March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] NHLT Specification
by Iker Perez del Palomar 26 Mar '19

26 Mar '19

Hello, My name is Iker Perez, I am contacting you because at the moment I am working with NHLT at my company, Codethink. This is the first time we have been involved with it, so we have been trying to find a specification or a document that could help us understand it better. We tried the Intel website [1] but it hasn't got the document it states is linked to it. I was wondering if someone could point us toward any documentation you are aware of? Thank you, Regards, Iker -- Iker Perez del Palomar, Software Engineer Codethink Ltd 35 Dale St, Manchester M1 2HF http://www.codethink.co.uk/ Manchester, M1 2JW, United Kingdom We respect your privacy. See https://www.codethink.co.uk/privacy.html

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Only enable with common clock
by Mark Brown 26 Mar '19

26 Mar '19

Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..419142111b6d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -1110,11 +1110,13 @@ config SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_I2C tristate "Texas Instruments TLV320AIC32x4 audio CODECs - I2C" depends on I2C + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_SPI tristate "Texas Instruments TLV320AIC32x4 audio CODECs - SPI" depends on SPI_MASTER + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC3X -- 2.20.1

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[alsa-devel] Applied "ASoC: Intel: Skylake: enable S24_LE format support" to the asoc tree
by Mark Brown 26 Mar '19

26 Mar '19

The patch ASoC: Intel: Skylake: enable S24_LE format support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 36e075ce74ec4e261a638bf09d10b3348ca4d883 Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Sat, 23 Mar 2019 18:40:10 +0530 Subject: [PATCH] ASoC: Intel: Skylake: enable S24_LE format support To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 28c4806b196a..4bf70b4429f0 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -483,6 +483,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.bit_depth = format->bit_depth; base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + base_cfg->audio_fmt.sample_type = format->sample_type; dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", format->bit_depth, format->valid_bit_depth, -- 2.20.1

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[alsa-devel] [PATCH v4 0/3] Add NXP AUDMIX device and machine drivers
by Viorel Suman 26 Mar '19

26 Mar '19

The patchset adds NXP Audio Mixer (AUDMIX) device and machine drivers and related DT bindings documentation. Changes since V3: 1. Removed machine driver DT bindings documentation. 2. Trigger machine driver probe from device driver as suggested by Nicolin. Changes since V2: 1. Moved "dais" node from machine driver DTS node to device driver DTS node as suggested by Rob. Changes since V1: 1. Original patch split into distinct patches for the device driver and DT binding documentation. 2. Replaced AMIX with AUDMIX in both code and file names as it looks more RM-compliant. 3. Removed polarity control from CPU DAI driver as suggested by Nicolin. 4. Added machine driver and related DT binding documentation. Viorel Suman (3): ASoC: fsl: Add Audio Mixer CPU DAI driver ASoC: add fsl_audmix DT binding documentation ASoC: fsl: Add Audio Mixer machine driver .../devicetree/bindings/sound/fsl,audmix.txt | 54 ++ sound/soc/fsl/Kconfig | 16 + sound/soc/fsl/Makefile | 5 + sound/soc/fsl/fsl_audmix.c | 576 +++++++++++++++++++++ sound/soc/fsl/fsl_audmix.h | 102 ++++ sound/soc/fsl/imx-audmix.c | 327 ++++++++++++ 6 files changed, 1080 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,audmix.txt create mode 100644 sound/soc/fsl/fsl_audmix.c create mode 100644 sound/soc/fsl/fsl_audmix.h create mode 100644 sound/soc/fsl/imx-audmix.c -- 2.7.4

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[alsa-devel] Applied "ASoC: fsl: Add Audio Mixer machine driver" to the asoc tree
by Mark Brown 26 Mar '19

26 Mar '19

The patch ASoC: fsl: Add Audio Mixer machine driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b86ef5367761603df5f66ce08fb014b991f6b51d Mon Sep 17 00:00:00 2001 From: Viorel Suman <viorel.suman(a)nxp.com> Date: Tue, 22 Jan 2019 11:14:30 +0000 Subject: [PATCH] ASoC: fsl: Add Audio Mixer machine driver This patch implements Audio Mixer machine driver for NXP iMX8 SOCs. It connects together Audio Mixer and related SAI instances. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-audmix.c | 327 +++++++++++++++++++++++++++++++++++++ 3 files changed, 338 insertions(+) create mode 100644 sound/soc/fsl/imx-audmix.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0af2e056d211..d87c842806bd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -303,6 +303,15 @@ config SND_SOC_FSL_ASOC_CARD CS4271, CS4272 and SGTL5000. Say Y if you want to add support for Freescale Generic ASoC Sound Card. +config SND_SOC_IMX_AUDMIX + tristate "SoC Audio support for i.MX boards with AUDMIX" + select SND_SOC_FSL_AUDMIX + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with Audio Mixer + Say Y if you want to add support for SoC audio on an i.MX board with + an Audio Mixer. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4172d5a3e36c..c0dd04422fe9 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -62,6 +62,7 @@ snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o +snd-soc-imx-audmix-objs := imx-audmix.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o +obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c new file mode 100644 index 000000000000..72e37ca6cfbb --- /dev/null +++ b/sound/soc/fsl/imx-audmix.c @@ -0,0 +1,327 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright 2017 NXP + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/pm_runtime.h> +#include "fsl_sai.h" +#include "fsl_audmix.h" + +struct imx_audmix { + struct platform_device *pdev; + struct snd_soc_card card; + struct platform_device *audmix_pdev; + struct platform_device *out_pdev; + struct clk *cpu_mclk; + int num_dai; + struct snd_soc_dai_link *dai; + int num_dai_conf; + struct snd_soc_codec_conf *dai_conf; + int num_dapm_routes; + struct snd_soc_dapm_route *dapm_routes; +}; + +static const u32 imx_audmix_rates[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 64000, 96000, +}; + +static const struct snd_pcm_hw_constraint_list imx_audmix_rate_constraints = { + .count = ARRAY_SIZE(imx_audmix_rates), + .list = imx_audmix_rates, +}; + +static int imx_audmix_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_audmix *priv = snd_soc_card_get_drvdata(rtd->card); + struct snd_pcm_runtime *runtime = substream->runtime; + struct device *dev = rtd->card->dev; + unsigned long clk_rate = clk_get_rate(priv->cpu_mclk); + int ret; + + if (clk_rate % 24576000 == 0) { + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_audmix_rate_constraints); + if (ret < 0) + return ret; + } else { + dev_warn(dev, "mclk may be not supported %lu\n", clk_rate); + } + + ret = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, + 1, 8); + if (ret < 0) + return ret; + + return snd_pcm_hw_constraint_mask64(runtime, SNDRV_PCM_HW_PARAM_FORMAT, + FSL_AUDMIX_FORMATS); +} + +static int imx_audmix_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + u32 channels = params_channels(params); + int ret, dir; + + /* For playback the AUDMIX is slave, and for record is master */ + fmt |= tx ? SND_SOC_DAIFMT_CBS_CFS : SND_SOC_DAIFMT_CBM_CFM; + dir = tx ? SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + + /* set DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_MAST1, 0, dir); + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + /* + * Per datasheet, AUDMIX expects 8 slots and 32 bits + * for every slot in TDM mode. + */ + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, BIT(channels) - 1, + BIT(channels) - 1, 8, 32); + if (ret) + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + + return ret; +} + +static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF; + int ret; + + if (!tx) + return 0; + + /* For playback the AUDMIX is slave */ + fmt |= SND_SOC_DAIFMT_CBM_CFM; + + /* set AUDMIX DAI configuration */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt); + if (ret) + dev_err(dev, "failed to set AUDMIX DAI fmt: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops imx_audmix_fe_ops = { + .startup = imx_audmix_fe_startup, + .hw_params = imx_audmix_fe_hw_params, +}; + +static struct snd_soc_ops imx_audmix_be_ops = { + .hw_params = imx_audmix_be_hw_params, +}; + +static int imx_audmix_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *audmix_np = NULL, *out_cpu_np = NULL; + struct platform_device *audmix_pdev = NULL; + struct platform_device *cpu_pdev; + struct of_phandle_args args; + struct imx_audmix *priv; + int i, num_dai, ret; + const char *fe_name_pref = "HiFi-AUDMIX-FE-"; + char *be_name, *be_pb, *be_cp, *dai_name, *capture_dai_name; + + if (pdev->dev.parent) { + audmix_np = pdev->dev.parent->of_node; + } else { + dev_err(&pdev->dev, "Missing parent device.