September 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] The best practice to tie platform DAPM widget and codec pin
by Anatol Pomozov 13 Sep '14

13 Sep '14

Hi I bringing up a board that has Realtek 5677 codec. The codec has 6 gpios and two of them are used by the board. pin3 is used to enable headphones amplifier. pin2 is used to gate DMIC clock for 3 of 4 DMICs - it allows to leave only one DMIC working in low-power mode. Now I need to configure the codec that when "Headphones" audio path is enabled then pin3 should go high. The same true for DMIC/pin2. I was looking for existing similar solutions but cannot find one. I need something similar to SND_SOC_DAPM_REGULATOR_SUPPLY that allows to turn on/off part of the system. But instead of manipulating a voltage regulator I just need to set/unset codec's gpio. What would be the best way to achieve it? The problem here is that codec does not know which of the outputs (LOUT1, LOUT2, LOUT3) corresponds to headphones. It is configured as audio path in *.dts file.

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[alsa-devel] [PATCH] ASoC: fsl-ssi: Support for SND_SOC_DAIFMT_CBM_CFS
by Fabio Falzoi 12 Sep '14

12 Sep '14

Add SND_SOC_DAIFMT_CBM_CFS support for Freescale architecture. Successfully tested on i.MX 6Quad Wandboard and UDOO boards connected to the pcm1792a codec. In CBM_CFS mode, when using a sample size of 16 bits, we cannot use CCSR_SSI_SCR_I2S_MODE_MASTER since we get a frame sync every 16 bits. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> Signed-off-by: Fabio Falzoi <fabio.falzoi84(a)gmail.com> Tested-by: Angelo Adamo <adamo.a60(a)gmail.com> --- sound/soc/fsl/fsl_ssi.c | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..2fc3e66 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -259,6 +259,11 @@ static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) SND_SOC_DAIFMT_CBS_CFS; } +static bool fsl_ssi_is_i2s_cbm_cfs(struct fsl_ssi_private *ssi_private) +{ + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFS; +} /** * fsl_ssi_isr: SSI interrupt handler * @@ -705,6 +710,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + if (!fsl_ssi_is_ac97(ssi_private)) { + u8 i2smode; + /* + * Switch to normal net mode in order to have a frame sync + * signal every 32 bits instead of 16 bits + */ + if (fsl_ssi_is_i2s_cbm_cfs(ssi_private) && sample_size == 16) + i2smode = CCSR_SSI_SCR_I2S_MODE_NORMAL | + CCSR_SSI_SCR_NET; + else + i2smode = ssi_private->i2s_mode; + + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, + channels == 1 ? 0 : i2smode); + } + /* * FIXME: The documentation says that SxCCR[WL] should not be * modified while the SSI is enabled. The only time this can @@ -724,11 +746,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(regs, CCSR_SSI_SRCCR, CCSR_SSI_SxCCR_WL_MASK, wl); - if (!fsl_ssi_is_ac97(ssi_private)) - regmap_update_bits(regs, CCSR_SSI_SCR, - CCSR_SSI_SCR_NET | CCSR_SSI_SCR_I2S_MODE_MASK, - channels == 1 ? 0 : ssi_private->i2s_mode); - return 0; } @@ -780,6 +797,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: 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, @@ -853,6 +871,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, case SND_SOC_DAIFMT_CBM_CFM: scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr &= ~CCSR_SSI_STCR_TXDIR; + strcr |= CCSR_SSI_STCR_TFDIR; + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; default: return -EINVAL; } -- 2.0.0

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[alsa-devel] [PATCH v2 0/2] ASoC/ARM: tegra: apalis/colibri t30: sgtl5000 audio
by Marcel Ziswiler 12 Sep '14

