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

Re: [alsa-devel] [PATCH] ASoC: mc13783: Add devicetree support
by Lee Jones 30 Apr '14

30 Apr '14

> > > > > > This patch adds devicetree support for mc13783-codec. > > > > > > > > > > Applied, thanks. > > > > > > > > No Mark, no! > > > > > > What a problem, Lee? > > > > There's no problem with the patch, don't worry. > > > > Neither Mark, nor I can apply this patch independently, we need to > > create a shared branch somewhere, or we'll risk conflicts at merge > > time. > > I can separate this patch for ASoC and MFD if need. > But this will work only if both is applied. Are there build dependencies, or only functional ones? -- Lee Jones Linaro STMicroelectronics Landing Team Lead Linaro.org │ Open source software for ARM SoCs Follow Linaro: Facebook | Twitter | Blog

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[alsa-devel] [PATCH] ASoC: mc13783: Add devicetree support
by Alexander Shiyan 30 Apr '14

30 Apr '14

This patch adds devicetree support for mc13783-codec. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- Documentation/devicetree/bindings/mfd/mc13xxx.txt | 3 +++ drivers/mfd/mc13xxx-core.c | 10 +++++++--- sound/soc/codecs/mc13783.c | 14 +++++++++++++- 3 files changed, 23 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/mfd/mc13xxx.txt b/Documentation/devicetree/bindings/mfd/mc13xxx.txt index 1413f39..8aba488 100644 --- a/Documentation/devicetree/bindings/mfd/mc13xxx.txt +++ b/Documentation/devicetree/bindings/mfd/mc13xxx.txt @@ -10,6 +10,9 @@ Optional properties: - fsl,mc13xxx-uses-touch : Indicate the touchscreen controller is being used Sub-nodes: +- codec: Contain the Audio Codec node. + - adc-port: Contain PMIC SSI port number used for ADC. + - dac-port: Contain PMIC SSI port number used for DAC. - leds : Contain the led nodes and initial register values in property "led-control". Number of register depends of used IC, for MC13783 is 6, for MC13892 is 4, for MC34708 is 1. See datasheet for bits definitions of diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 06e64b6..0c6c21c 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -673,9 +673,13 @@ int mc13xxx_common_init(struct device *dev) if (mc13xxx->flags & MC13XXX_USE_ADC) mc13xxx_add_subdevice(mc13xxx, "%s-adc"); - if (mc13xxx->flags & MC13XXX_USE_CODEC) - mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec", - pdata->codec, sizeof(*pdata->codec)); + if (mc13xxx->flags & MC13XXX_USE_CODEC) { + if (pdata) + mc13xxx_add_subdevice_pdata(mc13xxx, "%s-codec", + pdata->codec, sizeof(*pdata->codec)); + else + mc13xxx_add_subdevice(mc13xxx, "%s-codec"); + } if (mc13xxx->flags & MC13XXX_USE_RTC) mc13xxx_add_subdevice(mc13xxx, "%s-rtc"); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 0b0a4ca..9965277 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -22,6 +22,7 @@ */ #include <linux/module.h> #include <linux/device.h> +#include <linux/of.h> #include <linux/mfd/mc13xxx.h> #include <linux/slab.h> #include <sound/core.h> @@ -748,6 +749,7 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) { struct mc13783_priv *priv; struct mc13xxx_codec_platform_data *pdata = pdev->dev.platform_data; + struct device_node *np; int ret; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -758,7 +760,17 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) priv->adc_ssi_port = pdata->adc_ssi_port; priv->dac_ssi_port = pdata->dac_ssi_port; } else { - return -ENOSYS; + np = of_get_child_by_name(pdev->dev.parent->of_node, "codec"); + if (!np) + return -ENOSYS; + + ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); + if (ret) + return ret; + + ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); + if (ret) + return ret; } dev_set_drvdata(&pdev->dev, priv); -- 1.8.3.2

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[alsa-devel] Control the exact moment of output
by Rob Janssen 30 Apr '14

30 Apr '14

For a distributed system that requires synchronized output I would like to determine the exact moment when output samples are sent, preferably within +/- 1 sample time. For example, output is sent in blocks with a system time in usec or nsec as a parameter or header, and the the first sample leaves the soundcard as closely as possible to the specified system time. Is this possible within the ALSA API? Low latency is of course nice to have, but not the prime concern. What I need is synchronous output between systems that are physically separated and need to send audio that they receive from a central system over the network. My intention is to send blocks of audio with an accurate time of transmission in the header, and then have the systems output it exactly at that time, of course preferably managed mostly by the driver, not the application. Are some types of soundcard more suitable than others for this purpose? And are there soundcards available where the sample clock can somehow be locked to system time or an external 10MHz/1PPS reference? Rob

