From 4a4436573a6669516f73bac25016683d396ed4c4 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 31 Mar 2016 16:35:58 +0300 Subject: ALSA: pcm: add IEC958 channel status helper for hw_params Add IEC958 channel status helper that gets the audio properties from snd_pcm_hw_params instead of snd_pcm_runtime. This is needed to produce the channel status bits already in audio stream configuration phase. Signed-off-by: Jyri Sarha Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm_iec958.h | 2 ++ 1 file changed, 2 insertions(+) (limited to 'include') diff --git a/include/sound/pcm_iec958.h b/include/sound/pcm_iec958.h index 0eed397aca8e..36f023acb201 100644 --- a/include/sound/pcm_iec958.h +++ b/include/sound/pcm_iec958.h @@ -6,4 +6,6 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, size_t len); +int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params, + u8 *cs, size_t len); #endif -- cgit v1.2.3 From 09184118a8abae030539469848d475adcc0e5839 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 31 Mar 2016 16:36:00 +0300 Subject: ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders The hdmi-codec is a platform device driver to be registered from drivers of external HDMI encoders with I2S and/or spdif interface. The driver in turn registers an ASoC codec for the HDMI encoder's audio functionality. The structures and definitions in the API header are mostly redundant copies of similar structures in ASoC headers. This is on purpose to avoid direct dependencies to ASoC structures in video side driver. Signed-off-by: Jyri Sarha Acked-by: Arnaud Pouliquen Acked-by: PC Liao Signed-off-by: Mark Brown --- include/sound/hdmi-codec.h | 100 +++++++++++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 396 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 504 insertions(+) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c (limited to 'include') diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h new file mode 100644 index 000000000000..fc3a481ad91e --- /dev/null +++ b/include/sound/hdmi-codec.h @@ -0,0 +1,100 @@ +/* + * hdmi-codec.h - HDMI Codec driver API + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Jyri Sarha + * + * 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 __HDMI_CODEC_H__ +#define __HDMI_CODEC_H__ + +#include +#include +#include +#include + +/* + * Protocol between ASoC cpu-dai and HDMI-encoder + */ +struct hdmi_codec_daifmt { + enum { + HDMI_I2S, + HDMI_RIGHT_J, + HDMI_LEFT_J, + HDMI_DSP_A, + HDMI_DSP_B, + HDMI_AC97, + HDMI_SPDIF, + } fmt; + int bit_clk_inv:1; + int frame_clk_inv:1; + int bit_clk_master:1; + int frame_clk_master:1; +}; + +/* + * HDMI audio parameters + */ +struct hdmi_codec_params { + struct hdmi_audio_infoframe cea; + struct snd_aes_iec958 iec; + int sample_rate; + int sample_width; + int channels; +}; + +struct hdmi_codec_ops { + /* + * Called when ASoC starts an audio stream setup. + * Optional + */ + int (*audio_startup)(struct device *dev); + + /* + * Configures HDMI-encoder for audio stream. + * Mandatory + */ + int (*hw_params)(struct device *dev, + struct hdmi_codec_daifmt *fmt, + struct hdmi_codec_params *hparms); + + /* + * Shuts down the audio stream. + * Mandatory + */ + void (*audio_shutdown)(struct device *dev); + + /* + * Mute/unmute HDMI audio stream. + * Optional + */ + int (*digital_mute)(struct device *dev, bool enable); + + /* + * Provides EDID-Like-Data from connected HDMI device. + * Optional + */ + int (*get_eld)(struct device *dev, uint8_t *buf, size_t len); +}; + +/* HDMI codec initalization data */ +struct hdmi_codec_pdata { + const struct hdmi_codec_ops *ops; + uint i2s:1; + uint spdif:1; + int max_i2s_channels; +}; + +#define HDMI_CODEC_DRV_NAME "hdmi-audio-codec" + +#endif /* __HDMI_CODEC_H__ */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..06d0e0593ec3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -88,6 +88,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_NAU8825 if I2C + select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM179X_I2C if I2C select SND_SOC_PCM179X_SPI if SPI_MASTER @@ -477,6 +478,11 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..d7185dda58b8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -81,6 +81,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o +snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm179x-codec-objs := pcm179x.o snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o @@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c new file mode 100644 index 000000000000..b46b8edb9319 --- /dev/null +++ b/sound/soc/codecs/hdmi-codec.c @@ -0,0 +1,396 @@ +/* + * ALSA SoC codec for HDMI encoder drivers + * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Jyri Sarha + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include + +#include /* This is only to get MAX_ELD_BYTES */ + +struct hdmi_codec_priv { + struct hdmi_codec_pdata hcd; + struct snd_soc_dai_driver *daidrv; + struct hdmi_codec_daifmt daifmt[2]; + struct mutex current_stream_lock; + struct snd_pcm_substream *current_stream; + struct snd_pcm_hw_constraint_list ratec; + uint8_t eld[MAX_ELD_BYTES]; +}; + +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +enum { + DAI_ID_I2S = 0, + DAI_ID_SPDIF, +}; + +static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + mutex_lock(&hcp->current_stream_lock); + if (!hcp->current_stream) { + hcp->current_stream = substream; + } else if (hcp->current_stream != substream) { + dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + ret = -EINVAL; + } + mutex_unlock(&hcp->current_stream_lock); + + return ret; +} + +static int hdmi_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + if (hcp->hcd.ops->audio_startup) { + ret = hcp->hcd.ops->audio_startup(dai->dev->parent); + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); + return ret; + } + } + + if (hcp->hcd.ops->get_eld) { + ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld, + sizeof(hcp->eld)); + + if (!ret) { + ret = snd_pcm_hw_constraint_eld(substream->runtime, + hcp->eld); + if (ret) + return ret; + } + } + return 0; +} + +static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + WARN_ON(hcp->current_stream != substream); + + hcp->hcd.ops->audio_shutdown(dai->dev->parent); + + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); +} + +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_params hp = { + .iec = { + .status = { 0 }, + .subcode = { 0 }, + .pad = 0, + .dig_subframe = { 0 }, + } + }; + int ret; + + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, + params_width(params), params_rate(params), + params_channels(params)); + + if (params_width(params) > 24) + params->msbits = 24; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, + sizeof(hp.iec.status)); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + hdmi_audio_infoframe_init(&hp.cea); + hp.cea.channels = params_channels(params); + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; + + hp.sample_width = params_width(params); + hp.sample_rate = params_rate(params); + hp.channels = params_channels(params); + + return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id], + &hp); +} + +static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_daifmt cf = { 0 }; + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (dai->id == DAI_ID_SPDIF) { + cf.fmt = HDMI_SPDIF; + } else { + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + } + + hcp->daifmt[dai->id] = cf; + + return ret; +} + +static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (hcp->hcd.ops->digital_mute) + return hcp->hcd.ops->digital_mute(dai->dev->parent, mute); + + return 0; +} + +static const struct snd_soc_dai_ops hdmi_dai_ops = { + .startup = hdmi_codec_startup, + .shutdown = hdmi_codec_shutdown, + .hw_params = hdmi_codec_hw_params, + .set_fmt = hdmi_codec_set_fmt, + .digital_mute = hdmi_codec_digital_mute, +}; + + +#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +/* + * This list is only for formats allowed on the I2S bus. So there is + * some formats listed that are not supported by HDMI interface. For + * instance allowing the 32-bit formats enables 24-precision with CPU + * DAIs that do not support 24-bit formats. If the extra formats cause + * problems, we should add the video side driver an option to disable + * them. + */ +#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", + .id = DAI_ID_I2S, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = HDMI_RATES, + .formats = I2S_FORMATS, + .sig_bits = 24, + }, + .ops = &hdmi_dai_ops, +}; + +static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", + .id = DAI_ID_SPDIF, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = HDMI_RATES, + .formats = SPDIF_FORMATS, + }, + .ops = &hdmi_dai_ops, +}; + +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + +static int hdmi_codec_probe(struct platform_device *pdev) +{ + struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; + struct device *dev = &pdev->dev; + struct hdmi_codec_priv *hcp; + int dai_count, i = 0; + int ret; + + dev_dbg(dev, "%s()\n", __func__); + + if (!hcd) { + dev_err(dev, "%s: No plalform data\n", __func__); + return -EINVAL; + } + + dai_count = hcd->i2s + hcd->spdif; + if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params || + !hcd->ops->audio_shutdown) { + dev_err(dev, "%s: Invalid parameters\n", __func__); + return -EINVAL; + } + + hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL); + if (!hcp) + return -ENOMEM; + + hcp->hcd = *hcd; + mutex_init(&hcp->current_stream_lock); + + hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + GFP_KERNEL); + if (!hcp->daidrv) + return -ENOMEM; + + if (hcd->i2s) { + hcp->daidrv[i] = hdmi_i2s_dai; + hcp->daidrv[i].playback.channels_max = + hcd->max_i2s_channels; + i++; + } + + if (hcd->spdif) + hcp->daidrv[i] = hdmi_spdif_dai; + + ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, + dai_count); + if (ret) { + dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n", + __func__, ret); + return ret; + } + + dev_set_drvdata(dev, hcp); + return 0; +} + +static int hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_codec_driver = { + .driver = { + .name = HDMI_CODEC_DRV_NAME, + }, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, +}; + +module_platform_driver(hdmi_codec_driver); + +MODULE_AUTHOR("Jyri Sarha "); +MODULE_DESCRIPTION("HDMI Audio Codec Driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME); -- cgit v1.2.3 From 2f0ad49104cbb19db24442af736614659363d2ab Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 19 Apr 2016 13:12:35 +0800 Subject: ASoC: Change DAI link's be_id to a generic id The generic ID can be used by topology: - Toplogy can create FE links and set their ID, machine drivers will be notified and check this ID for machine-specific init. - Toplogy can use the ID to find existing BE & CC links and further configure them. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++++------ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 12 ++++++------ sound/soc/intel/boards/skl_rt286.c | 10 +++++----- 11 files changed, 25 insertions(+), 25 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a215fd75..ef25e86d51ee 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1002,7 +1002,7 @@ struct snd_soc_dai_link { */ const char *platform_name; struct device_node *platform_of_node; - int be_id; /* optional ID for machine driver BE identification */ + int id; /* optional ID for machine driver link identification */ const struct snd_soc_pcm_stream *params; unsigned int num_params; diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3f8a1e10bed0..7486a0022fde 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e753f0b..88efb62439ba 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1c95ccc886c4..35f591eab3c9 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index e609f089593a..6260df6bd49c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2a6f80843bc9..0618a7f1025b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2e5347f8f96c..df9d254baa18 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..863f1d5e2a2c 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b79a18d..9cc9240ed717 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -456,7 +456,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -472,7 +472,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -489,7 +489,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -501,7 +501,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -512,7 +512,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99ae9b0..53380b2cb1a8 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -505,7 +505,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -540,7 +540,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -552,7 +552,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -563,7 +563,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -574,7 +574,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a8e75..9e39fc1b89d3 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -375,7 +375,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -393,7 +393,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "dmic01", - .be_id = 1, + .id = 1, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -405,7 +405,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp1", - .be_id = 2, + .id = 2, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -416,7 +416,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp2", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -427,7 +427,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp3", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", -- cgit v1.2.3 From 305e9020f09d28560373c0112682e6fd11e909f6 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Tue, 19 Apr 2016 13:12:25 +0800 Subject: ASoC: Export snd_soc_find_dai() This API can be used by topology to find an existing BE dai by name and further configure it. Topology will also check DAI ID to avoid wrong match. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-core.c | 3 ++- 2 files changed, 5 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 02b4a215fd75..7687e2d4b0e4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card, int snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv); +struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc); + #include #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b159610..07663def2db6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,7 +930,7 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_dai *snd_soc_find_dai( +struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { struct snd_soc_component *component; @@ -959,6 +959,7 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_find_dai); static bool soc_is_dai_link_bound(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) -- cgit v1.2.3 From 73fe01cfb3babff01748a9fbc95cc3ea2079cc7f Mon Sep 17 00:00:00 2001 From: Matthias Reichl Date: Wed, 27 Apr 2016 15:26:51 +0200 Subject: ASoC: dmaengine_pcm: Add support for packed transfers dmaengine_pcm currently only supports setups where FIFO reads/writes correspond to exactly one sample, eg 16-bit sample data is transferred via 16-bit FIFO accesses, 32-bit data via 32-bit accesses. This patch adds support for setups with fixed width FIFOs where multiple samples are packed into a larger word. For example setups with a 32-bit wide FIFO register that expect 16-bit sample transfers to be done with the left+right sample data packed into a 32-bit word. Support for packed transfers is controlled via the SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags If this flag is set dmaengine_pcm doesn't put any restriction on the supported formats and sets the DMA transfer width to undefined. This means control over the constraints is now transferred to the DAI driver and it's responsible to provide proper configuration and check for possible corner cases that aren't handled by the ALSA core. Signed-off-by: Matthias Reichl Acked-by: Lars-Peter Clausen Tested-by: Martin Sperl Signed-off-by: Mark Brown --- include/sound/dmaengine_pcm.h | 12 ++++++++ sound/core/pcm_dmaengine.c | 11 +++++-- sound/soc/soc-generic-dmaengine-pcm.c | 57 +++++++++++++++++++++-------------- 3 files changed, 55 insertions(+), 25 deletions(-) (limited to 'include') diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f86ef5ea9b01..67be2445941a 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, void *filter_data); struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); +/* + * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word. + * If this flag is set the dmaengine driver won't put any restriction on + * the supported sample formats and set the DMA transfer size to undefined. + * The DAI driver is responsible to disable any unsupported formats in it's + * configuration and catch corner cases that are not already handled in + * the ALSA core. + */ +#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0) + /** * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data * @addr: Address of the DAI data source or destination register. @@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * requesting the DMA channel. * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes + * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data { void *filter_data; const char *chan_name; unsigned int fifo_size; + unsigned int flags; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166acf05..8eb58c709b14 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906af387..6cef3977507a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } -- cgit v1.2.3 From 4d2458507d0b465c62ae80f3e81b8c008ec96b05 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 May 2016 19:33:59 -0300 Subject: ASoC: fsl_sai: Allow setting the SAI MCLK direction On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for configuring the direction of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce the "fsl,sai-mclk-direction-output" optional property to allow configuring the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 5 +++++ include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 6 ++++++ sound/soc/fsl/fsl_sai.c | 20 ++++++++++++++++++++ 3 files changed, 31 insertions(+) (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 777b941d6cbe..740b467adf7d 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,11 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - fsl,sai-mclk-direction-output: This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db953eb..68353822afce 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,11 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) +#define IMX6UL_GPR1_SAI_MCLK_MASK (0x7 << 19) +#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \ + IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d8b673f7c577..2147994ab46f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include #include #include +#include +#include #include "fsl_sai.h" #include "imx-pcm.h" @@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + int index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -878,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + return PTR_ERR(gpr); + } + + index = of_alias_get_id(np, "sai"); + if (index < 0) + return index; + + regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index), + MCLK_DIR(index)); + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- cgit v1.2.3 From 32902177f7f6ae70e1d5e71d935aa1bfcae7f01c Mon Sep 17 00:00:00 2001 From: John Keeping Date: Thu, 12 May 2016 13:55:53 +0100 Subject: ASoC: dapm: deprecate MICBIAS widget type Commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget", 2011-09-23) says: A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs. but this has never been documented as a policy. Add some comments to make it clear. Signed-off-by: John Keeping Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 97069466c38d..3101d53468aa 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -100,6 +100,7 @@ struct device; { .