12 Sep '14

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Changes in v2: - Using simple-audio-card as suggested by Fabio. Marcel Ziswiler (2): ARM: tegra: apalis/colibri t30: integrate audio support ARM: tegra: enable sgtl5000 audio arch/arm/boot/dts/tegra30-apalis.dtsi | 65 ++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 65 ++++++++++++++++++++++++++++++++++ arch/arm/configs/tegra_defconfig | 2 ++ 3 files changed, 132 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH V2] ASoC: fsl_ssi: refine ipg clock usage in this module
by Shengjiu Wang 12 Sep '14

12 Sep '14

Move the ipg clock enable and disable operation to startup and shutdown, that is only enable ipg clock when ssi is working. Keep clock is disabled when ssi is in idle. otherwise, _fsl_ssi_set_dai_fmt function need to be called in probe, so add ipg clock control for it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- change log v2: update patch according to maintainer's review comments. sound/soc/fsl/fsl_ssi.c | 39 +++++++++++++++++++++++++++++++-------- 1 file changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2fc3e66..4447f95 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -530,6 +530,11 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + int ret; + + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; /* When using dual fifo mode, it is safer to ensure an even period * size. If appearing to an odd number while DMA always starts its @@ -544,6 +549,21 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** + * fsl_ssi_shutdown: shutdown the SSI + * + */ +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_ssi_private *ssi_private = + snd_soc_dai_get_drvdata(rtd->cpu_dai); + + clk_disable_unprepare(ssi_private->clk); + +} + +/** * fsl_ssi_set_bclk - configure Digital Audio Interface bit clock * * Note: This function can be only called when using SSI as DAI master @@ -771,6 +791,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; u8 wm; + int ret; ssi_private->dai_fmt = fmt; @@ -779,6 +800,10 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, return -EINVAL; } + ret = clk_prepare_enable(ssi_private->clk); + if (ret) + return ret; + fsl_ssi_setup_reg_vals(ssi_private); regmap_read(regs, CCSR_SSI_SCR, &scr); @@ -811,6 +836,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; break; default: + clk_disable_unprepare(ssi_private->clk); return -EINVAL; } @@ -836,6 +862,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_NORMAL; break; default: + clk_disable_unprepare(ssi_private->clk); return -EINVAL; } scr |= ssi_private->i2s_mode; @@ -859,6 +886,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, strcr ^= CCSR_SSI_STCR_TFSI; break; default: + clk_disable_unprepare(ssi_private->clk); return -EINVAL; } @@ -877,6 +905,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: + clk_disable_unprepare(ssi_private->clk); return -EINVAL; } @@ -925,6 +954,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, if (fmt & SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); + clk_disable_unprepare(ssi_private->clk); return 0; } @@ -1043,6 +1073,7 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, + .shutdown = fsl_ssi_shutdown, .hw_params = fsl_ssi_hw_params, .hw_free = fsl_ssi_hw_free, .set_fmt = fsl_ssi_set_dai_fmt, @@ -1175,12 +1206,6 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return ret; } - ret = clk_prepare_enable(ssi_private->clk); - if (ret) { - dev_err(&pdev->dev, "clk_prepare_enable failed: %d\n", ret); - return ret; - } - /* For those SLAVE implementations, we ingore non-baudclk cases * and, instead, abandon MASTER mode that needs baud clock. */ @@ -1236,7 +1261,6 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, return 0; error_pcm: - clk_disable_unprepare(ssi_private->clk); return ret; } @@ -1246,7 +1270,6 @@ static void fsl_ssi_imx_clean(struct platform_device *pdev, { if (!ssi_private->use_dma) imx_pcm_fiq_exit(pdev); - clk_disable_unprepare(ssi_private->clk); } static int fsl_ssi_probe(struct platform_device *pdev) -- 1.7.9.5

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[alsa-devel] usb sound cannot get frequency from enpoind
by vichy 12 Sep '14

12 Sep '14

hi all: My kernel is 3.8.0. and when I plug in "microsoft LifeCam", I get message at the end of mail about cannot get min/max values from camera. is there any usb audio class initial sequence we need to check when this message pop out? Appreciate your help in advance, usb 3-1: new high-speed USB device number 2 using Platform-ehci Platform-ehci Platform-ehci.0: port 1 reset complete, port enabled Platform-ehci Platform-ehci.0: GetStatus port:1 status 00000d 0 ACK sig=se0 PEC PE CONNECT usb 3-1: skipped 1 descriptor after configuration usb 3-1: skipped 6 descriptors after interface usb 3-1: skipped 1 descriptor after endpoint usb 3-1: skipped 28 descriptors after interface usb 3-1: skipped 1 descriptor after endpoint usb 3-1: skipped 4 descriptors after interface usb 3-1: skipped 2 descriptors after interface usb 3-1: skipped 1 descriptor after endpoint usb 3-1: default language 0x0409 usb 3-1: udev 2, busnum 3, minor = 257 usb 3-1: New USB device found, idVendor=045e, idProduct=075d usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 3-1: Product: MicrosoftÂ® LifeCam Cinema(TM) usb 3-1: Manufacturer: Microsoft usb 3-1: usb_probe_device usb 3-1: configuration #1 chosen from 1 choice usb 3-1: adding 3-1:1.0 (config #1, interface 0) uvcvideo 3-1:1.0: usb_probe_interface uvcvideo 3-1:1.0: usb_probe_interface - got id uvcvideo: Found UVC 1.00 device MicrosoftÂ® LifeCam Cinema(TM) (045e:075d) input: MicrosoftÂ® LifeCam Cinema(TM) as /devices/platform/Platform-ehci.0/usb3/3-1/3-1:1.0/input/input4 usb 3-1: adding 3-1:1.1 (config #1, interface 1) usb 3-1: adding 3-1:1.2 (config #1, interface 2) snd-usb-audio 3-1:1.2: usb_probe_interface snd-usb-audio 3-1:1.2: usb_probe_interface - got id ALSA sound/usb/stream.c:682 2:3:1: add audio endpoint 0x82 usb 3-1: khubd timed out on ep0in len=0/3 ALSA sound/usb/clock.c:237 2:3:1: cannot get freq at ep 0x82 ALSA sound/usb/mixer.c:1305 [5] FU [Mic Capture Switch] ch = 1, val = 0/1/1 ALSA sound/usb/mixer.c:929 5:2: cannot get min/max values for control 2 (id 5) ALSA sound/usb/mixer.c:1305 [5] FU [Mic Capture Volume] ch = 1, val = 0/1/1 ALSA sound/usb/mixer.c:929 5:2: cannot get min/max values for control 2 (id 5) usb 3-1: adding 3-1:1.3 (config #1, interface 3) hub 3-0:1.0: state 7 ports 1 chg 0000 evt 0002 Platform-ehci Platform-ehci.0: GetStatus port:1 status 00000d 0 ACK sig=se0 PEC PE CONNECT hub 3-0:1.0: port 1 enable change, status 00000403

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Re: [alsa-devel] Possible fix for snd-hda-intel model=no-jd failing since ~linux-3.9-rc1
by Adam Richter 12 Sep '14

12 Sep '14

On Thu, 9/11/14, Takashi Iwai <tiwai(a)suse.de> wrote: > Could you give alsa-info.sh output on your machine? Otherwise it's difficult to analyze. OK. I have attached it. In a few places, I replaced text with "[DELETED]" to avoid identifying the product. > Speaking of your patch: add_jack_kctl() itself has a check of is_jack_detectable() for phantom jacks, > so basically this shouldn'tmake any difference. So we need to check more deeply why this > change is really needed. Right. This is some kind of initialization order problem where the following data flow fails: codec->no_jack_detect ==> jack->phantom_jack ==> jack->pin_sense. The Intel driver has a lot of "codec" sub-drivers, and I wasn't sure if would would really be practical for me to understand and get the relevant codec maintainers to approve if necessary. The fix I proposed is something I see as a temporary very low overhead workaround (only adds a few instructions, and only at initialization) until a correct fix is implemented. However, if this can be fixed properly quicly enoughtso avoid the need for a temporary workaround, that would be great. >thanks, > >Takashi Thank you for your analysis of this. Adam