\n"); + return -EINVAL; + } + + if (!audmix_np) { + dev_err(&pdev->dev, "Missign DT node for parent device.\n"); + return -EINVAL; + } + + audmix_pdev = of_find_device_by_node(audmix_np); + if (!audmix_pdev) { + dev_err(&pdev->dev, "Missing AUDMIX platform device for %s\n", + np->full_name); + return -EINVAL; + } + + num_dai = of_count_phandle_with_args(audmix_np, "dais", NULL); + if (num_dai != FSL_AUDMIX_MAX_DAIS) { + dev_err(&pdev->dev, "Need 2 dais to be provided for %s\n", + audmix_np->full_name); + return -EINVAL; + } + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->num_dai = 2 * num_dai; + priv->dai = devm_kzalloc(&pdev->dev, priv->num_dai * + sizeof(struct snd_soc_dai_link), GFP_KERNEL); + if (!priv->dai) + return -ENOMEM; + + priv->num_dai_conf = num_dai; + priv->dai_conf = devm_kzalloc(&pdev->dev, priv->num_dai_conf * + sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!priv->dai_conf) + return -ENOMEM; + + priv->num_dapm_routes = 3 * num_dai; + priv->dapm_routes = devm_kzalloc(&pdev->dev, priv->num_dapm_routes * + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!priv->dapm_routes) + return -ENOMEM; + + for (i = 0; i < num_dai; i++) { + ret = of_parse_phandle_with_args(audmix_np, "dais", NULL, i, + &args); + if (ret < 0) { + dev_err(&pdev->dev, "of_parse_phandle_with_args failed\n"); + return ret; + } + + cpu_pdev = of_find_device_by_node(args.np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + + dai_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s%s", + fe_name_pref, args.np->full_name + 1); + + dev_info(pdev->dev.parent, "DAI FE name:%s\n", dai_name); + + if (i == 0) { + out_cpu_np = args.np; + capture_dai_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Capture"); + } + + priv->dai[i].name = dai_name; + priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; + priv->dai[i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[i].codec_name = "snd-soc-dummy"; + priv->dai[i].cpu_of_node = args.np; + priv->dai[i].cpu_dai_name = dev_name(&cpu_pdev->dev); + priv->dai[i].platform_of_node = args.np; + priv->dai[i].dynamic = 1; + priv->dai[i].dpcm_playback = 1; + priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); + priv->dai[i].ignore_pmdown_time = 1; + priv->dai[i].ops = &imx_audmix_fe_ops; + + /* Add AUDMIX Backend */ + be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "audmix-%d", i); + be_pb = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Playback-%d", i); + be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, + "AUDMIX-Capture-%d", i); + + priv->dai[num_dai + i].name = be_name; + priv->dai[num_dai + i].codec_dai_name = "snd-soc-dummy-dai"; + priv->dai[num_dai + i].codec_name = "snd-soc-dummy"; + priv->dai[num_dai + i].cpu_of_node = audmix_np; + priv->dai[num_dai + i].cpu_dai_name = be_name; + priv->dai[num_dai + i].platform_name = "snd-soc-dummy"; + priv->dai[num_dai + i].no_pcm = 1; + priv->dai[num_dai + i].dpcm_playback = 1; + priv->dai[num_dai + i].dpcm_capture = 1; + priv->dai[num_dai + i].ignore_pmdown_time = 1; + priv->dai[num_dai + i].ops = &imx_audmix_be_ops; + + priv->dai_conf[i].of_node = args.np; + priv->dai_conf[i].name_prefix = dai_name; + + priv->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + dai_name, "CPU-Playback"); + priv->dapm_routes[i].sink = be_pb; + priv->dapm_routes[num_dai + i].source = be_pb; + priv->dapm_routes[num_dai + i].sink = be_cp; + priv->dapm_routes[2 * num_dai + i].source = be_cp; + priv->dapm_routes[2 * num_dai + i].sink = capture_dai_name; + } + + cpu_pdev = of_find_device_by_node(out_cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find SAI platform device\n"); + return -EINVAL; + } + priv->cpu_mclk = devm_clk_get(&cpu_pdev->dev, "mclk1"); + if (IS_ERR(priv->cpu_mclk)) { + ret = PTR_ERR(priv->cpu_mclk); + dev_err(&cpu_pdev->dev, "failed to get DAI mclk1: %d\n", ret); + return -EINVAL; + } + + priv->audmix_pdev = audmix_pdev; + priv->out_pdev = cpu_pdev; + + priv->card.dai_link = priv->dai; + priv->card.num_links = priv->num_dai; + priv->card.codec_conf = priv->dai_conf; + priv->card.num_configs = priv->num_dai_conf; + priv->card.dapm_routes = priv->dapm_routes; + priv->card.num_dapm_routes = priv->num_dapm_routes; + priv->card.dev = pdev->dev.parent; + priv->card.owner = THIS_MODULE; + priv->card.name = "imx-audmix"; + + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); + + ret = devm_snd_soc_register_card(pdev->dev.parent, &priv->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed\n"); + return ret; + } + + return ret; +} + +static struct platform_driver imx_audmix_driver = { + .probe = imx_audmix_probe, + .driver = { + .name = "imx-audmix", + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(imx_audmix_driver); + +MODULE_DESCRIPTION("NXP AUDMIX ASoC machine driver"); +MODULE_AUTHOR("Viorel Suman <viorel.suman(a)nxp.com>"); +MODULE_ALIAS("platform:imx-audmix"); +MODULE_LICENSE("GPL v2"); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Only enable with common clock
by Mark Brown 26 Mar '19