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = wncontrols} +/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */ #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_micbias, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ @@ -473,7 +474,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_out_drv, /* output driver */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ - snd_soc_dapm_micbias, /* microphone bias (power) */ + snd_soc_dapm_micbias, /* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */ snd_soc_dapm_mic, /* microphone */ snd_soc_dapm_hp, /* headphones */ snd_soc_dapm_spk, /* speaker */ -- cgit v1.2.3 From 4446085d21e75dd6c0c45577f12db0bd7c7bf35f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 May 2016 08:58:53 +0530 Subject: ALSA: hdac: add link pm and ref counting The HDA links can be switched off when not is use, similarly command DMA can be stopped as well. This calls for a reference counting mechanism on the link by it's users to manage the link power. The DMA can be turned off when all links are off For this we add two APIs snd_hdac_ext_bus_link_get snd_hdac_ext_bus_link_put They help users to turn up/down link and manage the DMA as well Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Acked-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 66 +++++++++++++++++++++++++++++++++++++ 3 files changed, 82 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 07fa59237feb..b9593b201599 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -14,6 +14,8 @@ * @gtscap: gts capabilities pointer * @drsmcap: dma resume capabilities pointer * @hlink_list: link list of HDA links + * @lock: lock for link mgmt + * @cmd_dma_state: state of cmd DMAs: CORB and RIRB */ struct hdac_ext_bus { struct hdac_bus bus; @@ -27,6 +29,9 @@ struct hdac_ext_bus { void __iomem *drsmcap; struct list_head hlink_list; + + struct mutex lock; + bool cmd_dma_state; }; int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev, @@ -142,6 +147,9 @@ struct hdac_ext_link { void __iomem *ml_addr; /* link output stream reg pointer */ u32 lcaps; /* link capablities */ u16 lsdiid; /* link sdi identifier */ + + int ref_count; + struct list_head list; }; @@ -154,6 +162,11 @@ void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link, void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link, int stream); +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link); +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link); + /* update register macro */ #define snd_hdac_updatel(addr, reg, mask, val) \ writel(((readl(addr + reg) & ~(mask)) | (val)), \ diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..3b7ae24900fd 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, INIT_LIST_HEAD(&ebus->hlink_list); ebus->idx = idx++; + mutex_init(&ebus->lock); + ebus->cmd_dma_state = true; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 548cc1e4114b..860f8cad6602 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP); hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID); + /* since link in On, update the ref */ + hlink->ref_count = 1; + list_add_tail(&hlink->list, &ebus->hlink_list); } @@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); + +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + + mutex_lock(&ebus->lock); + + /* + * if we move from 0 to 1, count will be 1 so power up this link + * as well, also check the dma status and trigger that + */ + if (++link->ref_count == 1) { + if (!ebus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(&ebus->bus); + ebus->cmd_dma_state = true; + } + + ret = snd_hdac_ext_bus_link_power_up(link); + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); + +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + struct hdac_ext_link *hlink; + bool link_up = false; + + mutex_lock(&ebus->lock); + + /* + * if we move from 1 to 0, count will be 0 + * so power down this link as well + */ + if (--link->ref_count == 0) { + ret = snd_hdac_ext_bus_link_power_down(link); + + /* + * now check if all links are off, if so turn off + * cmd dma as well + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) { + link_up = true; + break; + } + } + + if (!link_up) { + snd_hdac_bus_stop_cmd_io(&ebus->bus); + ebus->cmd_dma_state = false; + } + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); -- cgit v1.2.3