7617 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -115,20 +115,20 @@ static int substream_free_pages(struct azx *chip, /* * Register access ops. Tegra HDA register access is DWORD only. */ -static void hda_tegra_writel(u32 value, u32 *addr) +static void hda_tegra_writel(u32 value, u32 __iomem *addr) { writel(value, addr); } -static u32 hda_tegra_readl(u32 *addr) +static u32 hda_tegra_readl(u32 __iomem *addr) { return readl(addr); } -static void hda_tegra_writew(u16 value, u16 *addr) +static void hda_tegra_writew(u16 value, u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -137,20 +137,20 @@ static void hda_tegra_writew(u16 value, u16 *addr) writel(v, dword_addr); } -static u16 hda_tegra_readw(u16 *addr) +static u16 hda_tegra_readw(u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); return (v >> shift) & 0xffff; } -static void hda_tegra_writeb(u8 value, u8 *addr) +static void hda_tegra_writeb(u8 value, u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -159,10 +159,10 @@ static void hda_tegra_writeb(u8 value, u8 *addr) writel(v, dword_addr); } -static u8 hda_tegra_readb(u8 *addr) +static u8 hda_tegra_readb(u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1483f85999ec..d0d5ad8beac5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -114,6 +114,9 @@ struct hdmi_ops { int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format); + void (*pin_cvt_fixup)(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid); }; struct hdmi_pcm { @@ -684,7 +687,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; - if (is_haswell_plus(codec)) + /* some HW (e.g. HSW+) needs reprogramming the amp at each time */ + 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); @@ -864,9 +868,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, struct hdmi_spec *spec = codec->spec; int err; - if (is_haswell_plus(codec)) - haswell_verify_D0(codec, cvt_nid, pin_nid); - err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); if (err) { @@ -884,7 +885,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, * of the pin. */ static int hdmi_choose_cvt(struct hda_codec *codec, - int pin_idx, int *cvt_id, int *mux_id) + int pin_idx, int *cvt_id) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; @@ -925,8 +926,6 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_id) *cvt_id = cvt_idx; - if (mux_id) - *mux_id = mux_idx; return 0; } @@ -1019,9 +1018,6 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, int mux_idx; struct hdmi_spec *spec = codec->spec; - if (!is_haswell_plus(codec) && !is_valleyview_plus(codec)) - return; - /* On Intel platform, the mapping of converter nid to * mux index of the pins are always the same. * The pin nid may be 0, this means all pins will not @@ -1032,6 +1028,17 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, intel_not_share_assigned_cvt(codec, pin_nid, mux_idx); } +/* skeleton caller of pin_cvt_fixup ops */ +static void pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->ops.pin_cvt_fixup) + spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid); +} + /* called in hdmi_pcm_open when no pin is assigned to the PCM * in dyn_pcm_assign mode. */ @@ -1049,7 +1056,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, if (pcm_idx < 0) return -EINVAL; - err = hdmi_choose_cvt(codec, -1, &cvt_idx, NULL); + err = hdmi_choose_cvt(codec, -1, &cvt_idx); if (err) return err; @@ -1057,7 +1064,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - intel_not_share_assigned_cvt_nid(codec, 0, per_cvt->cvt_nid); + pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid); set_bit(pcm_idx, &spec->pcm_in_use); /* todo: setup spdif ctls assign */ @@ -1089,7 +1096,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, { struct hdmi_spec *spec = codec->spec; struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, pcm_idx, mux_idx = 0; + int pin_idx, cvt_idx, pcm_idx; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; @@ -1118,7 +1125,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, } } - err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); if (err < 0) { mutex_unlock(&spec->pcm_lock); return err; @@ -1135,11 +1142,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + per_pin->mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); + pin_cvt_fixup(codec, per_pin, 0); snd_hda_spdif_ctls_assign(codec, pcm_idx, per_cvt->cvt_nid); @@ -1372,12 +1378,7 @@ static void update_eld(struct hda_codec *codec, * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, - per_pin->mux_idx); - } - + pin_cvt_fixup(codec, per_pin, 0); hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } @@ -1484,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1711,7 +1712,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, * skip pin setup and return 0 to make audio playback * be ongoing */ - intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); mutex_unlock(&spec->pcm_lock); @@ -1724,23 +1725,21 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - /* Verify pin:cvt selections to avoid silent audio after S3. - * After S3, the audio driver restores pin:cvt selections - * but this can happen before gfx is ready and such selection - * is overlooked by HW. Thus multiple pins can share a same - * default convertor and mute control will affect each other, - * which can cause a resumed audio playback become silent - * after S3. - */ - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); - } + + /* Verify pin:cvt selections to avoid silent audio after S3. + * After S3, the audio driver restores pin:cvt selections + * but this can happen before gfx is ready and such selection + * is overlooked by HW. Thus multiple pins can share a same + * default convertor and mute control will affect each other, + * which can cause a resumed audio playback become silent + * after S3. + */ + pin_cvt_fixup(codec, per_pin, 0); /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -1837,6 +1836,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2075,6 +2086,20 @@ static void hdmi_array_free(struct hdmi_spec *spec) snd_array_free(&spec->cvts); } +static void generic_spec_free(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec) { + if (spec->i915_bound) + snd_hdac_i915_exit(&codec->bus->core); + hdmi_array_free(spec); + kfree(spec); + codec->spec = NULL; + } + codec->dp_mst = false; +} + static void generic_hdmi_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2099,10 +2124,7 @@ static void generic_hdmi_free(struct hda_codec *codec) spec->pcm_rec[pcm_idx].jack = NULL; } - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - hdmi_array_free(spec); - kfree(spec); + generic_spec_free(codec); } #ifdef CONFIG_PM @@ -2140,6 +2162,55 @@ static const struct hdmi_ops generic_standard_hdmi_ops = { .setup_stream = hdmi_setup_stream, }; +/* allocate codec->spec and assign/initialize generic parser ops */ +static int alloc_generic_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + spec->ops = generic_standard_hdmi_ops; + mutex_init(&spec->pcm_lock); + snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); + + spec->chmap.ops.get_chmap = hdmi_get_chmap; + spec->chmap.ops.set_chmap = hdmi_set_chmap; + spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, + + codec->spec = spec; + hdmi_array_init(spec, 4); + + codec->patch_ops = generic_hdmi_patch_ops; + + return 0; +} + +/* generic HDMI parser */ +static int patch_generic_hdmi(struct hda_codec *codec) +{ + int err; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; + } + + generic_hdmi_init_per_pins(codec); + return 0; +} + +/* + * Intel codec parsers and helpers + */ + static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t nid) { @@ -2217,12 +2288,23 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, static void intel_pin_eld_notify(void *audio_ptr, int port) { struct hda_codec *codec = audio_ptr; - int pin_nid = port + 0x04; + int pin_nid; /* we assume only from port-B to port-D */ if (port < 1 || port > 3) return; + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + pin_nid = port + 0x03; + break; + default: + pin_nid = port + 0x04; + break; + } + /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ @@ -2236,93 +2318,159 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) check_presence_and_report(codec, pin_nid); } -static int patch_generic_hdmi(struct hda_codec *codec) +/* register i915 component pin_eld_notify callback */ +static void register_i915_notifier(struct hda_codec *codec) { - struct hdmi_spec *spec; + struct hdmi_spec *spec = codec->spec; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; + spec->use_acomp_notifier = true; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); +} - spec->ops = generic_standard_hdmi_ops; - mutex_init(&spec->pcm_lock); - snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); +/* setup_stream ops override for HSW+ */ +static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format) +{ + haswell_verify_D0(codec, cvt_nid, pin_nid); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); +} - spec->chmap.ops.get_chmap = hdmi_get_chmap; - spec->chmap.ops.set_chmap = hdmi_set_chmap; - spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; +/* pin_cvt_fixup ops override for HSW+ and VLV+ */ +static void i915_pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + if (per_pin) { + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, per_pin->pin_nid, + per_pin->mux_idx); + } else { + intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + } +} - codec->spec = spec; - hdmi_array_init(spec, 4); +/* Intel Haswell and onwards; audio component with eld notifier */ +static int patch_i915_hsw_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; -#ifdef CONFIG_SND_HDA_I915 - /* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */ - if ((codec->core.vendor_id >> 16) == 0x8086 && - is_haswell_plus(codec)) { -#if 0 - /* on-demand binding leads to an unbalanced refcount when - * both i915 and hda drivers are probed concurrently; - * disabled temporarily for now - */ - if (!codec->bus->core.audio_component) - if (!snd_hdac_i915_init(&codec->bus->core)) - spec->i915_bound = true; -#endif - /* use i915 audio component notifier for hotplug */ - if (codec->bus->core.audio_component) - spec->use_acomp_notifier = true; + /* HSW+ requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } -#endif - if (is_haswell_plus(codec)) { - intel_haswell_enable_all_pins(codec, true); - intel_haswell_fixup_enable_dp12(codec); - } + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - /* For Valleyview/Cherryview, only the display codec is in the display - * power well and can use link_power ops to request/release the power. - * For Haswell/Broadwell, the controller is also in the power well and + intel_haswell_enable_all_pins(codec, true); + intel_haswell_fixup_enable_dp12(codec); + + /* For Haswell/Broadwell, the controller is also in the power well and * can cover the codec power request, and so need not set this flag. - * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec) || - is_broxton(codec)) + if (!is_haswell(codec) && !is_broadwell(codec)) codec->core.link_power_control = 1; - if (hdmi_parse_codec(codec) < 0) { - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - codec->spec = NULL; - kfree(spec); - return -EINVAL; + codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; + + spec->ops.setup_stream = i915_hsw_setup_stream; + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - codec->patch_ops = generic_hdmi_patch_ops; - if (is_haswell_plus(codec)) { - codec->patch_ops.set_power_state = haswell_set_power_state; - codec->dp_mst = true; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; +} + +/* Intel Baytrail and Braswell; with eld notifier */ +static int patch_i915_byt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } - /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - codec->auto_runtime_pm = 1; + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - generic_hdmi_init_per_pins(codec); + /* For Valleyview/Cherryview, only the display codec is in the display + * power well and can use link_power ops to request/release the power. + */ + codec->core.link_power_control = 1; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; +} + +/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ +static int patch_i915_cpt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* no i915 component should have been bound before this */ + if (WARN_ON(codec->bus->core.audio_component)) + return -EBUSY; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; + + /* Try to bind with i915 now */ + err = snd_hdac_i915_init(&codec->bus->core); + if (err < 0) + goto error; + spec->i915_bound = true; + + err = hdmi_parse_codec(codec); + if (err < 0) + goto error; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); return 0; + + error: + generic_spec_free(codec); + return err; } /* @@ -3401,6 +3549,9 @@ static int patch_atihdmi(struct hda_codec *codec) spec->ops.pin_hbr_setup = atihdmi_pin_hbr_setup; spec->ops.setup_stream = atihdmi_setup_stream; + spec->chmap.ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel; + spec->chmap.ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; + if (!has_amd_full_remap_support(codec)) { /* override to ATI/AMD-specific versions with pairwise mapping */ spec->chmap.ops.chmap_cea_alloc_validate_get_type = @@ -3408,10 +3559,6 @@ static int patch_atihdmi(struct hda_codec *codec) spec->chmap.ops.cea_alloc_to_tlv_chmap = atihdmi_paired_cea_alloc_to_tlv_chmap; spec->chmap.ops.chmap_validate = atihdmi_paired_chmap_validate; - spec->chmap.ops.pin_get_slot_channel = - atihdmi_pin_get_slot_channel; - spec->chmap.ops.pin_set_slot_channel = - atihdmi_pin_set_slot_channel; } /* ATI/AMD converters do not advertise all of their capabilities */ @@ -3493,21 +3640,21 @@ HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), +HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac4490a96863..900bfbc3368c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -335,6 +335,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: + case 0x10ec0295: case 0x10ec0298: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; @@ -342,6 +343,14 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + case 0x10ec0700: + case 0x10ec0701: + case 0x10ec0703: + alc_update_coef_idx(codec, 0x10, 1<<15, 0); + break; case 0x10ec0662: if ((coef & 0x00f0) == 0x0030) alc_update_coef_idx(codec, 0x4, 1<<10, 0); /* EAPD Ctrl */ @@ -902,6 +911,7 @@ static struct alc_codec_rename_pci_table rename_pci_tbl[] = { { 0x10ec0298, 0x1028, 0, "ALC3266" }, { 0x10ec0256, 0x1028, 0, "ALC3246" }, { 0x10ec0225, 0x1028, 0, "ALC3253" }, + { 0x10ec0295, 0x1028, 0, "ALC3254" }, { 0x10ec0670, 0x1025, 0, "ALC669X" }, { 0x10ec0676, 0x1025, 0, "ALC679X" }, { 0x10ec0282, 0x1043, 0, "ALC3229" }, @@ -2647,6 +2657,8 @@ enum { ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, ALC269_TYPE_ALC225, + ALC269_TYPE_ALC294, + ALC269_TYPE_ALC700, }; /* @@ -2677,6 +2689,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: case ALC269_TYPE_ALC225: + case ALC269_TYPE_ALC294: + case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; default: @@ -3609,13 +3623,20 @@ static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, static void alc_headset_mode_unplugged(struct hda_codec *codec) { static struct coef_fw coef0255[] = { - WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */ WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */ UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/ WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */ WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */ {} }; + static struct coef_fw coef0255_1[] = { + WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */ + {} + }; + static struct coef_fw coef0256[] = { + WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */ + {} + }; static struct coef_fw coef0233[] = { WRITE_COEF(0x1b, 0x0c0b), WRITE_COEF(0x45, 0xc429), @@ -3668,7 +3689,11 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0255: + alc_process_coef_fw(codec, coef0255_1); + alc_process_coef_fw(codec, coef0255); + break; case 0x10ec0256: + alc_process_coef_fw(codec, coef0256); alc_process_coef_fw(codec, coef0255); break; case 0x10ec0233: @@ -3690,6 +3715,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) alc_process_coef_fw(codec, coef0668); break; case 0x10ec0225: + case 0x10ec0295: alc_process_coef_fw(codec, coef0225); break; } @@ -3790,6 +3816,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; case 0x10ec0225: + case 0x10ec0295: alc_update_coef_idx(codec, 0x45, 0x3f<<10, 0x31<<10); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0225); @@ -3847,6 +3874,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0225: + case 0x10ec0295: alc_process_coef_fw(codec, coef0225); break; case 0x10ec0255: @@ -3884,6 +3912,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) WRITE_COEFEX(0x57, 0x03, 0x8ea6), {} }; + static struct coef_fw coef0256[] = { + WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */ + WRITE_COEF(0x1b, 0x0c6b), + WRITE_COEFEX(0x57, 0x03, 0x8ea6), + {} + }; static struct coef_fw coef0233[] = { WRITE_COEF(0x45, 0xd429), WRITE_COEF(0x1b, 0x0c2b), @@ -3924,9 +3958,11 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0255: - case 0x10ec0256: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0256: + alc_process_coef_fw(codec, coef0256); + break; case 0x10ec0233: case 0x10ec0283: alc_process_coef_fw(codec, coef0233); @@ -3950,6 +3986,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_process_coef_fw(codec, coef0688); break; case 0x10ec0225: + case 0x10ec0295: alc_process_coef_fw(codec, coef0225); break; } @@ -3965,6 +4002,12 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) WRITE_COEFEX(0x57, 0x03, 0x8ea6), {} }; + static struct coef_fw coef0256[] = { + WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */ + WRITE_COEF(0x1b, 0x0c6b), + WRITE_COEFEX(0x57, 0x03, 0x8ea6), + {} + }; static struct coef_fw coef0233[] = { WRITE_COEF(0x45, 0xe429), WRITE_COEF(0x1b, 0x0c2b), @@ -4005,9 +4048,11 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0255: - case 0x10ec0256: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0256: + alc_process_coef_fw(codec, coef0256); + break; case 0x10ec0233: case 0x10ec0283: alc_process_coef_fw(codec, coef0233); @@ -4031,6 +4076,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_process_coef_fw(codec, coef0688); break; case 0x10ec0225: + case 0x10ec0295: alc_process_coef_fw(codec, coef0225); break; } @@ -4114,6 +4160,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) is_ctia = (val & 0x1c02) == 0x1c02; break; case 0x10ec0225: + case 0x10ec0295: alc_process_coef_fw(codec, coef0225); msleep(800); val = alc_read_coef_idx(codec, 0x46); @@ -4251,7 +4298,7 @@ static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec, static void alc255_set_default_jack_type(struct hda_codec *codec) { /* Set to iphone type */ - static struct coef_fw fw[] = { + static struct coef_fw alc255fw[] = { WRITE_COEF(0x1b, 0x880b), WRITE_COEF(0x45, 0xd089), WRITE_COEF(0x1b, 0x080b), @@ -4259,7 +4306,22 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) WRITE_COEF(0x1b, 0x0c0b), {} }; - alc_process_coef_fw(codec, fw); + static struct coef_fw alc256fw[] = { + WRITE_COEF(0x1b, 0x884b), + WRITE_COEF(0x45, 0xd089), + WRITE_COEF(0x1b, 0x084b), + WRITE_COEF(0x46, 0x0004), + WRITE_COEF(0x1b, 0x0c4b), + {} + }; + switch (codec->core.vendor_id) { + case 0x10ec0255: + alc_process_coef_fw(codec, alc255fw); + break; + case 0x10ec0256: + alc_process_coef_fw(codec, alc256fw); + break; + } msleep(30); } @@ -5459,8 +5521,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -5571,6 +5634,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), @@ -5586,6 +5650,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460), + SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -5704,6 +5770,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170130}, + {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, {0x21, 0x02211020}), @@ -5756,11 +5825,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60180}, {0x14, 0x90170130}, {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5565", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60180}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60170}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, @@ -6026,8 +6103,22 @@ static int patch_alc269(struct hda_codec *codec) alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; case 0x10ec0225: + case 0x10ec0295: spec->codec_variant = ALC269_TYPE_ALC225; break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + spec->codec_variant = ALC269_TYPE_ALC294; + break; + case 0x10ec0700: + case 0x10ec0701: + case 0x10ec0703: + spec->codec_variant = ALC269_TYPE_ALC700; + spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ + alc_update_coef_idx(codec, 0x4a, 0, 1 << 15); /* Combo jack auto trigger control */ + break; + } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -6426,6 +6517,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6672,6 +6764,15 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6694,11 +6795,14 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), @@ -6929,6 +7033,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), @@ -6939,6 +7044,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269), HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269), HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0274, "ALC274", patch_alc269), HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269), HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269), HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269), @@ -6951,6 +7057,8 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269), HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), @@ -6966,6 +7074,9 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662), HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662), HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680), + HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 59ab6cee1ad8..f0955fd7a2e7 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -13,7 +13,7 @@ static void (*old_vmaster_hook)(void *, int); static bool is_thinkpad(struct hda_codec *codec) { return (codec->core.subsystem_id >> 16 == 0x17aa) && - (acpi_dev_present("LEN0068") || acpi_dev_present("IBM0068")); + (acpi_dev_found("LEN0068") || acpi_dev_found("IBM0068")); } static void update_tpacpi_mute_led(void *private_data, int enabled) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8151318a69a2..9720a30dbfff 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -42,12 +42,6 @@ #include <asm/pgtable.h> #include <asm/cacheflush.h> -#ifdef CONFIG_KVM_GUEST -#include <linux/kvm_para.h> -#else -#define kvm_para_available() (0) -#endif - MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); @@ -2972,25 +2966,17 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) goto fini; } - /* detect KVM and Parallels virtual environments */ - result = kvm_para_available(); -#ifdef X86_FEATURE_HYPERVISOR - result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); -#endif - if (!result) - goto fini; - /* check for known (emulated) devices */ + result = 0; if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET && pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) { /* KVM emulated sound, PCI SSID: 1af4:1100 */ msg = "enable KVM"; + result = 1; } else if (pci->subsystem_vendor == 0x1ab8) { /* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */ msg = "enable Parallels VM"; - } else { - msg = "disable (unknown or VT-d) VM"; - result = 0; + result = 1; } fini: diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index f3d62020ef66..a80684bdc30d 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -644,7 +644,7 @@ static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, if (err < 0) return err; - if (current_state == state) + if (!err && current_state == state) return 0; mdelay(1); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3afae990e39..f3fb98f0a995 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C + select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C select SND_SOC_CS42L51_I2C if I2C @@ -64,7 +65,6 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC - select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C select SND_SOC_GTM601 @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -126,12 +127,14 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C + select SND_SOC_TAS5720 if I2C select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4_I2C if I2C + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -367,6 +370,9 @@ config SND_SOC_ALC5623 config SND_SOC_ALC5632 tristate +config SND_SOC_BT_SCO + tristate + config SND_SOC_CQ0093VC tristate @@ -473,16 +479,14 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_BT_SCO - tristate - config SND_SOC_DMIC tristate config SND_SOC_HDMI_CODEC - tristate - select SND_PCM_ELD - select SND_PCM_IEC958 + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + select HDMI config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" @@ -529,6 +533,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate @@ -748,8 +755,15 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers" + tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + depends on I2C + +config SND_SOC_TAS5720 + tristate "Texas Instruments TAS5720 Mono Audio amplifier" depends on I2C + help + Enable support for Texas Instruments TAS5720L/M high-efficiency mono + Class-D audio power amplifiers. config SND_SOC_TFA9879 tristate "NXP Semiconductors TFA9879 amplifier" @@ -780,6 +794,16 @@ config SND_SOC_TLV320AIC31XX config SND_SOC_TLV320AIC32X4 tristate +config SND_SOC_TLV320AIC32X4_I2C + tristate + depends on I2C + select SND_SOC_TLV320AIC32X4 + +config SND_SOC_TLV320AIC32X4_SPI + tristate + depends on SPI_MASTER + select SND_SOC_TLV320AIC32X4 + config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" depends on I2C @@ -920,7 +944,8 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate + tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b7b99416537f..0f548fd34ca3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -32,6 +32,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o snd-soc-cs42l51-objs := cs42l51.o @@ -55,7 +56,6 @@ snd-soc-da7218-objs := da7218.o snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o @@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o @@ -131,6 +132,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas5720-objs := tas5720.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -138,6 +140,8 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o +snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -243,6 +247,7 @@ obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o @@ -266,7 +271,6 @@ obj-$(CONFIG_SND_SOC_DA7218) += snd-soc-da7218.o obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o @@ -339,6 +343,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o @@ -346,6 +351,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 647f69de6baa..5013d2ba0c10 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -146,6 +146,7 @@ static const struct regmap_config ak4613_regmap_cfg = { .max_register = 0x16, .reg_defaults = ak4613_reg, .num_reg_defaults = ARRAY_SIZE(ak4613_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct of_device_id ak4613_of_match[] = { @@ -530,7 +531,6 @@ static int ak4613_i2c_remove(struct i2c_client *client) static struct i2c_driver ak4613_i2c_driver = { .driver = { .name = "ak4613-codec", - .owner = THIS_MODULE, .of_match_table = ak4613_of_match, }, .probe = ak4613_i2c_probe, diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1ee8506c06c7..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d6f4abbbf8a7..fb3885fe0afb 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty) if (!tty->disc_data) return -ENODEV; + tty->receive_room = 16; if (tty->ops->write(tty, v253_init, len) != len) { ret = -EIO; goto err; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 181cd3bf0b92..2abb742fc47b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1474,6 +1474,11 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) * exit, we call pm_runtime_suspend() so that will do for us */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); ret = create_fill_widget_route_map(dapm); @@ -1634,6 +1639,11 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) /* hold the ref while we probe */ hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + if (!hlink) { + dev_err(&edev->hdac.dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(edev->ebus, hlink); hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); @@ -1744,6 +1754,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) } hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_put(ebus, hlink); return 0; @@ -1765,6 +1780,11 @@ static int hdac_hdmi_runtime_resume(struct device *dev) return 0; hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + if (!hlink) { + dev_err(dev, "hdac link not found\n"); + return -EIO; + } + snd_hdac_ext_bus_link_get(ebus, hlink); err = snd_hdac_display_power(bus, true); diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 000000000000..cf0a39bb631a --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,441 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN, + 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0, + max98371_dht_max_gain), + SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0, + max98371_dht_min_gain), + SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0, + max98371_dht_rot_gain), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, ®); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 000000000000..9f6330964d98 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index a1aaffc20862..f80cfe4d2ef2 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -276,6 +276,8 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) } else { *mic = false; regmap_write(rt298->regmap, RT298_SET_MIC1, 0x20); + regmap_update_bits(rt298->regmap, + RT298_CBJ_CTRL1, 0x0400, 0x0000); } } else { regmap_read(rt298->regmap, RT298_GET_HP_SENSE, &buf); @@ -482,6 +484,26 @@ static int rt298_adc_event(struct snd_soc_dapm_widget *w, snd_soc_update_bits(codec, VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), 0x7080, 0x7000); + /* If MCLK doesn't exist, reset AD filter */ + if (!