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[alsa-devel] Configuring I2S2 bus as a path to speaker
by Anatol Pomozov 12 Sep '14

12 Sep '14

Hi I am working on a board that has following configuration: [SoC] <== I2S1 ==> [RT5677 Codec] <== I2S2 ==> [Speaker Amp] The right part is the interesting one. Traditionally I2S is used to connect to SoC, but in my case it is used to connect Speaker. Is there any board that has similar configuration? How to configure I2S2? *Where* and when such configuration should be done?

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Re: [alsa-devel] [PATCH v5] ASoC: tda998x: Add a codec to the HDMI transmitter
by Dave Airlie 12 Sep '14

12 Sep '14

On 10 September 2014 19:29, Jean-Francois Moine <moinejf(a)free.fr> wrote: > This patch adds a CODEC function to the NXP TDA998x HDMI transmitter. > > The CODEC handles both I2S and S/PDIF inputs. > It maintains the audio format and rate constraints according > to the HDMI device parameters (EDID) and does dynamic input > switch in the TDA998x I2C driver on start/stop audio streaming. > You should indicate on subsystem spanning patches what tree you think should merge it etc. If other tda998x ppl are okay with it, you can have my ack for merging via someone else. Dave.

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[alsa-devel] [PATCH] ASoC: rt5677: Add sidetone controls
by Anatol Pomozov 12 Sep '14

12 Sep '14

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rt5677.c | 7 +++++++ sound/soc/codecs/rt5677.h | 4 ++++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index cf41e89..4972b5b 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -541,6 +541,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); +static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ static unsigned int bst_tlv[] = { @@ -577,6 +578,12 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5677_IN1, RT5677_BST_SFT2, 8, 0, bst_tlv), + /* Sidetone Control */ + SOC_SINGLE("Sidetone Switch", RT5677_SIDETONE_CTRL, + RT5677_ST_EN_SFT, 1, 0), + SOC_SINGLE_TLV("Sidetone Volume", RT5677_SIDETONE_CTRL, + RT5677_ST_VOL_SFT, 31, 0, st_vol_tlv), + /* ADC Digital Volume Control */ SOC_DOUBLE("ADC1 Capture Switch", RT5677_STO1_ADC_DIG_VOL, RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 3c1b4dc..f28911d 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -382,6 +382,10 @@ #define RT5677_ST_SEL_SFT 9 #define RT5677_ST_EN (0x1 << 6) #define RT5677_ST_EN_SFT 6 +#define RT5677_ST_GAIN (0x1 << 5) +#define RT5677_ST_GAIN_SFT 5 +#define RT5677_ST_VOL_MASK (0x1f << 0) +#define RT5677_ST_VOL_SFT 0 /* Analog DAC1/2/3 Source Control (0x15) */ #define RT5677_ANA_DAC3_SRC_SEL_MASK (0x3 << 4) -- 2.1.0.rc2.206.gedb03e5

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[alsa-devel] [PATCH v5] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 11 Sep '14

11 Sep '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 143 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 146 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 6b59471..3ee031e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..51c6c4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -906,6 +906,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..1db2168 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1461,7 +1461,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1473,7 +1474,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index cdc837e..d0323f1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -729,6 +729,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2664,6 +2694,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3142,6 +3175,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config += w->params_select; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3222,8 +3258,35 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3231,14 +3294,48 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = kcalloc(num_params, sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + w_param_text[count] = kmemdup((void *)(config->stream_name), + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3246,6 +3343,25 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + private_value = + (unsigned long) kmemdup((void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + template.kcontrol_news = kmemdup(&kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3253,15 +3369,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1