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

[alsa-devel] aplay\arecord compare utility
by Muni Sekhar 27 Feb '18

27 Feb '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, I’m using an alsa utilities(aplay & arecord) for sound loopback testing. From Linux host , sending & receiving the wav file. At end I want to compare the sent & received wav file. Does Linux has any utility to support this one? -- Thanks, Sekhar

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[alsa-devel] [PATCH] ALSA: Use scnprintf() instead of snprintf()
by Jaejoong Kim 27 Feb '18

27 Feb '18

The show() method should use scnprintf() not snprintf() because snprintf() may returns a value that exceeds its second argument. Signed-off-by: Jaejoong Kim <climbbb.kim(a)gmail.com> --- sound/core/init.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 168ae03..4a51a06 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -670,7 +670,7 @@ card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%s\n", card->id); + return scnprintf(buf, PAGE_SIZE, "%s\n", card->id); } static ssize_t @@ -710,7 +710,7 @@ card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); - return snprintf(buf, PAGE_SIZE, "%i\n", card->number); + return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); -- 2.7.4

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[alsa-devel] No sound on MacbookPro 13,3 (A1707)
by Freaky 27 Feb '18

27 Feb '18

Hi there, we're trying to get sound working on a MacbookPro 13,3 (A1707). So far I've been unable to get the sound working. Did use the HDA analyzer and flipped the out-dir, enable and data on all 8 GPIO's, but none of them got the sound working on either the onboard speakers nor the jack output for headphone. Is there anyone willing to help on this issue? Driver development is a bit beyond my skills. I can provide hands, debug logs, etc. tho' :). Currently trying with a default Ubuntu 17.10.1 live USB, but a 4.15.3 kernel didn't help either. Alsa-info has been uploaded to http://www.alsa-project.org/db/?f=33cc432d3b648d1aa6adc997630817046656a086 The system was booted with an external USB-C dock from Dell, otherwise there's no network and no regular (USB-A) ports available (booting from stick) as the WiFi doesn't work and the device has only USB-C ports. The card 0 is from the dock - it's not relevant (and working). Before alsa-info ran there was also a USB headphone plugged in, that was card 1. Without those devices card 2 = 0 and 3 = 1 at boot. Issues are with what's listed as card 2 currently, the HDA-Intel - HDA Intel PCH. Any help is greatly appreciated. Some people already have some trackers for the current state of the devices. Not sure if they are of use, but might help: https://github.com/Dunedan/mbp-2016-linux/ https://gist.github.com/roadrunner2/1289542a748d9a104e7baec6a92f9cd7 Kind regards,

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[alsa-devel] [PATCH v3] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 27 Feb '18

27 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 v2: - Also remove the dt-bindings .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 ---- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------- 4 files changed, 378 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt deleted file mode 100644 index acea71b..0000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ /dev/null @@ -1,53 +0,0 @@ -Freescale i.MX audio complex with WM8962 codec - -Required properties: - - - compatible : "fsl,imx-audio-wm8962" - - - model : The user-visible name of this sound complex - - - ssi-controller : The phandle of the i.MX SSI controller - - - audio-codec : The phandle of the WM8962 audio codec - - - audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names could be power supplies, WM8962 - pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx6q-sabresd-wm8962", - "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "MICBIAS", "AMIC", - "IN3R", "MICBIAS", - "DMIC", "MICBIAS", - "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; -}; 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] 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.

2 3

[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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