(snd_soc_read(codec, RT298_VAD_CTRL) & 0x200)) { + pr_info("NO MCLK\n"); + switch (nid) { + case RT298_ADC_IN1: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x2); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x2, 0x0); + break; + case RT298_ADC_IN2: + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x4); + mdelay(10); + snd_soc_update_bits(codec, + RT298_D_FILTER_CTRL, 0x4, 0x0); + break; + } + } break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, @@ -520,30 +542,12 @@ static int rt298_mic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt298_vref_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0000); - mdelay(50); - break; - default: - return 0; - } - - return 0; -} - static const struct snd_soc_dapm_widget rt298_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("HV", 1, RT298_POWER_CTRL1, 12, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("VREF", RT298_POWER_CTRL1, - 0, 1, rt298_vref_event, SND_SOC_DAPM_PRE_PMU), + 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY_S("BG_MBIAS", 1, RT298_POWER_CTRL2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("LDO1", 1, RT298_POWER_CTRL2, @@ -934,18 +938,9 @@ static int rt298_set_bias_level(struct snd_soc_codec *codec, } break; - case SND_SOC_BIAS_ON: - mdelay(30); - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0400); - - break; - case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT298_SET_AUDIO_POWER, AC_PWRST_D3); - snd_soc_update_bits(codec, - RT298_CBJ_CTRL1, 0x0400, 0x0000); break; default: diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index d66f8847b676..3638f3d61209 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -137,6 +137,7 @@ #define RT298_A_BIAS_CTRL2 0x02 #define RT298_POWER_CTRL1 0x03 #define RT298_A_BIAS_CTRL3 0x04 +#define RT298_D_FILTER_CTRL 0x05 #define RT298_POWER_CTRL2 0x08 #define RT298_I2S_CTRL1 0x09 #define RT298_I2S_CTRL2 0x0a @@ -148,6 +149,7 @@ #define RT298_IRQ_CTRL 0x33 #define RT298_WIND_FILTER_CTRL 0x46 #define RT298_PLL_CTRL1 0x49 +#define RT298_VAD_CTRL 0x4e #define RT298_CBJ_CTRL1 0x4f #define RT298_CBJ_CTRL2 0x50 #define RT298_PLL_CTRL 0x63 diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 3c6594da6c9c..d70847c9eeb0 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -253,7 +253,7 @@ static const struct reg_default rt5650_reg[] = { { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, { 0x2d, 0x0000 }, - { 0x2f, 0x1002 }, + { 0x2f, 0x5002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, { 0x33, 0x0000 }, diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 49a9e7049e2b..0af5ddbef1da 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -619,7 +619,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, - 39, 0, out_vol_tlv), + 39, 1, out_vol_tlv), /* OUTPUT Control */ SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 33e290b703df..da9483c1c6fb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1241,60 +1241,46 @@ static int rt5677_dmic_use_asrc(struct snd_soc_dapm_widget *source, regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO1_CLK_SEL_MASK) >> RT5677_AD_STO1_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 10: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO2_CLK_SEL_MASK) >> RT5677_AD_STO2_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 9: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO3_CLK_SEL_MASK) >> RT5677_AD_STO3_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 8: regmap_read(rt5677->regmap, RT5677_ASRC_5, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_STO4_CLK_SEL_MASK) >> RT5677_AD_STO4_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 7: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOL_CLK_SEL_MASK) >> RT5677_AD_MONOL_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; case 6: regmap_read(rt5677->regmap, RT5677_ASRC_6, &asrc_setting); asrc_setting = (asrc_setting & RT5677_AD_MONOR_CLK_SEL_MASK) >> RT5677_AD_MONOR_CLK_SEL_SFT; - if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && - asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) - return 1; break; default: - break; + return 0; } + if (asrc_setting >= RT5677_CLK_SEL_I2S1_ASRC && + asrc_setting <= RT5677_CLK_SEL_I2S6_ASRC) + return 1; + return 0; } @@ -4520,14 +4506,9 @@ static int rt5677_set_bias_level(struct snd_soc_codec *codec, } #ifdef CONFIG_GPIOLIB -static inline struct rt5677_priv *gpio_to_rt5677(struct gpio_chip *chip) -{ - return container_of(chip, struct rt5677_priv, gpio_chip); -} - static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4548,7 +4529,7 @@ static void rt5677_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int rt5677_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4572,7 +4553,7 @@ static int rt5677_gpio_direction_out(struct gpio_chip *chip, static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); int value, ret; ret = regmap_read(rt5677->regmap, RT5677_GPIO_ST, &value); @@ -4584,7 +4565,7 @@ static int rt5677_gpio_get(struct gpio_chip *chip, unsigned offset) static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); switch (offset) { case RT5677_GPIO1 ... RT5677_GPIO5: @@ -4638,7 +4619,7 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset, static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset) { - struct rt5677_priv *rt5677 = gpio_to_rt5677(chip); + struct rt5677_priv *rt5677 = gpiochip_get_data(chip); struct regmap_irq_chip_data *data = rt5677->irq_data; int irq; @@ -4697,7 +4678,7 @@ static void rt5677_init_gpio(struct i2c_client *i2c) rt5677->gpio_chip.parent = &i2c->dev; rt5677->gpio_chip.base = -1; - ret = gpiochip_add(&rt5677->gpio_chip); + ret = gpiochip_add_data(&rt5677->gpio_chip, rt5677); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad41a34..b8d19b77bde9 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -4,6 +4,9 @@ * Copyright (C) 2015 Google, Inc. * Copyright (c) 2013 Daniel Mack <zonque@gmail.com> * + * TAS5721 support: + * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com> + * * 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 @@ -57,6 +60,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -167,6 +174,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } +static int tas571x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 sysctl2; + int ret; + + sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0; + + ret = snd_soc_update_bits(codec, + TAS571X_SYS_CTRL_2_REG, + TAS571X_SYS_CTRL_2_SDN_MASK, + sysctl2); + usleep_range(1000, 2000); + + return ret; +} + static int tas571x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -214,6 +238,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, + .digital_mute = tas571x_mute, }; static const char *const tas5711_supply_names[] = { @@ -241,6 +266,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -260,6 +305,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -314,6 +361,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ @@ -326,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = { .vol_reg_size = 2, }; +static const char *const tas5721_supply_names[] = { + "AVDD", + "DVDD", + "DRVDD", + "PVDD", +}; + +static const struct snd_kcontrol_new tas5721_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), +}; + +static const struct reg_default tas5721_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x00}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_CH3_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_PWM_CH_SDN_GROUP_REG, 0x30}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_CH4_SRC_SELECT_REG, 0x4303}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5721_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5721_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5721_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, +}; + + +static const struct tas571x_chip tas5721_chip = { + .supply_names = tas5721_supply_names, + .num_supply_names = ARRAY_SIZE(tas5721_supply_names), + .controls = tas5711_controls, + .num_controls = ARRAY_SIZE(tas5711_controls), + .regmap_config = &tas5721_regmap_config, + .vol_reg_size = 1, +}; + static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), @@ -386,11 +506,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, priv); of_id = of_match_device(tas571x_of_match, dev); - if (!of_id) { - dev_err(dev, "Unknown device type\n"); - return -EINVAL; - } - priv->chip = of_id->data; + if (of_id) + priv->chip = of_id->data; + else + priv->chip = (void *) id->driver_data; priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) { @@ -445,10 +564,6 @@ static int tas571x_i2c_probe(struct i2c_client *client, if (ret) return ret; - ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG, - TAS571X_SYS_CTRL_2_SDN_MASK, 0); - if (ret) - return ret; memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver)); priv->codec_driver.controls = priv->chip->controls; @@ -486,14 +601,16 @@ static const struct of_device_id tas571x_of_match[] = { { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, + { .compatible = "ti,tas5721", .data = &tas5721_chip, }, { } }; MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { - { "tas5711", 0 }, - { "tas5717", 0 }, - { "tas5719", 0 }, + { "tas5711", (kernel_ulong_t) &tas5711_chip }, + { "tas5717", (kernel_ulong_t) &tas5717_chip }, + { "tas5719", (kernel_ulong_t) &tas5717_chip }, + { "tas5721", (kernel_ulong_t) &tas5721_chip }, { } }; MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id); diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c new file mode 100644 index 000000000000..f54fb46b77c2 --- /dev/null +++ b/sound/soc/codecs/tas5720.c @@ -0,0 +1,620 @@ +/* + * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg <dannenberg@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/regulator/consumer.h> +#include <linux/delay.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tas5720.h" + +/* Define how often to check (and clear) the fault status register (in ms) */ +#define TAS5720_FAULT_CHECK_INTERVAL 200 + +static const char * const tas5720_supply_names[] = { + "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ + "pvdd", /* Class-D amp and analog power supply (connected). */ +}; + +#define TAS5720_NUM_SUPPLIES ARRAY_SIZE(tas5720_supply_names) + +struct tas5720_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas5720_client; + struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; + struct delayed_work fault_check_work; + unsigned int last_fault; +}; + +static int tas5720_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int rate = params_rate(params); + bool ssz_ds; + int ret; + + switch (rate) { + case 44100: + case 48000: + ssz_ds = false; + break; + case 88200: + case 96000: + ssz_ds = true; + break; + default: + dev_err(codec->dev, "unsupported sample rate: %u\n", rate); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SSZ_DS, ssz_ds); + if (ret < 0) { + dev_err(codec->dev, "error setting sample rate: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + int ret; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): + /* 1st data bit occur one BCLK cycle after the frame sync */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF): + /* + * Note that although the TAS5720 does not have a dedicated DSP + * mode it doesn't care about the LRCLK duty cycle during TDM + * operation. Therefore we can use the device's I2S mode with + * its delaying of the 1st data bit to receive DSP_A formatted + * data. See device datasheet for additional details. + */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF): + /* + * Similar to DSP_A, we can use the fact that the TAS5720 does + * not care about the LRCLK duty cycle during TDM to receive + * DSP_B formatted data in LEFTJ mode (no delaying of the 1st + * data bit). + */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): + /* No delay after the frame sync */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SAIF_FORMAT_MASK, + serial_format); + if (ret < 0) { + dev_err(codec->dev, "error setting SAIF format: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int first_slot; + int ret; + + if (!tx_mask) { + dev_err(codec->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + /* + * Determine the first slot that is being requested. We will only + * use the first slot that is found since the TAS5720 is a mono + * amplifier. + */ + first_slot = __ffs(tx_mask); + + if (first_slot > 7) { + dev_err(codec->dev, "slot selection out of bounds (%u)\n", + first_slot); + return -EINVAL; + } + + /* Enable manual TDM slot selection (instead of I2C ID based) */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* Configure the TDM slot to process audio from */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_TDM_SLOT_SEL_MASK, first_slot); + if (ret < 0) + goto error_snd_soc_update_bits; + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring TDM mode: %d\n", ret); + return ret; +} + +static int tas5720_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int ret; + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, mute ? TAS5720_MUTE : 0); + if (ret < 0) { + dev_err(codec->dev, "error (un-)muting device: %d\n", ret); + return ret; + } + + return 0; +} + +static void tas5720_fault_check_work(struct work_struct *work) +{ + struct tas5720_data *tas5720 = container_of(work, struct tas5720_data, + fault_check_work.work); + struct device *dev = tas5720->codec->dev; + unsigned int curr_fault; + int ret; + + ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault); + if (ret < 0) { + dev_err(dev, "failed to read FAULT register: %d\n", ret); + goto out; + } + + /* Check/handle all errors except SAIF clock errors */ + curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE; + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS5720 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE)) + dev_crit(dev, "experienced an over current hardware fault\n"); + + if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE)) + dev_crit(dev, "experienced a DC detection fault\n"); + + if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE)) + dev_crit(dev, "experienced an over temperature fault\n"); + + /* Store current fault value so we can detect any changes next time */ + tas5720->last_fault = curr_fault; + + if (!curr_fault) + goto out; + + /* + * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching + * faults as long as a fault condition persists. Always going through + * the full sequence no matter the first return value to minimizes + * chances for the device to end up in shutdown mode. + */ + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + +out: + /* Schedule the next fault check at the specified interval */ + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); +} + +static int tas5720_codec_probe(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + unsigned int device_id; + int ret; + + tas5720->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret != 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); + if (ret < 0) { + dev_err(codec->dev, "failed to read device ID register: %d\n", + ret); + goto probe_fail; + } + + if (device_id != TAS5720_DEVICE_ID) { + dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", + TAS5720_DEVICE_ID, device_id); + ret = -ENODEV; + goto probe_fail; + } + + /* Set device to mute */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, TAS5720_MUTE); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* + * Enter shutdown mode - our default when not playing audio - to + * minimize current consumption. On the TAS5720 there is no real down + * side doing so as all device registers are preserved and the wakeup + * of the codec is rather quick which we do using a dapm widget. + */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + goto error_snd_soc_update_bits; + + INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work); + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring device registers: %d\n", ret); + +probe_fail: + regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + return ret; +} + +static int tas5720_codec_remove(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + cancel_delayed_work_sync(&tas5720->fault_check_work); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +}; + +static int tas5720_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Take TAS5720 out of shutdown mode */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) { + dev_err(codec->dev, "error waking codec: %d\n", ret); + return ret; + } + + /* + * Observe codec shutdown-to-active time. The datasheet only + * lists a nominal value however just use-it as-is without + * additional padding to minimize the delay introduced in + * starting to play audio (actually there is other setup done + * by the ASoC framework that will provide additional delays, + * so we should always be safe). + */ + msleep(25); + + /* Turn on TAS5720 periodic fault checking/handling */ + tas5720->last_fault = 0; + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); + } else if (event & SND_SOC_DAPM_PRE_PMD) { + /* Disable TAS5720 periodic fault checking/handling */ + cancel_delayed_work_sync(&tas5720->fault_check_work); + + /* Place TAS5720 in shutdown mode to minimize current draw */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) { + dev_err(codec->dev, "error shutting down codec: %d\n", + ret); + return ret; + } + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas5720_suspend(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + regcache_cache_only(tas5720->regmap, true); + regcache_mark_dirty(tas5720->regmap); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +} + +static int tas5720_resume(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(tas5720->regmap, false); + + ret = regcache_sync(tas5720->regmap); + if (ret < 0) { + dev_err(codec->dev, "failed to sync regcache: %d\n", ret); + return ret; + } + + return 0; +} +#else +#define tas5720_suspend NULL +#define tas5720_resume NULL +#endif + +static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5720_DEVICE_ID_REG: + case TAS5720_FAULT_REG: + return true; + default: + return false; + } +} + +static const struct regmap_config tas5720_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5720_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, +}; + +/* + * DAC analog gain. There are four discrete values to select from, ranging + * from 19.2 dB to 26.3dB. + */ +static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, + 0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0), + 0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0), + 0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0), + 0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0), +); + +/* + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that + * setting the gain below -100 dB (register value <0x7) is effectively a MUTE + * as per device datasheet. + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); + +static const struct snd_kcontrol_new tas5720_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT") +}; + +static const struct snd_soc_dapm_route tas5720_audio_map[] = { + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5720 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + + .controls = tas5720_snd_controls, + .num_controls = ARRAY_SIZE(tas5720_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), +}; + +/* PCM rates supported by the TAS5720 driver */ +#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +/* Formats supported by TAS5720 driver */ +#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { + .hw_params = tas5720_hw_params, + .set_fmt = tas5720_set_dai_fmt, + .set_tdm_slot = tas5720_set_dai_tdm_slot, + .digital_mute = tas5720_mute, +}; + +/* + * TAS5720 DAI structure + * + * Note that were are advertising .playback.channels_max = 2 despite this being + * a mono amplifier. The reason for that is that some serial ports such as TI's + * McASP module have a minimum number of channels (2) that they can output. + * Advertising more channels than we have will allow us to interface with such + * a serial port without really any negative side effects as the TAS5720 will + * simply ignore any extra channel(s) asides from the one channel that is + * configured to be played back. + */ +static struct snd_soc_dai_driver tas5720_dai[] = { + { + .name = "tas5720-amplifier", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TAS5720_RATES, + .formats = TAS5720_FORMATS, + }, + .ops = &tas5720_speaker_dai_ops, + }, +}; + +static int tas5720_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas5720_data *data; + int ret; + int i; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->tas5720_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas5720_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + dev_set_drvdata(dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas5720, + tas5720_dai, ARRAY_SIZE(tas5720_dai)); + if (ret < 0) { + dev_err(dev, "failed to register codec: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_remove(struct i2c_client *client) +{ + struct device *dev = &client->dev; + + snd_soc_unregister_codec(dev); + + return 0; +} + +static const struct i2c_device_id tas5720_id[] = { + { "tas5720", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5720_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas5720_of_match[] = { + { .compatible = "ti,tas5720", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas5720_of_match); +#endif + +static struct i2c_driver tas5720_i2c_driver = { + .driver = { + .name = "tas5720", + .of_match_table = of_match_ptr(tas5720_of_match), + }, + .probe = tas5720_probe, + .remove = tas5720_remove, + .id_table = tas5720_id, +}; + +module_i2c_driver(tas5720_i2c_driver); + +MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>"); +MODULE_DESCRIPTION("TAS5720 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h new file mode 100644 index 000000000000..3d077c779b12 --- /dev/null +++ b/sound/soc/codecs/tas5720.h @@ -0,0 +1,90 @@ +/* + * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg <dannenberg@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * 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. + */ + +#ifndef __TAS5720_H__ +#define __TAS5720_H__ + +/* Register Address Map */ +#define TAS5720_DEVICE_ID_REG 0x00 +#define TAS5720_POWER_CTRL_REG 0x01 +#define TAS5720_DIGITAL_CTRL1_REG 0x02 +#define TAS5720_DIGITAL_CTRL2_REG 0x03 +#define TAS5720_VOLUME_CTRL_REG 0x04 +#define TAS5720_ANALOG_CTRL_REG 0x06 +#define TAS5720_FAULT_REG 0x08 +#define TAS5720_DIGITAL_CLIP2_REG 0x10 +#define TAS5720_DIGITAL_CLIP1_REG 0x11 +#define TAS5720_MAX_REG TAS5720_DIGITAL_CLIP1_REG + +/* TAS5720_DEVICE_ID_REG */ +#define TAS5720_DEVICE_ID 0x01 + +/* TAS5720_POWER_CTRL_REG */ +#define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) +#define TAS5720_SLEEP BIT(1) +#define TAS5720_SDZ BIT(0) + +/* TAS5720_DIGITAL_CTRL1_REG */ +#define TAS5720_HPF_BYPASS BIT(7) +#define TAS5720_TDM_CFG_SRC BIT(6) +#define TAS5720_SSZ_DS BIT(3) +#define TAS5720_SAIF_RIGHTJ_24BIT (0x0) +#define TAS5720_SAIF_RIGHTJ_20BIT (0x1) +#define TAS5720_SAIF_RIGHTJ_18BIT (0x2) +#define TAS5720_SAIF_RIGHTJ_16BIT (0x3) +#define TAS5720_SAIF_I2S (0x4) +#define TAS5720_SAIF_LEFTJ (0x5) +#define TAS5720_SAIF_FORMAT_MASK GENMASK(2, 0) + +/* TAS5720_DIGITAL_CTRL2_REG */ +#define TAS5720_MUTE BIT(4) +#define TAS5720_TDM_SLOT_SEL_MASK GENMASK(2, 0) + +/* TAS5720_ANALOG_CTRL_REG */ +#define TAS5720_PWM_RATE_6_3_FSYNC (0x0 << 4) +#define TAS5720_PWM_RATE_8_4_FSYNC (0x1 << 4) +#define TAS5720_PWM_RATE_10_5_FSYNC (0x2 << 4) +#define TAS5720_PWM_RATE_12_6_FSYNC (0x3 << 4) +#define TAS5720_PWM_RATE_14_7_FSYNC (0x4 << 4) +#define TAS5720_PWM_RATE_16_8_FSYNC (0x5 << 4) +#define TAS5720_PWM_RATE_20_10_FSYNC (0x6 << 4) +#define TAS5720_PWM_RATE_24_12_FSYNC (0x7 << 4) +#define TAS5720_PWM_RATE_MASK GENMASK(6, 4) +#define TAS5720_ANALOG_GAIN_19_2DBV (0x0 << 2) +#define TAS5720_ANALOG_GAIN_20_7DBV (0x1 << 2) +#define TAS5720_ANALOG_GAIN_23_5DBV (0x2 << 2) +#define TAS5720_ANALOG_GAIN_26_3DBV (0x3 << 2) +#define TAS5720_ANALOG_GAIN_MASK GENMASK(3, 2) +#define TAS5720_ANALOG_GAIN_SHIFT (0x2) + +/* TAS5720_FAULT_REG */ +#define TAS5720_OC_THRESH_100PCT (0x0 << 4) +#define TAS5720_OC_THRESH_75PCT (0x1 << 4) +#define TAS5720_OC_THRESH_50PCT (0x2 << 4) +#define TAS5720_OC_THRESH_25PCT (0x3 << 4) +#define TAS5720_OC_THRESH_MASK GENMASK(5, 4) +#define TAS5720_CLKE BIT(3) +#define TAS5720_OCE BIT(2) +#define TAS5720_DCE BIT(1) +#define TAS5720_OTE BIT(0) +#define TAS5720_FAULT_MASK GENMASK(3, 0) + +/* TAS5720_DIGITAL_CLIP1_REG */ +#define TAS5720_CLIP1_MASK GENMASK(7, 2) +#define TAS5720_CLIP1_SHIFT (0x2) + +#endif /* __TAS5720_H__ */ diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index ee4def4f819f..3c5e1df01c19 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -28,6 +28,7 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/acpi.h> #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/slab.h> @@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +#ifdef CONFIG_ACPI +static const struct acpi_device_id aic31xx_acpi_match[] = { + { "10TI3100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); +#endif + static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, .remove = aic31xx_i2c_remove, diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c new file mode 100644 index 000000000000..59606cf3008f --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -0,0 +1,74 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-i2c.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond <nh6z@nh6z.net> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * 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. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "tlv320aic32x4.h" + +static int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 8; + config.val_bits = 8; + + regmap = devm_regmap_init_i2c(i2c, &config); + return aic32x4_probe(&i2c->dev, regmap); +} + +static int aic32x4_i2c_remove(struct i2c_client *i2c) +{ + return aic32x4_remove(&i2c->dev); +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_i2c_probe, + .remove = aic32x4_i2c_remove, + .id_table = aic32x4_i2c_id, +}; + +module_i2c_driver(aic32x4_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C"); +MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c new file mode 100644 index 000000000000..724fcdd491b2 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -0,0 +1,76 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-spi.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond <nh6z@nh6z.net> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * 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. + */ + +#include <linux/spi/spi.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "tlv320aic32x4.h" + +static int aic32x4_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 7; + config.pad_bits = 1; + config.val_bits = 8; + config.read_flag_mask = 0x01; + + regmap = devm_regmap_init_spi(spi, &config); + return aic32x4_probe(&spi->dev, regmap); +} + +static int aic32x4_spi_remove(struct spi_device *spi) +{ + return aic32x4_remove(&spi->dev); +} + +static const struct spi_device_id aic32x4_spi_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct spi_driver aic32x4_spi_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_spi_probe, + .remove = aic32x4_spi_remove, + .id_table = aic32x4_spi_id, +}; + +module_spi_driver(aic32x4_spi_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI"); +MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f2d3191961e1..85d4978d0384 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -30,7 +30,6 @@ #include <linux/pm.h> #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <linux/i2c.h> #include <linux/cdev.h> #include <linux/slab.h> #include <linux/clk.h> @@ -160,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -181,16 +183,71 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = { SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), }; -static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +static const char * const resistor_text[] = { + "Off", "10 kOhm", "20 kOhm", "40 kOhm", }; -static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +/* Left mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text); + +static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum), +}; +static const struct snd_kcontrol_new cml_to_lmixer_controls[] = { + SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum), +}; +static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum), +}; +static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), +}; + +/* Right mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text); +static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum), +}; +static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = { + SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum), +}; +static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum), +}; +static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -214,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, - &left_input_mixer_controls[0], - ARRAY_SIZE(left_input_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, - &right_input_mixer_controls[0], - ARRAY_SIZE(right_input_mixer_controls)), - SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_rmixer_controls), + SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cmr_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_rmixer_controls), + + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_lmixer_controls), + SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cml_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_lmixer_controls), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), @@ -261,19 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, - /* Left input */ - {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, - {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, - {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, - - {"Left ADC", NULL, "Left Input Mixer"}, - /* Right Input */ - {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, - {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, - {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, - - {"Right ADC", NULL, "Right Input Mixer"}, + {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"}, + {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"}, + {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"}, + + {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"}, + {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"}, + + {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"}, + {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"}, + {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"}, + + {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"}, + {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"}, + + {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"}, + {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"}, + + /* Left Input */ + {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"}, + {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"}, + + {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"}, + {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"}, + {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"}, + + {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"}, + {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"}, + {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"}, + + {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"}, + {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"}, + + {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"}, + {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"}, }; static const struct regmap_range_cfg aic32x4_regmap_pages[] = { @@ -287,14 +427,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { }, }; -static const struct regmap_config aic32x4_regmap = { - .reg_bits = 8, - .val_bits = 8, - +const struct regmap_config aic32x4_regmap_config = { .max_register = AIC32X4_RMICPGAVOL, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; +EXPORT_SYMBOL(aic32x4_regmap_config); static inline int aic32x4_get_divs(int mclk, int rate) { @@ -567,7 +705,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -596,7 +734,7 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_probe(struct snd_soc_codec *codec) +static int aic32x4_codec_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; @@ -655,7 +793,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .probe = aic32x4_probe, + .probe = aic32x4_codec_probe, .set_bias_level = aic32x4_set_bias_level, .suspend_bias_off = true, @@ -777,24 +915,22 @@ error_ldo: return ret; } -static int aic32x4_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +int aic32x4_probe(struct device *dev, struct regmap *regmap) { - struct aic32x4_pdata *pdata = i2c->dev.platform_data; struct aic32x4_priv *aic32x4; - struct device_node *np = i2c->dev.of_node; + struct aic32x4_pdata *pdata = dev->platform_data; + struct device_node *np = dev->of_node; int ret; - aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv), GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; - aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); - if (IS_ERR(aic32x4->regmap)) - return PTR_ERR(aic32x4->regmap); - - i2c_set_clientdata(i2c, aic32x4); + dev_set_drvdata(dev, aic32x4); if (pdata) { aic32x4->power_cfg = pdata->power_cfg; @@ -804,7 +940,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, } else if (np) { ret = aic32x4_parse_dt(aic32x4, np); if (ret) { - dev_err(&i2c->dev, "Failed to parse DT node\n"); + dev_err(dev, "Failed to parse DT node\n"); return ret; } } else { @@ -814,71 +950,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk"); + aic32x4->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(aic32x4->mclk)) { - dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); return PTR_ERR(aic32x4->mclk); } if (gpio_is_valid(aic32x4->rstn_gpio)) { - ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) return ret; } - ret = aic32x4_setup_regulators(&i2c->dev, aic32x4); + ret = aic32x4_setup_regulators(dev, aic32x4); if (ret) { - dev_err(&i2c->dev, "Failed to setup regulators\n"); + dev_err(dev, "Failed to setup regulators\n"); return ret; } - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); if (ret) { - dev_err(&i2c->dev, "Failed to register codec\n"); + dev_err(dev, "Failed to register codec\n"); aic32x4_disable_regulators(aic32x4); return ret; } - i2c_set_clientdata(i2c, aic32x4); - return 0; } +EXPORT_SYMBOL(aic32x4_probe); -static int aic32x4_i2c_remove(struct i2c_client *client) +int aic32x4_remove(struct device *dev) { - struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client); + struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev); aic32x4_disable_regulators(aic32x4); - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(dev); + return 0; } - -static const struct i2c_device_id aic32x4_i2c_id[] = { - { "tlv320aic32x4", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); - -static const struct of_device_id aic32x4_of_id[] = { - { .compatible = "ti,tlv320aic32x4", }, - { /* senitel */ } -}; -MODULE_DEVICE_TABLE(of, aic32x4_of_id); - -static struct i2c_driver aic32x4_i2c_driver = { - .driver = { - .name = "tlv320aic32x4", - .of_match_table = aic32x4_of_id, - }, - .probe = aic32x4_i2c_probe, - .remove = aic32x4_i2c_remove, - .id_table = aic32x4_i2c_id, -}; - -module_i2c_driver(aic32x4_i2c_driver); +EXPORT_SYMBOL(aic32x4_remove); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 995f033a855d..a197dd51addc 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -10,6 +10,13 @@ #ifndef _TLV320AIC32X4_H #define _TLV320AIC32X4_H +struct device; +struct regmap_config; + +extern const struct regmap_config aic32x4_regmap_config; +int aic32x4_probe(struct device *dev, struct regmap *regmap); +int aic32x4_remove(struct device *dev); + /* tlv320aic32x4 register space (in decimal to match datasheet) */ #define AIC32X4_PAGE1 128 diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..1f7081043566 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, @@ -983,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i if (mute) { /* Power down drivers and DACs */ hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); } twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 171a23ddd15d..512a9d25fe6f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -17,6 +17,7 @@ #include <linux/export.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/pm_runtime.h> @@ -2236,14 +2237,9 @@ static irqreturn_t wm5100_edge_irq(int irq, void *data) } #ifdef CONFIG_GPIOLIB -static inline struct wm5100_priv *gpio_to_wm5100(struct gpio_chip *chip) -{ - return container_of(chip, struct wm5100_priv, gpio_chip); -} - static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_LVL, !!value << WM5100_GP1_LVL_SHIFT); @@ -2252,7 +2248,7 @@ static void wm5100_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm5100_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); int val, ret; val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT); @@ -2268,7 +2264,7 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip, static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2281,7 +2277,7 @@ static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm5100_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm5100_priv *wm5100 = gpio_to_wm5100(chip); + struct wm5100_priv *wm5100 = gpiochip_get_data(chip); return regmap_update_bits(wm5100->regmap, WM5100_GPIO_CTRL_1 + offset, WM5100_GP1_FN_MASK | WM5100_GP1_DIR, @@ -2313,7 +2309,7 @@ static void wm5100_init_gpio(struct i2c_client *i2c) else wm5100->gpio_chip.base = -1; - ret = gpiochip_add(&wm5100->gpio_chip); + ret = gpiochip_add_data(&wm5100->gpio_chip, wm5100); if (ret != 0) dev_err(&i2c->dev, "Failed to add GPIOs: %d\n", ret); } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da60e3fe5ee7..e7fe6b7b95b7 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1872,7 +1872,7 @@ static struct snd_soc_dai_driver wm5102_dai[] = { .capture = { .stream_name = "Audio Trace CPU", .channels_min = 1, - .channels_max = 6, + .channels_max = 4, .rates = WM5102_RATES, .formats = WM5102_FORMATS, }, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b5820e4d5471..d54f1b46c9ec 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1723,6 +1723,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "OUT2L", NULL, "SYSCLK" }, { "OUT2R", NULL, "SYSCLK" }, { "OUT3L", NULL, "SYSCLK" }, + { "OUT3R", NULL, "SYSCLK" }, { "OUT4L", NULL, "SYSCLK" }, { "OUT4R", NULL, "SYSCLK" }, { "OUT5L", NULL, "SYSCLK" }, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a82b8bc2cfc0..a26ca490cf31 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -20,7 +20,7 @@ #include <linux/init.h> #include <linux/completion.h> #include <linux/delay.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -1766,11 +1766,6 @@ static int wm8903_resume(struct snd_soc_codec *codec) } #ifdef CONFIG_GPIOLIB -static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8903_priv, gpio_chip); -} - static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) { if (offset >= WM8903_NUM_GPIO) @@ -1781,7 +1776,7 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1799,7 +1794,7 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int reg; regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, ®); @@ -1810,7 +1805,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); unsigned int mask, val; int ret; @@ -1828,7 +1823,7 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct wm8903_priv *wm8903 = gpiochip_get_data(chip); regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, WM8903_GP1_LVL_MASK, @@ -1860,7 +1855,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) else wm8903->gpio_chip.base = -1; - ret = gpiochip_add(&wm8903->gpio_chip); + ret = gpiochip_add_data(&wm8903->gpio_chip, wm8903); if (ret != 0) dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index f6f9395ea38e..1c600819f768 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -743,6 +743,7 @@ static const struct regmap_config wm8940_regmap = { .max_register = WM8940_MONOMIX, .reg_defaults = wm8940_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8940_reg_defaults), + .cache_type = REGCACHE_RBTREE, .readable_reg = wm8940_readable_register, .volatile_reg = wm8940_volatile_register, diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 720a14e0687d..f3109da24769 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -18,7 +18,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> -#include <linux/gpio.h> +#include <linux/gpio/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/pm_runtime.h> @@ -3307,14 +3307,9 @@ static void wm8962_set_gpio_mode(struct wm8962_priv *wm8962, int gpio) } #ifdef CONFIG_GPIOLIB -static inline struct wm8962_priv *gpio_to_wm8962(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8962_priv, gpio_chip); -} - static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); /* The WM8962 GPIOs aren't linearly numbered. For simplicity * we export linear numbers and error out if the unsupported @@ -3337,7 +3332,7 @@ static int wm8962_gpio_request(struct gpio_chip *chip, unsigned offset) static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, @@ -3347,7 +3342,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8962_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8962_priv *wm8962 = gpio_to_wm8962(chip); + struct wm8962_priv *wm8962 = gpiochip_get_data(chip); struct snd_soc_codec *codec = wm8962->codec; int ret, val; @@ -3386,7 +3381,7 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) else wm8962->gpio_chip.base = -1; - ret = gpiochip_add(&wm8962->gpio_chip); + ret = gpiochip_add_data(&wm8962->gpio_chip, wm8962); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); } @@ -3798,9 +3793,8 @@ static int wm8962_runtime_resume(struct device *dev) ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (ret != 0) { - dev_err(dev, - "Failed to enable supplies: %d\n", ret); - return ret; + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto disable_clock; } regcache_cache_only(wm8962->regmap, false); @@ -3838,6 +3832,10 @@ static int wm8962_runtime_resume(struct device *dev) msleep(5); return 0; + +disable_clock: + clk_disable_unprepare(wm8962->pdata.mclk); + return ret; } static int wm8962_runtime_suspend(struct device *dev) diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h index 910aafd09d21..e63a318a3015 100644 --- a/sound/soc/codecs/wm8962.h +++ b/sound/soc/codecs/wm8962.h @@ -16,9 +16,9 @@ #include <asm/types.h> #include <sound/soc.h> -#define WM8962_SYSCLK_MCLK 1 -#define WM8962_SYSCLK_FLL 2 -#define WM8962_SYSCLK_PLL3 3 +#define WM8962_SYSCLK_MCLK 0 +#define WM8962_SYSCLK_FLL 1 +#define WM8962_SYSCLK_PLL3 2 #define WM8962_FLL 1 diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f99b34f7647b..a73044251218 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/gpio/driver.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -2139,14 +2140,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } #ifdef CONFIG_GPIOLIB -static inline struct wm8996_priv *gpio_to_wm8996(struct gpio_chip *chip) -{ - return container_of(chip, struct wm8996_priv, gpio_chip); -} - static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_LVL, !!value << WM8996_GP1_LVL_SHIFT); @@ -2155,7 +2151,7 @@ static void wm8996_gpio_set(struct gpio_chip *chip, unsigned offset, int value) static int wm8996_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); int val; val = (1 << WM8996_GP1_FN_SHIFT) | (!!value << WM8996_GP1_LVL_SHIFT); @@ -2167,7 +2163,7 @@ static int wm8996_gpio_direction_out(struct gpio_chip *chip, static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); unsigned int reg; int ret; @@ -2180,7 +2176,7 @@ static int wm8996_gpio_get(struct gpio_chip *chip, unsigned offset) static int wm8996_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { - struct wm8996_priv *wm8996 = gpio_to_wm8996(chip); + struct wm8996_priv *wm8996 = gpiochip_get_data(chip); return regmap_update_bits(wm8996->regmap, WM8996_GPIO_1 + offset, WM8996_GP1_FN_MASK | WM8996_GP1_DIR, @@ -2211,7 +2207,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) else wm8996->gpio_chip.base = -1; - ret = gpiochip_add(&wm8996->gpio_chip); + ret = gpiochip_add_data(&wm8996->gpio_chip, wm8996); if (ret != 0) dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret); } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0f66fda2c772..237dc67002ef 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1513,8 +1513,9 @@ static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { - .tx_dma_offset = 0x200, - .rx_dma_offset = 0x284, + /* The CFG port offset will be calculated if it is needed */ + .tx_dma_offset = 0, + .rx_dma_offset = 0, .version = MCASP_VERSION_4, }; @@ -1734,6 +1735,52 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_EDMA; } +static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->tx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == TX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_TXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + +static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->rx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == RX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_RXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1862,7 +1909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->tx_dma_offset; + dma_data->addr = mem->start + davinci_mcasp_txdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -1883,7 +1930,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->rx_dma_offset; + dma_data->addr = + mem->start + davinci_mcasp_rxdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 1e8787fb3fb7..afddc8010c54 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -85,9 +85,9 @@ (n << 2)) /* Transmit Buffer for Serializer n */ -#define DAVINCI_MCASP_TXBUF_REG 0x200 +#define DAVINCI_MCASP_TXBUF_REG(n) (0x200 + (n << 2)) /* Receive Buffer for Serializer n */ -#define DAVINCI_MCASP_RXBUF_REG 0x280 +#define DAVINCI_MCASP_RXBUF_REG(n) (0x280 + (n << 2)) /* McASP FIFO Registers */ #define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 632ecc0e3956..bedec4a32581 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -952,16 +952,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, ssi_private->i2s_mode = CCSR_SSI_SCR_NET; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + regmap_update_bits(regs, CCSR_SSI_STCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); + regmap_update_bits(regs, CCSR_SSI_SRCCR, + CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(2)); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFS: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_MASTER; - regmap_update_bits(regs, CCSR_SSI_STCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); - regmap_update_bits(regs, CCSR_SSI_SRCCR, - CCSR_SSI_SxCCR_DC_MASK, - CCSR_SSI_SxCCR_DC(2)); break; case SND_SOC_DAIFMT_CBM_CFM: ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2389ab47e25f..466492b7d4f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, diff --git a/sound/soc/intel/atom/sst-mfld-platform-compress.c b/sound/soc/intel/atom/sst-mfld-platform-compress.c index 395168986462..1bead81bb510 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-compress.c +++ b/sound/soc/intel/atom/sst-mfld-platform-compress.c @@ -182,24 +182,29 @@ static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) case SNDRV_PCM_TRIGGER_START: if (stream->compr_ops->stream_start) return stream->compr_ops->stream_start(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_STOP: if (stream->compr_ops->stream_drop) return stream->compr_ops->stream_drop(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_DRAIN: if (stream->compr_ops->stream_drain) return stream->compr_ops->stream_drain(sst->dev, stream->id); + break; case SND_COMPR_TRIGGER_PARTIAL_DRAIN: if (stream->compr_ops->stream_partial_drain) return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (stream->compr_ops->stream_pause) return stream->compr_ops->stream_pause(sst->dev, stream->id); + break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (stream->compr_ops->stream_pause_release) return stream->compr_ops->stream_pause_release(sst->dev, stream->id); - default: - return -EINVAL; + break; } + return -EINVAL; } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 6260df6bd49c..cdcced9f32b6 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -296,7 +296,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (!drv) return -ENOMEM; - drv->ts3a227e_present = acpi_dev_present("104C227E"); + drv->ts3a227e_present = acpi_dev_found("104C227E"); if (!drv->ts3a227e_present) { /* no need probe TI jack detection chip */ snd_soc_card_cht.aux_dev = NULL; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 0618a7f1025b..d7ef292c402d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) return -ENOMEM; for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) { - if (acpi_dev_present(snd_soc_cards[i].codec_id)) { + if (acpi_dev_found(snd_soc_cards[i].codec_id)) { dev_dbg(&pdev->dev, "found codec %s\n", snd_soc_cards[i].codec_id); card = snd_soc_cards[i].soc_card; diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index ef4881e7753a..25993527370b 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -203,7 +203,7 @@ static struct dw_dma_chip *dw_probe(struct device *dev, struct resource *mem, chip->dev = dev; - err = dw_dma_probe(chip, NULL); + err = dw_dma_probe(chip); if (err) return ERR_PTR(err); diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 965ce40ce752..8b95e09e23e8 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -291,6 +291,7 @@ int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + INIT_LIST_HEAD(&sst->module_list); ret = skl_ipc_init(dev, skl); if (ret) return ret; diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 132bb83f8e99..bc3c7b5ac752 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,7 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST + depends on HAS_DMA help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index f7e789e97fbc..3abf51c07851 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676 depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 + select SND_SOC_HDMI_CODEC help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5676 codecs. diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 5c4c58c69c51..bb593926c62d 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = { enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, + DAI_LINK_HDMI, DAI_LINK_CODEC_I2S, + DAI_LINK_HDMI_I2S, DAI_LINK_INTERCODEC }; @@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dynamic = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI] = { + .name = "HDMI", + .stream_name = "HDMI PCM", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, /* Back End DAI links */ [DAI_LINK_CODEC_I2S] = { @@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI_I2S] = { + .name = "HDMI BE", + .cpu_dai_name = "HDMIO", + .no_pcm = 1, + .codec_dai_name = "i2s-hifi", + .dpcm_playback = 1, + }, /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */ [DAI_LINK_INTERCODEC] = { .name = "rt5650_rt5676 intercodec", @@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node = mt8173_rt5650_rt5676_codecs[1].of_node; + mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2); + if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1b7f1c..a27a6673dbe3 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2c12fb..2b5df2ef51a3 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -123,6 +123,7 @@ #define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8) #define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4) #define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3) +#define AFE_TDM_CON1_LRCK_INV (0x1 << 2) #define AFE_TDM_CON1_BCK_INV (0x1 << 1) #define AFE_TDM_CON1_EN (0x1 << 0) @@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, runtime->rate * runtime->channels * 32); val = AFE_TDM_CON1_BCK_INV | + AFE_TDM_CON1_LRCK_INV | AFE_TDM_CON1_1_BCK_DELAY | AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */ AFE_TDM_CON1_WLEN_32BIT | diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index c7563e230c7d..4a16e778966b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp) if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 1); + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + /* Enable McBSP Sidetone */ w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); @@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp) w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 0); } diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 99381a27295b..a84f677234f0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_chan *chan; int err = 0; + memset(&config, 0x00, sizeof(config)); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index ec522e94b0e2..b6cb9950f05d 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -133,3 +133,4 @@ module_platform_driver(mmp_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("ALSA SoC Brownstone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 5c8f9db50a47..d1661fa6ee08 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver); MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 51e790d006f5..96df9b2d8fc4 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("MMP Soc Audio DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index eca60c29791a..ca8b23f8c525 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("MMP SSPA SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4e74d9573f03..bcc81e920a67 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver); MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index da03fad1b9cd..3cad990dad2c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f3de615aacd7..9615e6de1306 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 9f390398d518..410d48b93031 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 6e8665430bd5..db000c6987a1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -474,7 +474,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) struct lpass_data *drvdata = snd_soc_platform_get_drvdata(soc_runtime->platform); struct lpass_variant *v = drvdata->variant; - int ret; + int ret = -EINVAL; struct lpass_pcm_data *data; size_t size = lpass_platform_pcm_hardware.buffer_bytes_max; @@ -491,7 +491,7 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->rdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR_VALUE(data->rdma_ch)) + if (data->rdma_ch < 0) return data->rdma_ch; drvdata->substream[data->rdma_ch] = psubstream; @@ -518,8 +518,10 @@ static int lpass_platform_pcm_new(struct snd_soc_pcm_runtime *soc_runtime) data->wrdma_ch = v->alloc_dma_channel(drvdata, SNDRV_PCM_STREAM_CAPTURE); - if (IS_ERR_VALUE(data->wrdma_ch)) + if (data->wrdma_ch < 0) { + ret = data->wrdma_ch; goto capture_alloc_err; + } drvdata->substream[data->wrdma_ch] = csubstream; diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0891014c262f..c4c51a4d3c8f 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8fd7bdc..6bc93cbb3049 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67258ca..a8f61d79333b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 15d6ffe8be74..e39f916d0f2f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv) i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (ret) goto rsnd_src_probe_done; +skip: i++; } diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 7e0acd83b0e6..bc4a55bb3fd9 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -59,8 +59,7 @@ static void soc_ac97_device_release(struct device *dev) #ifdef CONFIG_GPIOLIB static inline struct snd_soc_codec *gpio_to_codec(struct gpio_chip *chip) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); return gpio_priv->codec; } @@ -98,8 +97,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { - struct snd_ac97_gpio_priv *gpio_priv = - container_of(chip, struct snd_ac97_gpio_priv, gpio_chip); + struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); struct snd_soc_codec *codec = gpio_to_codec(chip); gpio_priv->gpios_set &= ~(1 << offset); @@ -145,7 +143,7 @@ static int snd_soc_ac97_init_gpio(struct snd_ac97 *ac97, gpio_priv->gpio_chip.parent = codec->dev; gpio_priv->gpio_chip.base = -1; - ret = gpiochip_add(&gpio_priv->gpio_chip); + ret = gpiochip_add_data(&gpio_priv->gpio_chip, gpio_priv); if (ret != 0) dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); return ret; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1cf94d7fb9f4..ee7f15aa46fc 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1476,6 +1481,8 @@ widget: widget->dobj.type = SND_SOC_DOBJ_WIDGET; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; + kfree(template.sname); + kfree(template.name); list_add(&widget->dobj.list, &tplg->comp->dobj_list); return 0; @@ -1499,10 +1506,17 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; @@ -1586,6 +1600,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return snd_soc_register_dai(tplg->comp, dai_drv); } +/* create the FE DAI link */ static int soc_tplg_link_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1598,6 +1613,16 @@ static int soc_tplg_link_create(struct soc_tplg *tplg, link->name = pcm->pcm_name; link->stream_name = pcm->pcm_name; + link->id = pcm->pcm_id; + + link->cpu_dai_name = pcm->dai_name; + link->codec_name = "snd-soc-dummy"; + link->codec_dai_name = "snd-soc-dummy-dai"; + + /* enable DPCM */ + link->dynamic = 1; + link->dpcm_playback = pcm->playback; + link->dpcm_capture = pcm->capture; /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link); @@ -1639,8 +1664,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1650,7 +1673,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1670,6 +1699,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1686,6 +1720,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 39bcefe5eea0..488ef4ed8fba 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -11,6 +11,142 @@ #include "uniperif.h" /* + * User frame size shall be 2, 4, 6 or 8 32-bits words length + * (i.e. 8, 16, 24 or 32 bytes) + * This constraint comes from allowed values for + * UNIPERIF_I2S_FMT_NUM_CH register + */ +#define UNIPERIF_MAX_FRAME_SZ 0x20 +#define UNIPERIF_ALLOWED_FRAME_SZ (0x08 | 0x10 | 0x18 | UNIPERIF_MAX_FRAME_SZ) + +int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *uni = priv->dai_data.uni; + int i, frame_size, avail_slots; + + if (!UNIPERIF_TYPE_IS_TDM(uni)) { + dev_err(uni->dev, "cpu dai not in tdm mode\n"); + return -EINVAL; + } + + /* store info in unip context */ + uni->tdm_slot.slots = slots; + uni->tdm_slot.slot_width = slot_width; + /* unip is unidirectionnal */ + uni->tdm_slot.mask = (tx_mask != 0) ? tx_mask : rx_mask; + + /* number of available timeslots */ + for (i = 0, avail_slots = 0; i < uni->tdm_slot.slots; i++) { + if ((uni->tdm_slot.mask >> i) & 0x01) + avail_slots++; + } + uni->tdm_slot.avail_slots = avail_slots; + + /* frame size in bytes */ + frame_size = uni->tdm_slot.avail_slots * uni->tdm_slot.slot_width / 8; + + /* check frame size is allowed */ + if ((frame_size > UNIPERIF_MAX_FRAME_SZ) || + (frame_size & ~(int)UNIPERIF_ALLOWED_FRAME_SZ)) { + dev_err(uni->dev, "frame size not allowed: %d bytes\n", + frame_size); + return -EINVAL; + } + + return 0; +} + +int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct uniperif *uni = rule->private; + struct snd_interval t; + + t.min = uni->tdm_slot.avail_slots; + t.max = uni->tdm_slot.avail_slots; + t.openmin = 0; + t.openmax = 0; + t.integer = 0; + + return snd_interval_refine(hw_param_interval(params, rule->var), &t); +} + +int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct uniperif *uni = rule->private; + struct snd_mask *maskp = hw_param_mask(params, rule->var); + u64 format; + + switch (uni->tdm_slot.slot_width) { + case 16: + format = SNDRV_PCM_FMTBIT_S16_LE; + break; + case 32: + format = SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + dev_err(uni->dev, "format not supported: %d bits\n", + uni->tdm_slot.slot_width); + return -EINVAL; + } + + maskp->bits[0] &= (u_int32_t)format; + maskp->bits[1] &= (u_int32_t)(format >> 32); + /* clear remaining indexes */ + memset(maskp->bits + 2, 0, (SNDRV_MASK_MAX - 64) / 8); + + if (!maskp->bits[0] && !maskp->bits[1]) + return -EINVAL; + + return 0; +} + +int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, + unsigned int *word_pos) +{ + int slot_width = uni->tdm_slot.slot_width / 8; + int slots_num = uni->tdm_slot.slots; + unsigned int slots_mask = uni->tdm_slot.mask; + int i, j, k; + unsigned int word16_pos[4]; + + /* word16_pos: + * word16_pos[0] = WORDX_LSB + * word16_pos[1] = WORDX_MSB, + * word16_pos[2] = WORDX+1_LSB + * word16_pos[3] = WORDX+1_MSB + */ + + /* set unip word position */ + for (i = 0, j = 0, k = 0; (i < slots_num) && (k < WORD_MAX); i++) { + if ((slots_mask >> i) & 0x01) { + word16_pos[j] = i * slot_width; + + if (slot_width == 4) { + word16_pos[j + 1] = word16_pos[j] + 2; + j++; + } + j++; + + if (j > 3) { + word_pos[k] = word16_pos[1] | + (word16_pos[0] << 8) | + (word16_pos[3] << 16) | + (word16_pos[2] << 24); + j = 0; + k++; + } + } + } + + return 0; +} + +/* * sti_uniperiph_dai_create_ctrl * This function is used to create Ctrl associated to DAI but also pcm device. * Request is done by front end to associate ctrl with pcm device id @@ -45,10 +181,16 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *uni = priv->dai_data.uni; struct snd_dmaengine_dai_dma_data *dma_data; int transfer_size; - transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES; + if (uni->info->type == SND_ST_UNIPERIF_TYPE_TDM) + /* transfer size = user frame size (in 32-bits FIFO cell) */ + transfer_size = snd_soc_params_to_frame_size(params) / 32; + else + transfer_size = params_channels(params) * UNIPERIF_FIFO_FRAMES; dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = transfer_size; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index f0fd5a9944e9..eb9933c62ad6 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -25,7 +25,7 @@ writel_relaxed((((value) & mask) << shift), ip->base + offset) /* - * AUD_UNIPERIF_SOFT_RST reg + * UNIPERIF_SOFT_RST reg */ #define UNIPERIF_SOFT_RST_OFFSET(ip) 0x0000 @@ -50,7 +50,7 @@ UNIPERIF_SOFT_RST_SOFT_RST_MASK(ip)) /* - * AUD_UNIPERIF_FIFO_DATA reg + * UNIPERIF_FIFO_DATA reg */ #define UNIPERIF_FIFO_DATA_OFFSET(ip) 0x0004 @@ -58,7 +58,7 @@ writel_relaxed(value, ip->base + UNIPERIF_FIFO_DATA_OFFSET(ip)) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -105,7 +105,7 @@ writel_relaxed(value, ip->base + UNIPERIF_CHANNEL_STA_REG5_OFFSET(ip)) /* - * AUD_UNIPERIF_ITS reg + * UNIPERIF_ITS reg */ #define UNIPERIF_ITS_OFFSET(ip) 0x000C @@ -143,7 +143,7 @@ 0 : (BIT(UNIPERIF_ITS_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITS_BCLR reg + * UNIPERIF_ITS_BCLR reg */ /* FIFO_ERROR */ @@ -160,7 +160,7 @@ writel_relaxed(value, ip->base + UNIPERIF_ITS_BCLR_OFFSET(ip)) /* - * AUD_UNIPERIF_ITM reg + * UNIPERIF_ITM reg */ #define UNIPERIF_ITM_OFFSET(ip) 0x0018 @@ -188,7 +188,7 @@ 0 : (BIT(UNIPERIF_ITM_UNDERFLOW_REC_FAILED_SHIFT(ip)))) /* - * AUD_UNIPERIF_ITM_BCLR reg + * UNIPERIF_ITM_BCLR reg */ #define UNIPERIF_ITM_BCLR_OFFSET(ip) 0x001c @@ -213,7 +213,7 @@ UNIPERIF_ITM_BCLR_DMA_ERROR_MASK(ip)) /* - * AUD_UNIPERIF_ITM_BSET reg + * UNIPERIF_ITM_BSET reg */ #define UNIPERIF_ITM_BSET_OFFSET(ip) 0x0020 @@ -767,7 +767,7 @@ SET_UNIPERIF_REG(ip, \ UNIPERIF_CTRL_OFFSET(ip), \ UNIPERIF_CTRL_READER_OUT_SEL_SHIFT(ip), \ - CORAUD_UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) + UNIPERIF_CTRL_READER_OUT_SEL_MASK(ip), 1) /* UNDERFLOW_REC_WINDOW */ #define UNIPERIF_CTRL_UNDERFLOW_REC_WINDOW_SHIFT(ip) 20 @@ -1046,7 +1046,7 @@ UNIPERIF_STATUS_1_UNDERFLOW_DURATION_MASK(ip), value) /* - * AUD_UNIPERIF_CHANNEL_STA_REGN reg + * UNIPERIF_CHANNEL_STA_REGN reg */ #define UNIPERIF_CHANNEL_STA_REGN(ip, n) (0x0060 + (4 * n)) @@ -1057,7 +1057,7 @@ UNIPERIF_CHANNEL_STA_REGN(ip, n)) /* - * AUD_UNIPERIF_USER_VALIDITY reg + * UNIPERIF_USER_VALIDITY reg */ #define UNIPERIF_USER_VALIDITY_OFFSET(ip) 0x0090 @@ -1101,12 +1101,136 @@ UNIPERIF_DBG_STANDBY_LEFT_SP_MASK(ip), value) /* + * UNIPERIF_TDM_ENABLE + */ +#define UNIPERIF_TDM_ENABLE_OFFSET(ip) 0x0118 +#define GET_UNIPERIF_TDM_ENABLE(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip)) +#define SET_UNIPERIF_TDM_ENABLE(ip, value) \ + writel_relaxed(value, ip->base + UNIPERIF_TDM_ENABLE_OFFSET(ip)) + +/* TDM_ENABLE */ +#define UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip) 0x0 +#define UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip) 0x1 +#define GET_UNIPERIF_TDM_ENABLE_EN_TDM(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip)) +#define SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 1) +#define SET_UNIPERIF_TDM_ENABLE_TDM_DISABLE(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_ENABLE_OFFSET(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_SHIFT(ip), \ + UNIPERIF_TDM_ENABLE_EN_TDM_MASK(ip), 0) + +/* + * UNIPERIF_TDM_FS_REF_FREQ + */ +#define UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip) 0x011c +#define GET_UNIPERIF_TDM_FS_REF_FREQ(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ(ip, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip)) + +/* REF_FREQ */ +#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip) 0x0 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) 0 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) 1 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) 2 +#define VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) 3 +#define UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip) 0x3 +#define GET_UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_8KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_16KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_32KHZ(ip)) +#define SET_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_FREQ_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_FREQ_REF_FREQ_MASK(ip), \ + VALUE_UNIPERIF_TDM_FS_REF_FREQ_48KHZ(ip)) + +/* + * UNIPERIF_TDM_FS_REF_DIV + */ +#define UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip) 0x0120 +#define GET_UNIPERIF_TDM_FS_REF_DIV(ip) \ + readl_relaxed(ip->base + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip)) +#define SET_UNIPERIF_TDM_FS_REF_DIV(ip, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip)) + +/* NUM_TIMESLOT */ +#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip) 0x0 +#define UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip) 0xff +#define GET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip) \ + GET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip)) +#define SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(ip, value) \ + SET_UNIPERIF_REG(ip, \ + UNIPERIF_TDM_FS_REF_DIV_OFFSET(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_SHIFT(ip), \ + UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT_MASK(ip), value) + +/* + * UNIPERIF_TDM_WORD_POS_X_Y + * 32 bits of UNIPERIF_TDM_WORD_POS_X_Y register shall be set in 1 shot + */ +#define UNIPERIF_TDM_WORD_POS_1_2_OFFSET(ip) 0x013c +#define UNIPERIF_TDM_WORD_POS_3_4_OFFSET(ip) 0x0140 +#define UNIPERIF_TDM_WORD_POS_5_6_OFFSET(ip) 0x0144 +#define UNIPERIF_TDM_WORD_POS_7_8_OFFSET(ip) 0x0148 +#define GET_UNIPERIF_TDM_WORD_POS(ip, words) \ + readl_relaxed(ip->base + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip)) +#define SET_UNIPERIF_TDM_WORD_POS(ip, words, value) \ + writel_relaxed(value, ip->base + \ + UNIPERIF_TDM_WORD_POS_##words##_OFFSET(ip)) +/* * uniperipheral IP capabilities */ #define UNIPERIF_FIFO_SIZE 70 /* FIFO is 70 cells deep */ #define UNIPERIF_FIFO_FRAMES 4 /* FDMA trigger limit in frames */ +#define UNIPERIF_TYPE_IS_HDMI(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_HDMI) +#define UNIPERIF_TYPE_IS_PCM(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_PCM) +#define UNIPERIF_TYPE_IS_SPDIF(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_SPDIF) +#define UNIPERIF_TYPE_IS_IEC958(p) \ + (UNIPERIF_TYPE_IS_HDMI(p) || \ + UNIPERIF_TYPE_IS_SPDIF(p)) +#define UNIPERIF_TYPE_IS_TDM(p) \ + ((p)->info->type == SND_ST_UNIPERIF_TYPE_TDM) + /* * Uniperipheral IP revisions */ @@ -1125,10 +1249,11 @@ enum uniperif_version { }; enum uniperif_type { - SND_ST_UNIPERIF_PLAYER_TYPE_NONE, - SND_ST_UNIPERIF_PLAYER_TYPE_HDMI, - SND_ST_UNIPERIF_PLAYER_TYPE_PCM, - SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF + SND_ST_UNIPERIF_TYPE_NONE, + SND_ST_UNIPERIF_TYPE_HDMI, + SND_ST_UNIPERIF_TYPE_PCM, + SND_ST_UNIPERIF_TYPE_SPDIF, + SND_ST_UNIPERIF_TYPE_TDM }; enum uniperif_state { @@ -1145,9 +1270,17 @@ enum uniperif_iec958_encoding_mode { UNIPERIF_IEC958_ENCODING_MODE_ENCODED }; +enum uniperif_word_pos { + WORD_1_2, + WORD_3_4, + WORD_5_6, + WORD_7_8, + WORD_MAX +}; + struct uniperif_info { int id; /* instance value of the uniperipheral IP */ - enum uniperif_type player_type; + enum uniperif_type type; int underflow_enabled; /* Underflow recovery mode */ }; @@ -1156,12 +1289,20 @@ struct uniperif_iec958_settings { struct snd_aes_iec958 iec958; }; +struct dai_tdm_slot { + unsigned int mask; + int slots; + int slot_width; + unsigned int avail_slots; +}; + struct uniperif { /* System information */ struct uniperif_info *info; struct device *dev; int ver; /* IP version, used by register access macros */ struct regmap_field *clk_sel; + struct regmap_field *valid_sel; /* capabilities */ const struct snd_pcm_hardware *hw; @@ -1192,6 +1333,7 @@ struct uniperif { /* dai properties */ unsigned int daifmt; + struct dai_tdm_slot tdm_slot; /* DAI callbacks */ const struct snd_soc_dai_ops *dai_ops; @@ -1209,6 +1351,28 @@ struct sti_uniperiph_data { struct sti_uniperiph_dai dai_data; }; +static const struct snd_pcm_hardware uni_tdm_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID, + + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE, + + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + + .channels_min = 1, + .channels_max = 32, + + .periods_min = 2, + .periods_max = 10, + + .period_bytes_min = 128, + .period_bytes_max = 64 * PAGE_SIZE, + .buffer_bytes_max = 256 * PAGE_SIZE +}; + /* uniperiph player*/ int uni_player_init(struct platform_device *pdev, struct uniperif *uni_player); @@ -1226,4 +1390,28 @@ int sti_uniperiph_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); +static inline int sti_uniperiph_get_user_frame_size( + struct snd_pcm_runtime *runtime) +{ + return (runtime->channels * snd_pcm_format_width(runtime->format) / 8); +} + +static inline int sti_uniperiph_get_unip_tdm_frame_size(struct uniperif *uni) +{ + return (uni->tdm_slot.slots * uni->tdm_slot.slot_width / 8); +} + +int sti_uniperiph_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width); + +int sti_uniperiph_get_tdm_word_pos(struct uniperif *uni, + unsigned int *word_pos); + +int sti_uniperiph_fix_tdm_chan(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + +int sti_uniperiph_fix_tdm_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule); + #endif diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 7aca6b92f718..ee1c7c245bc7 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -21,23 +21,14 @@ /* sys config registers definitions */ #define SYS_CFG_AUDIO_GLUE 0xA4 -#define SYS_CFG_AUDI0_GLUE_PCM_CLKX 8 /* * Driver specific types. */ -#define UNIPERIF_PLAYER_TYPE_IS_HDMI(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_HDMI) -#define UNIPERIF_PLAYER_TYPE_IS_PCM(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_PCM) -#define UNIPERIF_PLAYER_TYPE_IS_SPDIF(p) \ - ((p)->info->player_type == SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF) -#define UNIPERIF_PLAYER_TYPE_IS_IEC958(p) \ - (UNIPERIF_PLAYER_TYPE_IS_HDMI(p) || \ - UNIPERIF_PLAYER_TYPE_IS_SPDIF(p)) #define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999 #define UNIPERIF_PLAYER_CLK_ADJ_MAX 1000000 +#define UNIPERIF_PLAYER_I2S_OUT 1 /* player id connected to I2S/TDM TX bus */ /* * Note: snd_pcm_hardware is linked to DMA controller but is declared here to @@ -444,18 +435,11 @@ static int uni_player_prepare_pcm(struct uniperif *player, /* Force slot width to 32 in I2S mode (HW constraint) */ if ((player->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == - SND_SOC_DAIFMT_I2S) { + SND_SOC_DAIFMT_I2S) slot_width = 32; - } else { - switch (runtime->format) { - case SNDRV_PCM_FORMAT_S16_LE: - slot_width = 16; - break; - default: - slot_width = 32; - break; - } - } + else + slot_width = snd_pcm_format_width(runtime->format); + output_frame_size = slot_width * runtime->channels; clk_div = player->mclk / runtime->rate; @@ -530,7 +514,6 @@ static int uni_player_prepare_pcm(struct uniperif *player, SET_UNIPERIF_CONFIG_ONE_BIT_AUD_DISABLE(player); SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); - SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(player); /* No iec958 formatting as outputting to DAC */ SET_UNIPERIF_CTRL_SPDIF_FMT_OFF(player); @@ -538,6 +521,55 @@ static int uni_player_prepare_pcm(struct uniperif *player, return 0; } +static int uni_player_prepare_tdm(struct uniperif *player, + struct snd_pcm_runtime *runtime) +{ + int tdm_frame_size; /* unip tdm frame size in bytes */ + int user_frame_size; /* user tdm frame size in bytes */ + /* default unip TDM_WORD_POS_X_Y */ + unsigned int word_pos[4] = { + 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A}; + int freq, ret; + + tdm_frame_size = + sti_uniperiph_get_unip_tdm_frame_size(player); + user_frame_size = + sti_uniperiph_get_user_frame_size(runtime); + + /* fix 16/0 format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(player); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(player); + + /* number of words inserted on the TDM line */ + SET_UNIPERIF_I2S_FMT_NUM_CH(player, user_frame_size / 4 / 2); + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(player); + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(player); + + /* Enable the tdm functionality */ + SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(player); + + /* number of 8 bits timeslots avail in unip tdm frame */ + SET_UNIPERIF_TDM_FS_REF_DIV_NUM_TIMESLOT(player, tdm_frame_size); + + /* set the timeslot allocation for words in FIFO */ + sti_uniperiph_get_tdm_word_pos(player, word_pos); + SET_UNIPERIF_TDM_WORD_POS(player, 1_2, word_pos[WORD_1_2]); + SET_UNIPERIF_TDM_WORD_POS(player, 3_4, word_pos[WORD_3_4]); + SET_UNIPERIF_TDM_WORD_POS(player, 5_6, word_pos[WORD_5_6]); + SET_UNIPERIF_TDM_WORD_POS(player, 7_8, word_pos[WORD_7_8]); + + /* set unip clk rate (not done vai set_sysclk ops) */ + freq = runtime->rate * tdm_frame_size * 8; + mutex_lock(&player->ctrl_lock); + ret = uni_player_clk_set_rate(player, freq); + if (!ret) + player->mclk = freq; + mutex_unlock(&player->ctrl_lock); + + return 0; +} + /* * ALSA uniperipheral iec958 controls */ @@ -668,11 +700,29 @@ static int uni_player_startup(struct snd_pcm_substream *substream, { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct uniperif *player = priv->dai_data.uni; + int ret; + player->substream = substream; player->clk_adj = 0; - return 0; + if (!UNIPERIF_TYPE_IS_TDM(player)) + return 0; + + /* refine hw constraint in tdm mode */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + sti_uniperiph_fix_tdm_chan, + player, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + if (ret < 0) + return ret; + + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + sti_uniperiph_fix_tdm_format, + player, SNDRV_PCM_HW_PARAM_FORMAT, + -1); } static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -682,7 +732,7 @@ static int uni_player_set_sysclk(struct snd_soc_dai *dai, int clk_id, struct uniperif *player = priv->dai_data.uni; int ret; - if (dir == SND_SOC_CLOCK_IN) + if (UNIPERIF_TYPE_IS_TDM(player) || (dir == SND_SOC_CLOCK_IN)) return 0; if (clk_id != 0) @@ -714,7 +764,13 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, } /* Calculate transfer size (in fifo cells and bytes) for frame count */ - transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + if (player->info->type == SND_ST_UNIPERIF_TYPE_TDM) { + /* transfer size = user frame size (in 32 bits FIFO cell) */ + transfer_size = + sti_uniperiph_get_user_frame_size(runtime) / 4; + } else { + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + } /* Calculate number of empty cells available before asserting DREQ */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) { @@ -738,16 +794,19 @@ static int uni_player_prepare(struct snd_pcm_substream *substream, SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(player, trigger_limit); /* Uniperipheral setup depends on player type */ - switch (player->info->player_type) { - case SND_ST_UNIPERIF_PLAYER_TYPE_HDMI: + switch (player->info->type) { + case SND_ST_UNIPERIF_TYPE_HDMI: ret = uni_player_prepare_iec958(player, runtime); break; - case SND_ST_UNIPERIF_PLAYER_TYPE_PCM: + case SND_ST_UNIPERIF_TYPE_PCM: ret = uni_player_prepare_pcm(player, runtime); break; - case SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF: + case SND_ST_UNIPERIF_TYPE_SPDIF: ret = uni_player_prepare_iec958(player, runtime); break; + case SND_ST_UNIPERIF_TYPE_TDM: + ret = uni_player_prepare_tdm(player, runtime); + break; default: dev_err(player->dev, "invalid player type"); return -EINVAL; @@ -852,8 +911,8 @@ static int uni_player_start(struct uniperif *player) * will not take affect and hang the player. */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) - if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) - SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); + if (UNIPERIF_TYPE_IS_IEC958(player)) + SET_UNIPERIF_CTRL_SPDIF_FMT_ON(player); /* Force channel status update (no update if clk disable) */ if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) @@ -954,27 +1013,30 @@ static void uni_player_shutdown(struct snd_pcm_substream *substream, player->substream = NULL; } -static int uni_player_parse_dt_clk_glue(struct platform_device *pdev, - struct uniperif *player) +static int uni_player_parse_dt_audio_glue(struct platform_device *pdev, + struct uniperif *player) { - int bit_offset; struct device_node *node = pdev->dev.of_node; struct regmap *regmap; - - bit_offset = SYS_CFG_AUDI0_GLUE_PCM_CLKX + player->info->id; + struct reg_field regfield[2] = { + /* PCM_CLK_SEL */ + REG_FIELD(SYS_CFG_AUDIO_GLUE, + 8 + player->info->id, + 8 + player->info->id), + /* PCMP_VALID_SEL */ + REG_FIELD(SYS_CFG_AUDIO_GLUE, 0, 1) + }; regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg"); - if (regmap) { - struct reg_field regfield = - REG_FIELD(SYS_CFG_AUDIO_GLUE, bit_offset, bit_offset); - - player->clk_sel = regmap_field_alloc(regmap, regfield); - } else { + if (!regmap) { dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n"); return -EINVAL; } + player->clk_sel = regmap_field_alloc(regmap, regfield[0]); + player->valid_sel = regmap_field_alloc(regmap, regfield[1]); + return 0; } @@ -1012,19 +1074,21 @@ static int uni_player_parse_dt(struct platform_device *pdev, } if (strcasecmp(mode, "hdmi") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; + info->type = SND_ST_UNIPERIF_TYPE_HDMI; else if (strcasecmp(mode, "pcm") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; + info->type = SND_ST_UNIPERIF_TYPE_PCM; else if (strcasecmp(mode, "spdif") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; + info->type = SND_ST_UNIPERIF_TYPE_SPDIF; + else if (strcasecmp(mode, "tdm") == 0) + info->type = SND_ST_UNIPERIF_TYPE_TDM; else - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + info->type = SND_ST_UNIPERIF_TYPE_NONE; /* Save the info structure */ player->info = info; - /* Get the PCM_CLK_SEL bit from audio-glue-ctrl SoC register */ - if (uni_player_parse_dt_clk_glue(pdev, player)) + /* Get PCM_CLK_SEL & PCMP_VALID_SEL from audio-glue-ctrl SoC reg */ + if (uni_player_parse_dt_audio_glue(pdev, player)) return -EINVAL; return 0; @@ -1037,7 +1101,8 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, - .set_sysclk = uni_player_set_sysclk + .set_sysclk = uni_player_set_sysclk, + .set_tdm_slot = sti_uniperiph_set_tdm_slot }; int uni_player_init(struct platform_device *pdev, @@ -1047,7 +1112,6 @@ int uni_player_init(struct platform_device *pdev, player->dev = &pdev->dev; player->state = UNIPERIF_STATE_STOPPED; - player->hw = &uni_player_pcm_hw; player->dai_ops = &uni_player_dai_ops; ret = uni_player_parse_dt(pdev, player); @@ -1057,6 +1121,11 @@ int uni_player_init(struct platform_device *pdev, return ret; } + if (UNIPERIF_TYPE_IS_TDM(player)) + player->hw = &uni_tdm_hw; + else + player->hw = &uni_player_pcm_hw; + /* Get uniperif resource */ player->clk = of_clk_get(pdev->dev.of_node, 0); if (IS_ERR(player->clk)) @@ -1073,6 +1142,17 @@ int uni_player_init(struct platform_device *pdev, } } + /* connect to I2S/TDM TX bus */ + if (player->valid_sel && + (player->info->id == UNIPERIF_PLAYER_I2S_OUT)) { + ret = regmap_field_write(player->valid_sel, player->info->id); + if (ret) { + dev_err(player->dev, + "%s: unable to connect to tdm bus", __func__); + return ret; + } + } + ret = devm_request_irq(&pdev->dev, player->irq, uni_player_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), player); @@ -1087,7 +1167,7 @@ int uni_player_init(struct platform_device *pdev, SET_UNIPERIF_CTRL_SPDIF_LAT_OFF(player); SET_UNIPERIF_CONFIG_IDLE_MOD_DISABLE(player); - if (UNIPERIF_PLAYER_TYPE_IS_IEC958(player)) { + if (UNIPERIF_TYPE_IS_IEC958(player)) { /* Set default iec958 status bits */ /* Consumer, PCM, copyright, 2ch, mode 0 */ diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 8a0eb2050169..eb74a328c928 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -73,55 +73,10 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) return ret; } -static int uni_reader_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int uni_reader_prepare_pcm(struct snd_pcm_runtime *runtime, + struct uniperif *reader) { - struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); - struct uniperif *reader = priv->dai_data.uni; - struct snd_pcm_runtime *runtime = substream->runtime; - int transfer_size, trigger_limit; int slot_width; - int count = 10; - - /* The reader should be stopped */ - if (reader->state != UNIPERIF_STATE_STOPPED) { - dev_err(reader->dev, "%s: invalid reader state %d", __func__, - reader->state); - return -EINVAL; - } - - /* Calculate transfer size (in fifo cells and bytes) for frame count */ - transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; - - /* Calculate number of empty cells available before asserting DREQ */ - if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) - trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; - else - /* - * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 - * FDMA_TRIGGER_LIMIT also controls when the state switches - * from OFF or STANDBY to AUDIO DATA. - */ - trigger_limit = transfer_size; - - /* Trigger limit must be an even number */ - if ((!trigger_limit % 2) || - (trigger_limit != 1 && transfer_size % 2) || - (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { - dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); - return -EINVAL; - } - - SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit); - - switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_IB_IF: - case SND_SOC_DAIFMT_NB_IF: - SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); - break; - default: - SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); - } /* Force slot width to 32 in I2S mode */ if ((reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) @@ -173,6 +128,109 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, return -EINVAL; } + /* Number of channels must be even */ + if ((runtime->channels % 2) || (runtime->channels < 2) || + (runtime->channels > 10)) { + dev_err(reader->dev, "%s: invalid nb of channels", __func__); + return -EINVAL; + } + + SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2); + SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); + + return 0; +} + +static int uni_reader_prepare_tdm(struct snd_pcm_runtime *runtime, + struct uniperif *reader) +{ + int frame_size; /* user tdm frame size in bytes */ + /* default unip TDM_WORD_POS_X_Y */ + unsigned int word_pos[4] = { + 0x04060002, 0x0C0E080A, 0x14161012, 0x1C1E181A}; + + frame_size = sti_uniperiph_get_user_frame_size(runtime); + + /* fix 16/0 format */ + SET_UNIPERIF_CONFIG_MEM_FMT_16_0(reader); + SET_UNIPERIF_I2S_FMT_DATA_SIZE_32(reader); + + /* number of words inserted on the TDM line */ + SET_UNIPERIF_I2S_FMT_NUM_CH(reader, frame_size / 4 / 2); + + SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); + SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); + SET_UNIPERIF_TDM_ENABLE_TDM_ENABLE(reader); + + /* + * set the timeslots allocation for words in FIFO + * + * HW bug: (LSB word < MSB word) => this config is not possible + * So if we want (LSB word < MSB) word, then it shall be + * handled by user + */ + sti_uniperiph_get_tdm_word_pos(reader, word_pos); + SET_UNIPERIF_TDM_WORD_POS(reader, 1_2, word_pos[WORD_1_2]); + SET_UNIPERIF_TDM_WORD_POS(reader, 3_4, word_pos[WORD_3_4]); + SET_UNIPERIF_TDM_WORD_POS(reader, 5_6, word_pos[WORD_5_6]); + SET_UNIPERIF_TDM_WORD_POS(reader, 7_8, word_pos[WORD_7_8]); + + return 0; +} + +static int uni_reader_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + struct snd_pcm_runtime *runtime = substream->runtime; + int transfer_size, trigger_limit, ret; + int count = 10; + + /* The reader should be stopped */ + if (reader->state != UNIPERIF_STATE_STOPPED) { + dev_err(reader->dev, "%s: invalid reader state %d", __func__, + reader->state); + return -EINVAL; + } + + /* Calculate transfer size (in fifo cells and bytes) for frame count */ + if (reader->info->type == SND_ST_UNIPERIF_TYPE_TDM) { + /* transfer size = unip frame size (in 32 bits FIFO cell) */ + transfer_size = + sti_uniperiph_get_user_frame_size(runtime) / 4; + } else { + transfer_size = runtime->channels * UNIPERIF_FIFO_FRAMES; + } + + /* Calculate number of empty cells available before asserting DREQ */ + if (reader->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) + trigger_limit = UNIPERIF_FIFO_SIZE - transfer_size; + else + /* + * Since SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0 + * FDMA_TRIGGER_LIMIT also controls when the state switches + * from OFF or STANDBY to AUDIO DATA. + */ + trigger_limit = transfer_size; + + /* Trigger limit must be an even number */ + if ((!trigger_limit % 2) || + (trigger_limit != 1 && transfer_size % 2) || + (trigger_limit > UNIPERIF_CONFIG_DMA_TRIG_LIMIT_MASK(reader))) { + dev_err(reader->dev, "invalid trigger limit %d", trigger_limit); + return -EINVAL; + } + + SET_UNIPERIF_CONFIG_DMA_TRIG_LIMIT(reader, trigger_limit); + + if (UNIPERIF_TYPE_IS_TDM(reader)) + ret = uni_reader_prepare_tdm(runtime, reader); + else + ret = uni_reader_prepare_pcm(runtime, reader); + if (ret) + return ret; + switch (reader->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: SET_UNIPERIF_I2S_FMT_ALIGN_LEFT(reader); @@ -191,21 +249,26 @@ static int uni_reader_prepare(struct snd_pcm_substream *substream, return -EINVAL; } - SET_UNIPERIF_I2S_FMT_ORDER_MSB(reader); - - /* Data clocking (changing) on the rising edge */ - SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); - - /* Number of channels must be even */ - - if ((runtime->channels % 2) || (runtime->channels < 2) || - (runtime->channels > 10)) { - dev_err(reader->dev, "%s: invalid nb of channels", __func__); - return -EINVAL; + /* Data clocking (changing) on the rising/falling edge */ + switch (reader->daifmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); + break; + case SND_SOC_DAIFMT_NB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_RISING(reader); + break; + case SND_SOC_DAIFMT_IB_NF: + SET_UNIPERIF_I2S_FMT_LR_POL_LOW(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader); + break; + case SND_SOC_DAIFMT_IB_IF: + SET_UNIPERIF_I2S_FMT_LR_POL_HIG(reader); + SET_UNIPERIF_I2S_FMT_SCLK_EDGE_FALLING(reader); + break; } - SET_UNIPERIF_I2S_FMT_NUM_CH(reader, runtime->channels / 2); - /* Clear any pending interrupts */ SET_UNIPERIF_ITS_BCLR(reader, GET_UNIPERIF_ITS(reader)); @@ -293,6 +356,32 @@ static int uni_reader_trigger(struct snd_pcm_substream *substream, } } +static int uni_reader_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); + struct uniperif *reader = priv->dai_data.uni; + int ret; + + if (!UNIPERIF_TYPE_IS_TDM(reader)) + return 0; + + /* refine hw constraint in tdm mode */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + sti_uniperiph_fix_tdm_chan, + reader, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); + if (ret < 0) + return ret; + + return snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + sti_uniperiph_fix_tdm_format, + reader, SNDRV_PCM_HW_PARAM_FORMAT, + -1); +} + static void uni_reader_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -310,6 +399,7 @@ static int uni_reader_parse_dt(struct platform_device *pdev, { struct uniperif_info *info; struct device_node *node = pdev->dev.of_node; + const char *mode; /* Allocate memory for the info structure */ info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); @@ -322,6 +412,17 @@ static int uni_reader_parse_dt(struct platform_device *pdev, return -EINVAL; } + /* Read the device mode property */ + if (of_property_read_string(node, "st,mode", &mode)) { + dev_err(&pdev->dev, "uniperipheral mode not defined"); + return -EINVAL; + } + + if (strcasecmp(mode, "tdm") == 0) + info->type = SND_ST_UNIPERIF_TYPE_TDM; + else + info->type = SND_ST_UNIPERIF_TYPE_PCM; + /* Save the info structure */ reader->info = info; @@ -329,11 +430,13 @@ static int uni_reader_parse_dt(struct platform_device *pdev, } static const struct snd_soc_dai_ops uni_reader_dai_ops = { + .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, + .set_tdm_slot = sti_uniperiph_set_tdm_slot }; int uni_reader_init(struct platform_device *pdev, @@ -343,7 +446,6 @@ int uni_reader_init(struct platform_device *pdev, reader->dev = &pdev->dev; reader->state = UNIPERIF_STATE_STOPPED; - reader->hw = &uni_reader_pcm_hw; reader->dai_ops = &uni_reader_dai_ops; ret = uni_reader_parse_dt(pdev, reader); @@ -352,6 +454,11 @@ int uni_reader_init(struct platform_device *pdev, return ret; } + if (UNIPERIF_TYPE_IS_TDM(reader)) + reader->hw = &uni_tdm_hw; + else + reader->hw = &uni_reader_pcm_hw; + ret = devm_request_irq(&pdev->dev, reader->irq, uni_reader_irq_handler, IRQF_SHARED, dev_name(&pdev->dev), reader); diff --git a/sound/usb/card.c b/sound/usb/card.c index 3fc63583a537..69860da473ea 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -350,6 +350,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: break; default: dev_err(&dev->dev, "unknown device speed %d\n", snd_usb_get_speed(dev)); @@ -450,6 +451,9 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_SUPER: strlcat(card->longname, ", super speed", sizeof(card->longname)); break; + case USB_SPEED_SUPER_PLUS: + strlcat(card->longname, ", super speed plus", sizeof(card->longname)); + break; default: break; } diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf6a147..26dd5f20f149 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 51ed1ac825fd..7712e2b84183 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -120,6 +120,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: if (get_endpoint(alts, 0)->bInterval >= 1 && get_endpoint(alts, 0)->bInterval <= 4) return get_endpoint(alts, 0)->bInterval - 1; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 47de8af42f16..7ba92921bf28 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -911,6 +911,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, switch (snd_usb_get_speed(ep->umidi->dev)) { case USB_SPEED_HIGH: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: count = 1; break; default: diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757009b3..2f8c388ef84f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); @@ -2307,6 +2374,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, __u8 unitid = (index >> 8) & 0xff; __u8 control = (value >> 8) & 0xff; __u8 channel = value & 0xff; + unsigned int count = 0; if (channel >= MAX_CHANNELS) { usb_audio_dbg(mixer->chip, @@ -2315,6 +2383,12 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + count++; + + if (count == 0) + return; + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { struct usb_mixer_elem_info *info; @@ -2322,7 +2396,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, continue; info = (struct usb_mixer_elem_info *)list; - if (info->control != control) + if (count > 1 && info->control != control) continue; switch (attribute) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0adfd9537cf7..6adde457b602 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,8 +1137,11 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85555cb..4d5c89a7ba2b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -47,6 +47,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ |