February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling"" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling" 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 b84acf44d5b86597cebcb6aca16e7dd9ca2b0294 Mon Sep 17 00:00:00 2001 From: Fabio Estevam <fabio.estevam(a)nxp.com> Date: Mon, 26 Feb 2018 15:55:25 -0300 Subject: [PATCH] ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling" This reverts commit f91b1e73ccde71d4bc69ae10d475196df38844ab. As explained by Lars-Peter Clausen: "This creates a resource leak. dmaengine_pcm_request_chan_of() requests bothtransmit and receive channels. It might return with an error if one of them fails, but the other one succeeded. In this case we need to call dmaengine_pcm_release_chan() to free the requested channel" So revert the commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 567fbdfd1ca9..56a541b9ff9e 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -504,7 +504,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_pcm; + goto err_free_dma; if (config && config->process) ret = snd_soc_add_component(dev, &pcm->component, @@ -520,7 +520,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); -err_free_pcm: kfree(pcm); return ret; } -- 2.16.1

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[alsa-devel] [PATCH v2] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Instead of fixing the imx-wm8962 driver private strcuc, better remove it as wm8962 can be handled by fsl-asoc-card. (Nicolin) sound/soc/fsl/Kconfig | 11 -- sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------------------------------- 3 files changed, 325 deletions(-) delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * 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/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: imx-wm8962: Use a single private structure
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 8f7206d69ab8c ("ASoC: imx-wm8962: Remove global variables") ended up by assigning the same value for two different structs: struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); Fix it by consolidating all the private data into a single structure. Fixes: 8f7206d69ab8c ("ASoC: imx-wm8962: Remove global variables") Reported-by: Leonard Crestez <leonard.crestez(a)nxp.com> Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/imx-wm8962.c | 64 +++++++++++++++++++--------------------------- 1 file changed, 26 insertions(+), 38 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 206b898..29d7e9b 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -28,15 +28,12 @@ #define DAI_NAME_SIZE 32 -struct imx_wm8962_data { +struct imx_priv { struct snd_soc_dai_link dai; struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; unsigned int clk_frequency; -}; - -struct imx_priv { struct platform_device *pdev; int sample_rate; snd_pcm_format_t sample_format; @@ -72,7 +69,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -91,7 +87,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, pll_out = priv->sample_rate * 256; ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, + WM8962_FLL_MCLK, priv->clk_frequency, pll_out); if (ret < 0) { dev_err(dev, "failed to start FLL: %d\n", ret); @@ -111,7 +107,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, case SND_SOC_BIAS_STANDBY: if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, + WM8962_SYSCLK_MCLK, priv->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) { dev_err(dev, @@ -141,14 +137,13 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); + priv->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) dev_err(dev, "failed to set sysclk in %s\n", __func__); @@ -161,7 +156,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_wm8962_data *data; struct imx_priv *priv; struct clk *codec_clk; int int_port, ext_port; @@ -232,12 +226,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(codec_clk)) { ret = PTR_ERR(codec_clk); @@ -245,39 +233,39 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data->clk_frequency = clk_get_rate(codec_clk); + priv->clk_frequency = clk_get_rate(codec_clk); clk_put(codec_clk); - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + priv->dai.name = "HiFi"; + priv->dai.stream_name = "HiFi"; + priv->dai.codec_dai_name = "wm8962"; + priv->dai.codec_of_node = codec_np; + priv->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + priv->dai.platform_of_node = ssi_np; + priv->dai.ops = &imx_hifi_ops; + priv->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); + priv->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&priv->card, "model"); if (ret) goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); if (ret) goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + priv->card.num_links = 1; + priv->card.owner = THIS_MODULE; + priv->card.dai_link = &priv->dai; + priv->card.dapm_widgets = imx_wm8962_dapm_widgets; + priv->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; + priv->card.late_probe = imx_wm8962_late_probe; + priv->card.set_bias_level = imx_wm8962_set_bias_level; - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); + platform_set_drvdata(pdev, &priv->card); + snd_soc_card_set_drvdata(&priv->card, priv); - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Revert "ASoC: soc-generic-dmaengine-pcm: Fix error handling"
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> This reverts commit f91b1e73ccde71d4bc69ae10d475196df38844ab. As explained by Lars-Peter Clausen: "This creates a resource leak. dmaengine_pcm_request_chan_of() requests bothtransmit and receive channels. It might return with an error if one of them fails, but the other one succeeded. In this case we need to call dmaengine_pcm_release_chan() to free the requested channel" So revert the commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 32ea16d..768247f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -450,7 +450,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_pcm; + goto err_free_dma; ret = snd_soc_add_component(dev, &pcm->component, &dmaengine_pcm_component, NULL, 0); @@ -461,7 +461,6 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); -err_free_pcm: kfree(pcm); return ret; } -- 2.7.4

