November 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] mx28: bad unlock balance detected
by Fabio Estevam 03 Dec '15

03 Dec '15

Hi, Running linux-next 20151126 on a mx28evk leads to the following bug: sgtl5000 0-000a: sgtl5000 revision 0x11 sgtl5000 0-000a: Using internal LDO instead of VDDD ===================================== [ BUG: bad unlock balance detected! ] 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Not tainted ------------------------------------- kworker/u2:0/6 is trying to release lock (&rdev->mutex) at: [<c02e7858>] regulator_get_voltage+0x30/0x44 but there are no more locks to release! other info that might help us debug this: 5 locks held by kworker/u2:0/6: #0: ("%s""deferwq"){++++.+}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #1: (deferred_probe_work){+.+.+.}, at: [<c002f1f8>] process_one_work+0x10c/0x49c #2: (&dev->mutex){......}, at: [<c03197bc>] __device_attach+0x20/0x108 #3: (client_mutex){+.+.+.}, at: [<c042dc34>] snd_soc_register_card+0x280/0x1108 #4: (&card->mutex){+.+.+.}, at: [<c042dc40>] snd_soc_register_card+0x28c/0x1108 stack backtrace: CPU: 0 PID: 6 Comm: kworker/u2:0 Not tainted 4.4.0-rc2-next-20151126-00001-g44eb6b2 #156 Hardware name: Freescale MXS (Device Tree) Workqueue: deferwq deferred_probe_work_func [<c000eea0>] (unwind_backtrace) from [<c000cfa0>] (show_stack+0x10/0x14) [<c000cfa0>] (show_stack) from [<c0049350>] (print_unlock_imbalance_bug+0xac/0xdc) [<c0049350>] (print_unlock_imbalance_bug) from [<c004c2f8>] (lock_release+0x2a4/0x38c) [<c004c2f8>] (lock_release) from [<c0513e70>] (__mutex_unlock_slowpath+0x7c/0x15c) [<c0513e70>] (__mutex_unlock_slowpath) from [<c02e7858>] (regulator_get_voltage+0x30/0x44) [<c02e7858>] (regulator_get_voltage) from [<c043c200>] (sgtl5000_probe+0x1c0/0x5f0) [<c043c200>] (sgtl5000_probe) from [<c0428a74>] (snd_soc_codec_drv_probe+0x14/0x18) [<c0428a74>] (snd_soc_codec_drv_probe) from [<c042cd54>] (soc_probe_component+0x200/0x330) [<c042cd54>] (soc_probe_component) from [<c042e178>] (snd_soc_register_card+0x7c4/0x1108) [<c042e178>] (snd_soc_register_card) from [<c0439cdc>] (devm_snd_soc_register_card+0x30/0x70) [<c0439cdc>] (devm_snd_soc_register_card) from [<c043d1cc>] (mxs_sgtl5000_probe+0xc8/0x138) [<c043d1cc>] (mxs_sgtl5000_probe) from [<c031b1fc>] (platform_drv_probe+0x50/0xac) [<c031b1fc>] (platform_drv_probe) from [<c0319ab0>] (driver_probe_device+0x1e4/0x2b0) [<c0319ab0>] (driver_probe_device) from [<c03180d8>] (bus_for_each_drv+0x44/0x8c) [<c03180d8>] (bus_for_each_drv) from [<c0319840>] (__device_attach+0xa4/0x108) [<c0319840>] (__device_attach) from [<c0318f84>] (bus_probe_device+0x88/0x90) [<c0318f84>] (bus_probe_device) from [<c0319370>] (deferred_probe_work_func+0x64/0x94) [<c0319370>] (deferred_probe_work_func) from [<c002f290>] (process_one_work+0x1a4/0x49c) [<c002f290>] (process_one_work) from [<c002f5e8>] (worker_thread+0x28/0x468) [<c002f5e8>] (worker_thread) from [<c0034dbc>] (kthread+0xc0/0xdc) [<c0034dbc>] (kthread) from [<c000a2ec>] (ret_from_fork+0x14/0x28) mxs-sgtl5000 sound: sgtl5000 <-> 80042000.saif mapping ok By inspecting the sgtl5000 codec driver I could not find where this "bad unlock balance" is happening. Any ideas? Thanks, Fabio Estevam

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[alsa-devel] [PATCH v4 0/3]ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 02 Dec '15

02 Dec '15

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.3-rc5 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v4 Changes from v3: - Fix the subject lines to match the style for the subsystem Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 54 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 859 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1508 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 129 ++ 6 files changed, 2564 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] snd-fireworks requires larger period sizes than FFADO
by Dominic Sacré 02 Dec '15

02 Dec '15

Hi, I'm trying to use an Echo AudioFire 4 with the snd-fireworks driver. With FFADO the device works reasonably well at a period size of 64 samples (on a PREEMPT_RT kernel), whereas with snd-fireworks the lowest possible period size seems to be 256 samples. This is what JACK says when I try to start it with a period size of 128 samples: configuring for 48000Hz, period = 128 frames (2.7 ms), buffer = 2 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: cannot set period size to 128 frames for capture The lowest achievable analog roundtrip latency seems to be around 19ms (at 2x256 frames), as opposed to 8.3ms with FFADO (at 2x64). Are there any known limitations of the snd-fireworks driver in this regard? Thanks, Dominic

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[alsa-devel] [PATCH 0/14] ASoC: rsnd: TDM Extend Mode support
by Kuninori Morimoto 02 Dec '15

02 Dec '15

Hi Mark These patch-set adds TDM Extend Mode to Renesas R-Car sound driver. 1) - 9) tidyup / cleanup patches 10) - 14) TDM Extend Mode support Unfortunately, our board can't use this TDM Extend Mode, because our SoC and codec are using different style TDM. Thus I have tested this patches on special board which doesn't have codec chip, and I checked signal via oscilloscope. oKuninori Morimoto (14): 1) ASoC: rsnd: don't use normal *mod in adg.c 2) ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU 3) ASoC: rsnd: add missing SRC_O_BUSIF_MODE register 4) ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() 5) ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() 6) ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() 7) ASoC: rsnd: src: add rsnd_src_halt() 8) ASoC: rsnd: mix: add rsnd_mix_halt() 9) ASoC: rsnd: dvc: add rsnd_dvc_halt() 10) ASoC: rsnd: add rsnd_ssi_config_init() 11) ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() 12) ASoC: rsnd: ssi enables non-stereo sound 13) ASoC: rsnd: dvc enables non-stereo sound 14) ASoC: rsnd: add TDM Extend Mode support sound/soc/sh/rcar/adg.c | 30 +++++++++++------------------- sound/soc/sh/rcar/core.c | 65 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----- sound/soc/sh/rcar/dvc.c | 32 +++++++++++++++++++++++++++----- sound/soc/sh/rcar/gen.c | 11 ++++++++++- sound/soc/sh/rcar/mix.c | 12 ++++++++++-- sound/soc/sh/rcar/rsnd.h | 19 +++++++++++++++++-- sound/soc/sh/rcar/src.c | 18 ++++++++++++++---- sound/soc/sh/rcar/ssi.c | 126 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++--------------------------------------------------- sound/soc/sh/rcar/ssiu.c | 12 +++++++++++- 9 files changed, 235 insertions(+), 90 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: arizona: Add 32uS delay after putting FLL into freerun
by Charles Keepax 01 Dec '15

01 Dec '15

When switching between two clock sources using the FLL freerun to smooth the transition we should wait 32uS after putting the FLL into freerun before we proceed. In practice we appear to be getting enough delay from the surrounding code, but better to make it explicit. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e76ecc7..a23f7d1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2212,9 +2212,9 @@ static int arizona_enable_fll(struct arizona_fll *fll) /* Facilitate smooth refclk across the transition */ regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); - regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, - ARIZONA_FLL1_FREERUN, - ARIZONA_FLL1_FREERUN); + regmap_update_bits(fll->arizona->regmap, fll->base + 1, + ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN); + udelay(32); } /* -- 2.1.4

2 1

[alsa-devel] [asoc:topic/da7218 2/2] sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool
by kbuild test robot 01 Dec '15

01 Dec '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/da7218 head: 4d50934abd2261fd467320d52c470efff309fd74 commit: 4d50934abd2261fd467320d52c470efff309fd74 [2/2] ASoC: da7218: Add da7218 codec driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/da7218.c:3214:9-10: WARNING: return of 0/1 in function 'da7218_volatile_register' with return type bool Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH 0/3] ASoC: Implement DAI links in a list & some code refactoring
by mengdong.lin＠linux.intel.com 01 Dec '15

01 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This is still a preparation for topology to create PCM at dynamically. First two patches are code refactoring. The 3rd patch implements the DAI links into a list, so that the number of links can add/shrink by topology at runtime. The use scenario may be like this: - When instantiating the soc card, probing a component with topology can bring new DAI links. The ASoC will probe the new links (in later patches). - When unregistering the soc card, removing this component will also remove DAI links created by it. For backward compatiblity, the machine driver can still define a static link array. And the ASoC core will add these links into the list when instantiating the soc card. Mengdong Lin (3): ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add/remove a link include/sound/soc.h | 15 ++- sound/soc/soc-core.c | 251 +++++++++++++++++++++++++++++++++------------------ 2 files changed, 177 insertions(+), 89 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 0/2] correction of iec958 channel status sampling freq setting
by Moise Gergaud 01 Dec '15

