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

[alsa-devel] [PATCH] ASoC: ak4613: add Digital Playback Volume support
by Kuninori Morimoto 20 Sep '15

20 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For L/ROUT1 to L/ROUT6 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index fd96a8f..07a2664 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -22,6 +22,7 @@ #include <linux/regmap.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include <sound/tlv.h> #define PW_MGMT1 0x00 /* Power Management 1 */ #define PW_MGMT2 0x01 /* Power Management 2 */ @@ -91,6 +92,31 @@ struct ak4613_interface { struct ak4613_formats playback; }; +/* + * Playback Volume + * + * max : 0x00 : 0 dB + * ( 0.5 dB step ) + * min : 0xFE : -127.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new ak4613_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6, + 0, 0xFF, 1, out_tlv), +}; + static const struct reg_default ak4613_reg[] = { { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, @@ -397,6 +423,8 @@ static int ak4613_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .resume = ak4613_resume, .set_bias_level = ak4613_set_bias_level, + .controls = ak4613_snd_controls, + .num_controls = ARRAY_SIZE(ak4613_snd_controls), .dapm_widgets = ak4613_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), .dapm_routes = ak4613_intercon, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency
by Axel Lin 20 Sep '15

20 Sep '15

Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on CLKDEV_LOOKUP to fix below build warning: warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP) Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/rockchip/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 58bae8e..5709057 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -17,7 +17,7 @@ config SND_SOC_ROCKCHIP_I2S config SND_SOC_ROCKCHIP_MAX98090 tristate "ASoC support for Rockchip boards using a MAX98090 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98090 select SND_SOC_TS3A227E @@ -27,7 +27,7 @@ config SND_SOC_ROCKCHIP_MAX98090 config SND_SOC_ROCKCHIP_RT5645 tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_RT5645 help -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm0010: disable regulator on error
by Sudip Mukherjee 20 Sep '15

20 Sep '15

We have done regulator_bulk_enable() while booting the DSP but on the error exit path we have not disbled it. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/wm0010.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8434d45..79a7cd3 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -739,8 +739,6 @@ static int wm0010_boot(struct snd_soc_codec *codec) abort: /* Put the chip back into reset */ wm0010_halt(codec); - mutex_unlock(&wm0010->lock); - return ret; err_core: mutex_unlock(&wm0010->lock); -- 1.9.1

3 9

[alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 20 Sep '15

20 Sep '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 8

[alsa-devel] [PATCH RFC v2 0/7] Implement generic ASoC HDMI codec
by Jyri Sarha 20 Sep '15

20 Sep '15

Only the first three patches are meant for serious review. The ASoC part should be ready for review in other respects but the EDID SADs handling is waiting for Russell King's DRM ELD helper patches. There is a copy-pasted not-to-be-merged patch with the same functionality in the patch series. The tda998x part is not complete and I do not think Jean-Francois' binding are ready as such either. At least the usage of reg property in a port node for something that is not an address or even guaranteed to be unique is not a good idea. Some documentation of how to configure AP0-AP4 pins (REG_ENA_AP?) would help in making a proper binding for the tda998x audio configuration. These patches, my tilcdc refactoring[1], and my latest BCLK fixes for davinci-mcasp driver [2], can be found in a branch that produces working HDMI audio on Beaglebone-Black here: https://github.com/jsarha/linux.git linux-master-bbb-hdmi-20150512 [1] http://lists.freedesktop.org/archives/dri-devel/2015-May/082537.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090974.html Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports DO NOT MERGE Jyri Sarha (6): ASoC: core: If component doesn't have of_node use parent's node instead ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add ELD based audio pcm rules DO NOT MERGE drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality DO NOT MERGE ARM: dts: am335x-boneblack: Add HDMI audio support DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++ arch/arm/boot/dts/am335x-boneblack.dts | 78 ++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 328 +++++++++++- include/sound/hdmi-codec.h | 99 ++++ include/sound/tda998x.h | 8 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 560 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ---- sound/soc/soc-core.c | 14 +- 11 files changed, 1129 insertions(+), 125 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

2 12

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fixing invalid fmt SND_SOC_DAIFMT_AC97 test
by Arnaud Mouiche 20 Sep '15

20 Sep '15

SND_SOC_DAIFMT_AC97 (6) has similar bits than SND_SOC_DAIFMT_DSP_A (4) and can't be tested with a simple AND operation Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..c25a1e8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: add wm8960 support
by Zidan Wang 19 Sep '15