26 Mar '19

Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..5eafadc855bf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C -- 2.20.1

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[alsa-devel] [PATCH v4] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 26 Mar '19

26 Mar '19

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. V2 :- o Max wait calc now in ms. o checkpatch. v3:- o Remove driver supplied time validation. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 17 ++++++++++++----- 2 files changed, 13 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..fcdf358a25f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + long wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..eecbfdc7a2c9 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,19 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10; + + if (runtime->rate) { + long t = runtime->period_size * 2 / + runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.17.1

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Re: [alsa-devel] 'modern dailink' transition
by Kuninori Morimoto 26 Mar '19

26 Mar '19

Oops ?? I exchanged mail address - alsa-devel(a)alsa-devel.org + alsa-devel(a)alsa-project.org > Hi Pierre, Mark > > > > I am however struggling with the notion of a 'snd-soc-dummy' platform that > > > exists in some legacy Intel machine drivers. I changed the code following > > > the pattern below but I have really no idea if this is correct. Shouldn't > > > all dailinks either point to a real platform driver or not provide any > > > information about the platform at all? Is there any specific expectation on > > > the ASoC side here? > > I guess my posted "no Platform" and "no implicit snd-soc-dummy" patch idea > confused you. If so, I'm so sorry about that. > I'm not sure this idea is Good or Bad. > > > I'd expect the dummy driver to just get automatically substituted when > > required, I'd not expect users to explicitly list it. > > Yes agree. > > This is my understanding, please correct me if I was wrong. > I think current many sound card which doesn't need "platfrom" are 2 patterns. > > 1) select snd-soc-dummy as platfrom > 2) select cpu component as platfrom > > Current ASoC selects 1) automatically if .platfrom_name was NULL. > And driver needs to have below if it want to be 2) > > dai_link->platform_of_node = dai_link->cpu_of_node > > But, I think one of them is enough ? > I mean select 2) automatically can be OK? > In other words, current some sound card which doesn't need > platfrom is calling snd-soc-dummy platfrom method in 1) case. > But, is it needed ? I'm not sure... > > It seems snd-soc-dummy platfrom is caring about DPCM-BE case, > but, I think CPU is snd-soc-dummy in such case. > Maybe we need same cade to dummy CPU (?), but *my* DPCM system > is working correctly without it. > > Best regards > --- > Kuninori Morimoto