01 Dec '15

patch "ASoC: sti: set iec958 channel status sampling freq" corresponds to the V2 of "[PATCH 4/4] ASoC: sti: reset iec60958 settings on close" Moise Gergaud (2): ASoC: sti: set iec958 channel status sampling freq ASoC: sti: correction for HBRA support sound/soc/sti/uniperif_player.c | 73 +++++++++++++++++++++++++---------------- 1 file changed, 44 insertions(+), 29 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sun4i: Implement MIC1 capture
by Maxime Ripard 01 Dec '15

01 Dec '15

One of the input path used in the Allwinner codec is the MIC1. Add support for it. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/soc/sunxi/sun4i-codec.c | 228 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 197 insertions(+), 31 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da168b6..30c9e9260491 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -69,6 +69,7 @@ /* Codec ADC register offsets and bit fields */ #define SUN4I_CODEC_ADC_FIFOC (0x1c) +#define SUN4I_CODEC_ADC_FIFOC_ADC_FS (29) #define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) #define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24) #define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8) @@ -102,6 +103,7 @@ struct sun4i_codec { struct clk *clk_apb; struct clk *clk_module; + struct snd_dmaengine_dai_dma_data capture_dma_data; struct snd_dmaengine_dai_dma_data playback_dma_data; }; @@ -136,26 +138,54 @@ static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) 0); } +static void sun4i_codec_start_capture(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO high here on some boards + */ + + /* Enable ADC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN)); +} + +static void sun4i_codec_stop_capture(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO low here on some boards + */ + + /* Disable ADC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN), 0); +} + static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - sun4i_codec_start_playback(scodec); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sun4i_codec_start_playback(scodec); + else + sun4i_codec_start_capture(scodec); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - sun4i_codec_stop_playback(scodec); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sun4i_codec_stop_playback(scodec); + else + sun4i_codec_stop_capture(scodec); break; default: @@ -165,15 +195,54 @@ static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int sun4i_codec_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int sun4i_codec_prepare_capture(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - u32 val; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; + + /* Flush RX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH)); + + + /* Set RX FIFO trigger level */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + 0xf << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL, + 0x7 << SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL); + + /* + * FIXME: Undocumented in the datasheet, but + * Allwinner's code mentions that it is related + * related to microphone gain + */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_ACTL, + 0x3 << 25, + 0x1 << 25); + + if (of_device_is_compatible(scodec->dev->of_node, + "allwinner,sun7i-a20-codec")) + /* FIXME: Undocumented bits */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_TUNE, + 0x3 << 8, + 0x1 << 8); + + /* Fill most significant bits with valid data MSB */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE), + BIT(SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE)); + + return 0; +} + +static int sun4i_codec_prepare_playback(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + u32 val; /* Flush the TX FIFO */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, @@ -202,6 +271,15 @@ static int sun4i_codec_prepare(struct snd_pcm_substream *substream, 0); return 0; +}; + +static int sun4i_codec_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return sun4i_codec_prepare_playback(substream, dai); + + return sun4i_codec_prepare_capture(substream, dai); } static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params) @@ -276,30 +354,34 @@ static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params) } } -static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int sun4i_codec_hw_params_capture(struct sun4i_codec *scodec, + struct snd_pcm_hw_params *params, + unsigned int hwrate) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - unsigned long clk_freq; - int ret, hwrate; u32 val; - if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) - return -ENOTSUPP; + /* Set ADC sample rate */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + 7 << SUN4I_CODEC_ADC_FIFOC_ADC_FS, + hwrate << SUN4I_CODEC_ADC_FIFOC_ADC_FS); - clk_freq = sun4i_codec_get_mod_freq(params); - if (!clk_freq) - return -EINVAL; + /* Set the number of channels we want to use */ + if (params_channels(params) == 1) + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN)); + else + regmap_update_bits(scodec->regmap, SUN4I_CODEC_ADC_FIFOC, + BIT(SUN4I_CODEC_ADC_FIFOC_MONO_EN), 0); - ret = clk_set_rate(scodec->clk_module, clk_freq); - if (ret) - return ret; + return 0; +} - hwrate = sun4i_codec_get_hw_rate(params); - if (hwrate < 0) - return hwrate; +static int sun4i_codec_hw_params_playback(struct sun4i_codec *scodec, + struct snd_pcm_hw_params *params, + unsigned int hwrate) +{ + u32 val; /* Set DAC sample rate */ regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, @@ -344,6 +426,34 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, return 0; } +static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + unsigned long clk_freq; + int hwrate; + + clk_freq = sun4i_codec_get_mod_freq(params); + if (!clk_freq) + return -EINVAL; + + if (clk_set_rate(scodec->clk_module, clk_freq)) + return -EINVAL; + + hwrate = sun4i_codec_get_hw_rate(params); + if (hwrate < 0) + return hwrate; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return sun4i_codec_hw_params_playback(scodec, params, + hwrate); + + return sun4i_codec_hw_params_capture(scodec, params, + hwrate); +} + static int sun4i_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -394,6 +504,20 @@ static struct snd_soc_dai_driver sun4i_codec_dai = { SNDRV_PCM_FMTBIT_S32_LE, .sig_bits = 24, }, + .capture = { + .stream_name = "Codec Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, }; /*** Codec ***/ @@ -429,11 +553,22 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { }; static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { + /* Digital parts of the ADCs */ + SND_SOC_DAPM_SUPPLY("ADC", SUN4I_CODEC_ADC_FIFOC, + SUN4I_CODEC_ADC_FIFOC_EN_AD, 0, + NULL, 0), + /* Digital parts of the DACs */ SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC, SUN4I_CODEC_DAC_DPC_EN_DA, 0, NULL, 0), + /* Analog parts of the ADCs */ + SND_SOC_DAPM_ADC("Left ADC", "Codec Capture", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_ADC_L_EN, 0), + SND_SOC_DAPM_ADC("Right ADC", "Codec Capture", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_ADC_R_EN, 0), + /* Analog parts of the DACs */ SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_DACAENL, 0), @@ -452,6 +587,14 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), + /* VMIC */ + SND_SOC_DAPM_SUPPLY("VMIC", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_VMICEN, 0, NULL, 0), + + /* Mic Pre-Amplifiers */ + SND_SOC_DAPM_PGA("MIC1 Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, 0, NULL, 0), + /* Pre-Amplifier */ SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, SUN4I_CODEC_ADC_ACTL_PA_EN, 0, @@ -460,15 +603,19 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, &sun4i_codec_pa_mute), + SND_SOC_DAPM_INPUT("Mic1"), + SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), }; static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { - /* Left DAC Routes */ + /* Left ADC / DAC Routes */ + { "Left ADC", NULL, "ADC" }, { "Left DAC", NULL, "DAC" }, - /* Right DAC Routes */ + /* Right ADC / DAC Routes */ + { "Right ADC", NULL, "ADC" }, { "Right DAC", NULL, "DAC" }, /* Right Mixer Routes */ @@ -490,6 +637,12 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Mic1 Routes */ + { "Left ADC", NULL, "MIC1 Pre-Amplifier" }, + { "Right ADC", NULL, "MIC1 Pre-Amplifier" }, + { "MIC1 Pre-Amplifier", NULL, "Mic1"}, + { "Mic1", NULL, "VMIC" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -515,7 +668,7 @@ static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data, - NULL); + &scodec->capture_dma_data); return 0; } @@ -531,6 +684,14 @@ static struct snd_soc_dai_driver dummy_cpu_dai = { .formats = SUN4I_CODEC_FORMATS, .sig_bits = 24, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, + .sig_bits = 24, + }, }; static const struct regmap_config sun4i_codec_regmap_config = { @@ -638,6 +799,11 @@ static int sun4i_codec_probe(struct platform_device *pdev) scodec->playback_dma_data.maxburst = 4; scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + /* DMA configuration for RX FIFO */ + scodec->capture_dma_data.addr = res->start + SUN4I_CODEC_ADC_RXDATA; + scodec->capture_dma_data.maxburst = 4; + scodec->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, &sun4i_codec_dai, 1); if (ret) { -- 2.6.3

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[alsa-devel] [PATCH 0/7] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 30 Nov '15

30 Nov '15

Hi, this is a patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. The initial draft was posted to alsa-devel as an RFC some weeks ago, and this is a revised version, posted to both i915 and alsa-devel for review. The current patchset is found in sound git tree test/hdmi-jack branch. Takashi Takashi Iwai (7): drm/i915: Remove superfluous NULL check drm/i915: Add get_eld audio component drm/i915: refactoring audio component functions ALSA: hda - Split ELD update code from hdmi_present_sense() ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Do zero-clear in snd_hdmi_parse_eld() itself ALSA: hda - Skip ELD notification during PM process drivers/gpu/drm/i915/intel_audio.c | 75 ++++++++++---- drivers/gpu/drm/i915/intel_drv.h | 1 + include/drm/i915_component.h | 3 + sound/pci/hda/hda_eld.c | 1 + sound/pci/hda/patch_hdmi.c | 195 ++++++++++++++++++++++++++----------- 5 files changed, 200 insertions(+), 75 deletions(-) -- 2.6.3

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