19 Sep '15

add wm8960 support for fsl-asoc-card Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ sound/soc/fsl/fsl-asoc-card.c | 37 ++++++++++++++++++++-- 2 files changed, 36 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..fa0a10d 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -59,6 +59,8 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. + - gpr : The gpr handle + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..c058e68 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -16,6 +16,7 @@ #include <linux/of_platform.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <linux/mfd/syscon.h> #include "fsl_esai.h" #include "fsl_sai.h" @@ -23,6 +24,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -90,6 +92,7 @@ struct fsl_asoc_card_priv { u32 asrc_format; u32 dai_fmt; char name[32]; + struct regmap *gpr; }; /** @@ -400,7 +403,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) static int fsl_asoc_card_probe(struct platform_device *pdev) { - struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *cpu_np, *codec_np, *asrc_np, *gpr_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct platform_device *cpu_pdev; @@ -443,6 +446,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); + gpr_np = of_parse_phandle(np, "gpr", 0); + if (gpr_np) { + priv->gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(priv->gpr)) { + ret = PTR_ERR(priv->gpr); + dev_err(&pdev->dev, "failed to get gpr regmap\n"); + goto fail; + } + /* + * for imx6ul we should set SAI2_MCLK_DIR to enable + * codec MCLK + */ + if (of_device_is_compatible(np, "fsl,imx6ul-evk-wm8960")) + regmap_update_bits(priv->gpr, 4, 1 << 20, 1 << 20); + } + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ codec_clk = clk_get(&codec_dev->dev, NULL); if (!IS_ERR(codec_clk)) { @@ -475,6 +494,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -521,7 +545,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[0].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[0].codec_dai_name = codec_dev->name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +557,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[2].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[2].codec_dai_name = codec_dev->name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; @@ -578,6 +608,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; -- 1.9.1

4 15

[alsa-devel] [RESEND PATCH 1/1] ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
by Adam Thomson 19 Sep '15

19 Sep '15

Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Patch based on v4.3-rc1. sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8ec6fb2..37c5cd4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.3

3 2

[alsa-devel] [PATCH v2] ASoC: fsl-asoc-card: add AC'97 support
by Maciej S. Szmigiero 19 Sep '15

19 Sep '15

Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: don't use of_find_device_by_node() to get pointer to I2S/PCM CODEC device as this only matches platform bus devices while I2S/PCM CODECs supported by this driver sit on I2C bus. .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 0901d5e..1b05d1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +533,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +565,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +591,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +621,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,14 +658,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", },

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Re: [alsa-devel] Asus N751JK - push/merge patches on mailinglist
by Raymond Yau 18 Sep '15

18 Sep '15

2015-09-13 16:13 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > On Sat, 12 Sep 2015 23:09:43 +0200, > Arthur Borsboom wrote: > > > > Hi Takashi, > > > > Previous month Raymond Yau has done great work to get the audio working > for > > the Asus N751JK laptop. > > He developed 7 patches, which I have tested and confirmed working on this > > laptop. > > > > Raymond is not responding for about two weeks (maybe on holiday?). > > Would you mind to post these patches on the mailinglist (the correct way) > > and/or merge them in a way the get into the mainline kernel a.s.a.p? > > Sorry, it can't be done like that. The patches aren't yet in > acceptable forms, and it needs more changes to be merged to mainline. > It must be done by someone who did it originally or who has the > hardware and test. > I don't have the hardware, the alternate approach is to create headset jack (SND_JACK_HEADSET) and skip the creation of headset mic phanton jack instead of headphone jack and headset mic jack since implement sense_nid and slave_nid is not easy to debug What you need is adding "Jack Headset" to pulseaudio analog-input-headset-mic.conf since "Headset Jack" kctl already in analog-out-headphones.conf static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: case AC_JACK_SPEAKER: return SND_JACK_LINEOUT; case AC_JACK_HP_OUT: + if (spec->hs_mic_use_hp_sense && + (nid == auto_cfg_hp_pins(cfg)[0])) + return SND_JACK_HEADSET; return SND_JACK_HEADPHONE; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_gen_spec *spec = codec->spec; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -482,6 +499,12 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense && + !is_jack_detectable(codec, cfg->inputs[i].pin)) { + err = 0; + + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -489,14 +512,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, } for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, NULL); + if (is_subwoofer(codec, *p, i, cfg->line_outs) && + is_external_pin(codec, *p)) + err = add_jack_kctl(codec, *p, cfg, "Ext Subwoofer"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, NULL); + if (i == 0 && spec->hs_mic_use_hp_sense) + err = add_jack_kctl(codec, *p, cfg, "Headset"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } change the pin to hp pin when mic pin is headset mic +bool is_headset_mic(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) + if (nid == cfg->inputs[i].pin) + return cfg->inputs[i].is_headset_mic; + return false; +} + @@ -4384,6 +4399,8 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; + if (is_headset_mic(codec, pin) && spec->hs_mic_use_hp_sense) + pin = auto_cfg_hp_pins(cfg)[0]; if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return;

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