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[alsa-devel] [asoc:for-5.2 73/82] sound/soc/codecs/tlv320aic32x4-clk.c:258:2: error: unknown field 'prepare' specified in initializer
by kbuild test robot 25 Mar '19

25 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 6d56ee1550b8a81bc63c80051ff78d8d704b09ba commit: 514b044cba667e4b7c383ec79b42b997e624b91d [73/82] ASoC: tlv320aic32x4: Model PLL in CCF config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 514b044cba667e4b7c383ec79b42b997e624b91d # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/codecs/tlv320aic32x4-clk.c:19:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ In file included from include/linux/init.h:5:0, from include/linux/io.h:22, from include/linux/clk-provider.h:9, from sound/soc/codecs/tlv320aic32x4-clk.c:10: sound/soc/codecs/tlv320aic32x4-clk.c: In function 'clk_aic32x4_pll_prepare': include/linux/kernel.h:979:32: error: dereferencing pointer to incomplete type 'struct clk_hw' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~ include/linux/compiler.h:324:9: note: in definition of macro '__compiletime_assert' if (!(condition)) \ ^~~~~~~~~ include/linux/compiler.h:344:2: note: in expansion of macro '_compiletime_assert' _compiletime_assert(condition, msg, __compiletime_assert_, __LINE__) ^~~~~~~~~~~~~~~~~~~ include/linux/build_bug.h:39:37: note: in expansion of macro 'compiletime_assert' #define BUILD_BUG_ON_MSG(cond, msg) compiletime_assert(!(cond), msg) ^~~~~~~~~~~~~~~~~~ include/linux/kernel.h:979:2: note: in expansion of macro 'BUILD_BUG_ON_MSG' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~~~~~~ include/linux/kernel.h:979:20: note: in expansion of macro '__same_type' BUILD_BUG_ON_MSG(!__same_type(*(ptr), ((type *)0)->member) && \ ^~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:17:29: note: in expansion of macro 'container_of' #define to_clk_aic32x4(_hw) container_of(_hw, struct clk_aic32x4, hw) ^~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:49:28: note: in expansion of macro 'to_clk_aic32x4' struct clk_aic32x4 *pll = to_clk_aic32x4(hw); ^~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:21: error: variable 'aic32x4_pll_ops' has initializer but incomplete type static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:258:2: error: unknown field 'prepare' specified in initializer .prepare = clk_aic32x4_pll_prepare, ^ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: warning: excess elements in struct initializer .prepare = clk_aic32x4_pll_prepare, ^~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:258:13: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:259:2: error: unknown field 'unprepare' specified in initializer .unprepare = clk_aic32x4_pll_unprepare, ^ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: warning: excess elements in struct initializer .unprepare = clk_aic32x4_pll_unprepare, ^~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:259:15: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:260:2: error: unknown field 'is_prepared' specified in initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: warning: excess elements in struct initializer .is_prepared = clk_aic32x4_pll_is_prepared, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:260:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:261:2: error: unknown field 'recalc_rate' specified in initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: warning: excess elements in struct initializer .recalc_rate = clk_aic32x4_pll_recalc_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:261:17: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:262:2: error: unknown field 'round_rate' specified in initializer .round_rate = clk_aic32x4_pll_round_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: warning: excess elements in struct initializer .round_rate = clk_aic32x4_pll_round_rate, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:262:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:263:2: error: unknown field 'set_rate' specified in initializer .set_rate = clk_aic32x4_pll_set_rate, ^ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: warning: excess elements in struct initializer .set_rate = clk_aic32x4_pll_set_rate, ^~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:263:14: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:264:2: error: unknown field 'set_parent' specified in initializer .set_parent = clk_aic32x4_pll_set_parent, ^ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: warning: excess elements in struct initializer .set_parent = clk_aic32x4_pll_set_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:264:16: note: (near initialization for 'aic32x4_pll_ops') >> sound/soc/codecs/tlv320aic32x4-clk.c:265:2: error: unknown field 'get_parent' specified in initializer .get_parent = clk_aic32x4_pll_get_parent, ^ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: warning: excess elements in struct initializer .get_parent = clk_aic32x4_pll_get_parent, ^~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:265:16: note: (near initialization for 'aic32x4_pll_ops') sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:282:23: error: storage size of 'init' isn't known struct clk_init_data init; ^~~~ >> sound/soc/codecs/tlv320aic32x4-clk.c:302:9: error: implicit declaration of function 'devm_clk_register' [-Werror=implicit-function-declaration] return devm_clk_register(dev, &priv->hw); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c:282:23: warning: unused variable 'init' [-Wunused-variable] struct clk_init_data init; ^~~~ sound/soc/codecs/tlv320aic32x4-clk.c: At top level: sound/soc/codecs/tlv320aic32x4-clk.c:257:29: error: storage size of 'aic32x4_pll_ops' isn't known static const struct clk_ops aic32x4_pll_ops = { ^~~~~~~~~~~~~~~ sound/soc/codecs/tlv320aic32x4-clk.c: In function 'aic32x4_register_clk': sound/soc/codecs/tlv320aic32x4-clk.c:303:1: warning: control reaches end of non-void function [-Wreturn-type] } ^ cc1: some warnings being treated as errors vim +/prepare +258 sound/soc/codecs/tlv320aic32x4-clk.c 255 256 257 static const struct clk_ops aic32x4_pll_ops = { > 258 .prepare = clk_aic32x4_pll_prepare, > 259 .unprepare = clk_aic32x4_pll_unprepare, > 260 .is_prepared = clk_aic32x4_pll_is_prepared, > 261 .recalc_rate = clk_aic32x4_pll_recalc_rate, > 262 .round_rate = clk_aic32x4_pll_round_rate, > 263 .set_rate = clk_aic32x4_pll_set_rate, > 264 .set_parent = clk_aic32x4_pll_set_parent, > 265 .get_parent = clk_aic32x4_pll_get_parent, 266 }; 267 268 static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { 269 { 270 .name = "pll", 271 .parent_names = 272 (const char* []) { "mclk", "bclk", "gpio", "din" }, 273 .num_parents = 4, 274 .ops = &aic32x4_pll_ops, 275 .reg = 0, 276 }, 277 }; 278 279 static struct clk *aic32x4_register_clk(struct device *dev, 280 struct aic32x4_clkdesc *desc) 281 { 282 struct clk_init_data init; 283 struct clk_aic32x4 *priv; 284 const char *devname = dev_name(dev); 285 286 init.ops = desc->ops; 287 init.name = desc->name; 288 init.parent_names = desc->parent_names; 289 init.num_parents = desc->num_parents; 290 init.flags = 0; 291 292 priv = devm_kzalloc(dev, sizeof(struct clk_aic32x4), GFP_KERNEL); 293 if (priv == NULL) 294 return (struct clk *) -ENOMEM; 295 296 priv->dev = dev; 297 priv->hw.init = &init; 298 priv->regmap = dev_get_regmap(dev, NULL); 299 priv->reg = desc->reg; 300 301 clk_hw_register_clkdev(&priv->hw, desc->name, devname); > 302 return devm_clk_register(dev, &priv->hw); 303 } 304 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] sound: codecs: fix a potential NULL pointer dereference
by Kangjie Lu 25 Mar '19

25 Mar '19

In case devm_kzalloc fails, the patch returns -ENOMEM to avoid potential NULL pointer dereference. Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> --- sound/soc/codecs/rt5663.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index da6647015708..909ab99a1995 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3480,6 +3480,8 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + if (!rt5663->imp_table) + return -ENOMEM; device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); -- 2.17.1

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