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[alsa-devel] Commit 8f7206d69a ("ASoC: imx-wm8962: Remove global variables") looks strange
by Daniel Baluta 26 Feb '18

26 Feb '18

Hi Fabio, Leonard (Cc'ed) noticed that commit in $subject looks very strange. For example: @@ -139,7 +140,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct imx_priv *priv = &card_priv; + struct imx_priv *priv = snd_soc_card_get_drvdata(card); struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); So, now data and priv point to the same memory area but they are completely different structures! Unless we are missing something this is obviously a bug. I will try to have a look at the end of the week, but any comments are welcome. I think the fix would be to have one global priv structure and to unify imx_wm8962_data with imx_priv. thanks, Daniel.

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[alsa-devel] [PATCH] ALSA: hda - Fix pincfg at resume on Lenovo T470 dock
by Takashi Iwai 26 Feb '18

26 Feb '18

We've added a quirk to enable the recent Lenovo dock support, where it overwrites the pin configs of NID 0x17 and 19, not only updating the pin config cache. It works right after the boot, but the problem is that the pin configs are occasionally cleared when the machine goes to PM. Meanwhile the quirk writes the pin configs only at the pre-probe, so this won't be applied any longer. For addressing that issue, this patch moves the code to overwrite the pin configs into HDA_FIXUP_ACT_INIT section so that it's always applied at both probe and resume time. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195161 Fixes: 61fcf8ece9b6 ("ALSA: hda/realtek - Enable Thinkpad Dock device for ALC298 platform") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce28f7ce64e6..b9c93fa0a51c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4997,13 +4997,14 @@ static void alc_fixup_tpt470_dock(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + snd_hda_apply_pincfgs(codec, pincfgs); + } else if (action == HDA_FIXUP_ACT_INIT) { /* Enable DOCK device */ snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0); /* Enable DOCK device */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0); - snd_hda_apply_pincfgs(codec, pincfgs); } } -- 2.16.2

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[alsa-devel] [PATCH 1/2] ASoC: soc-generic-dmaengine-pcm: Fix error handling
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> When dmaengine_pcm_request_chan_of() fails it should release the previously acquired resources, which in this case is to call kfree(pcm), so jump to the correct point in the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 768247f..32ea16d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -450,7 +450,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_dma; + goto err_free_pcm; ret = snd_soc_add_component(dev, &pcm->component, &dmaengine_pcm_component, NULL, 0); @@ -461,6 +461,7 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); +err_free_pcm: kfree(pcm); return ret; } -- 2.7.4

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[alsa-devel] [RFC PATCH 0/3] ASoC: stm32: add S/PDIF support to SAI
by Olivier Moysan 26 Feb '18

26 Feb '18

This patch-set adds S/PDIF IEC6958 protocol support to STM32 SAI. Only S/PDIF playback is supported here, as SAI peripheral does not support S/PDIF capture. SAI peripheral does not take as input full IEC60958 frames as it generates preamble and parity bit. However, it cannot handle raw PCM data, as it is not designed to insert channel status and user data in the generated IEC frames. Moreover, SAI data register format does not match exactly IEC60958 subframe format. (see figure below). Channel status (C) and User data (U) are at 26-25 bits position, in data register, instead of 30-29 bits position. IEC60958 subframe bits: 31-28 27-4 3-0 P C U V data (16|24 bits) preamble SAI data register format: 27-24 23-0 P C U V data (16|24 bits) The impact in software is that the data have to be processed before DMA transfer. These constraints should be managed at driver level to minimize the impact on application. This means that a callback is necessary to perform data processing (shift and channel status insertion) This patch-set introduces a callback to allow processing on samples. The implementation is based on previous discussions available here: https://patchwork.kernel.org/patch/9570255/ BRs Olivier Olivier Moysan (3): ASoC: stm32: Add S/PDIF to SAI bindings ASoC: dmaengine_pcm: add processing support ASoC: stm32: sai: Add support of S/PDIF playback .../devicetree/bindings/sound/st,stm32-sai.txt | 6 + include/sound/dmaengine_pcm.h | 3 + sound/soc/soc-generic-dmaengine-pcm.c | 62 ++++++++- sound/soc/stm/stm32_sai.c | 2 + sound/soc/stm/stm32_sai.h | 2 + sound/soc/stm/stm32_sai_sub.c | 153 +++++++++++++++++---- 6 files changed, 202 insertions(+), 26 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: uniphier: fix broken sound if use SRC in replay
by Katsuhiro Suzuki 26 Feb '18

26 Feb '18

This patch fixes settings for ports with SRC. These ports need to set the fixed audio rate and clock. If not, the SRC outputs broken sound. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 1e5f053d9f6b..1711361fc0c2 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -501,7 +501,7 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) OPORTMXCTR2_MSSEL_MASTER | OPORTMXCTR2_EXTLSIFSSEL_36 | OPORTMXCTR2_DACCKSEL_1_2; - } else { + } else if (sub->swm->type == PORT_TYPE_SPDIF) { if (sub->aio->pll_out >= ARRAY_SIZE(v_pll)) { dev_err(dev, "PLL(%d) is invalid\n", sub->aio->pll_out); @@ -521,6 +521,11 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) v |= OPORTMXCTR2_EXTLSIFSSEL_24; break; } + } else { + v = OPORTMXCTR2_ACLKSEL_A1 | + OPORTMXCTR2_MSSEL_MASTER | + OPORTMXCTR2_EXTLSIFSSEL_36 | + OPORTMXCTR2_DACCKSEL_1_2; } regmap_write(r, OPORTMXCTR2(sub->swm->oport.map), v); } else { @@ -550,11 +555,19 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params) { struct regmap *r = sub->aio->chip->regmap; + unsigned int rate; u32 v; int ret; if (!pass_through) { - ret = aio_port_set_rate(sub, params_rate(params)); + if (sub->swm->type == PORT_TYPE_EVE || + sub->swm->type == PORT_TYPE_CONV) { + rate = 48000; + } else { + rate = params_rate(params); + } + + ret = aio_port_set_rate(sub, rate); if (ret) return ret; -- 2.16.1

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[alsa-devel] [PATCH] ASoC: sirf: remove duplicated bit-wise or of USP_RXFIFO_THD_INT
by Colin King 26 Feb '18

26 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> Bit pattern USP_RXFIFO_THD_INT is being bit-wise or'd twice; remove the redundant 2nd USP_RXFIFO_THD_INT Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sirf/sirf-usp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h index bf0201cb15bc..e22e13a619e8 100644 --- a/sound/soc/sirf/sirf-usp.h +++ b/sound/soc/sirf/sirf-usp.h @@ -154,7 +154,7 @@ USP_RX_IO_DMA_INT|\ USP_RXFIFO_FULL_INT|\ USP_RXFIFO_THD_INT|\ - USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + USP_RX_TIMEOUT_INT) #define USP_INT_ALL 0x1FFF -- 2.15.1

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