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[alsa-devel] [PATCH v3 00/18] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 30 Apr '14

30 Apr '14

Hi, This series is a cleanup of the fsl-ssi driver. There is no DT binding for regmap yet, as this should be solved in the regmap framework first. The current configuration should at least work on the same SoCs it was previously used. As soon as we have a common DT binding to set the endianess we replace it in this driver. Best regards, Markus Changes in v3: - Some new patches to improve/fix i2s master mode by Sascha - baudclock is enabled/disabled in startup/shutdown now - bitclock setup moved to a seperate function (not set_dai_sysclk) - Regmap config changed to NATIVE now. Markus Pargmann (11): ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: make fsl,mode property optional ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Use regmap Sascha Hauer (7): ASoC: fsl-ssi: introduce SoC specific data ASoC: fsl-ssi: Only enable baudclk when used ASoC: fsl-ssi: Move fsl_ssi_set_dai_sysclk above fsl_ssi_hw_params ASoC: fsl-ssi: set bitclock in master mode from hw_params ASoC: fsl-ssi: remove unnecessary spinlock ASoC: fsl-ssi: Allow first stream to set the bitclock ASoC: fsl-ssi: Set framerate divider correctly for i2s master mode sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 3 +- sound/soc/fsl/fsl_ssi.c | 1249 +++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 112 +++- sound/soc/fsl/fsl_ssi_dbg.c | 163 ++++++ 5 files changed, 807 insertions(+), 721 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.9.1

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[alsa-devel] abdul, please add me to your LinkedIn network
by Raffaele Recalcati via LinkedIn 30 Apr '14

30 Apr '14

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[alsa-devel] [PATCH V3] ASoC: samsung: Add sound card driver for Snow board
by Tushar Behera 29 Apr '14

29 Apr '14

Added machine driver to instantiate I2S based sound card on Snow board. It has MAX98095 audio codec on board. There are some other variants for Snow board which have MAX98090 audio codec. Hence support for MAX98090 is also added to this driver. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Changes for V3: 1. Updated documentation (I2S0 -> I2S) 2. Removed 'Capture' DAI and renamed 'Playback' DAI as 'Primary'. 3. Updated patch subject to match sub-system naming style. Changes for V2: Addressed various comments from Mark Brown. 1. Moved DAIFMT flags to dai_link. Removed snd_soc_dai_set_fmt calls. 2. Removed BCLK settings. 3. Moved sysclk settings to late_probe. 4. Removed now empty snow_hw_params. 5. Removed snow_init 6. Used devm_snd_soc_register_card. Also removed empty remove function. 7. Removed non-DT handling as the driver is DT only. Additional nits: 1. Renamed snow_driver_probe to snow_probe 2. Removed unused pm_ops. 3. Updated clock settings for I2S block (Earlier it was I2S_CDCLK, whereas it should be I2S_RCLKSRC_0). 4. Removed 'card-name' DT property as it was not used in earlier sound-card driver DT bindings. Will be added later. Documentation/devicetree/bindings/sound/snow.txt | 17 +++ sound/soc/samsung/Kconfig | 10 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/snow.c | 122 ++++++++++++++++++++++ 4 files changed, 151 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/snow.txt create mode 100644 sound/soc/samsung/snow.c diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt new file mode 100644 index 0000000..678b191 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -0,0 +1,17 @@ +Audio Binding for Snow boards + +Required properties: +- compatible : Can be one of the following, + "google,snow-audio-max98090" or + "google,snow-audio-max98095" +- samsung,i2s-controller: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Example: + +sound { + compatible = "google,snow-audio-max98095"; + + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98095>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f2e2891..50aa289 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -231,3 +231,13 @@ config SND_SOC_LITTLEMILL select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 + +config SND_SOC_SNOW + tristate "Audio support for Google Snow boards" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SOC_MAX98095 + select SND_SAMSUNG_I2S + help + Say Y if you want to add audio support for various Snow + boards based on Exynos5 series of SoCs. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 86715d8..6d0212b 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -34,6 +34,7 @@ snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm8994-objs := smdk_wm8994.o +snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o @@ -58,6 +59,7 @@ obj-$(CONFIG_SND_SOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o +obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c new file mode 100644 index 0000000..0fa89a4 --- /dev/null +++ b/sound/soc/samsung/snow.c @@ -0,0 +1,122 @@ +/* + * ASoC machine driver for Snow boards + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <linux/of_device.h> + +#include <sound/soc.h> + +#include "i2s.h" + +#define FIN_PLL_RATE 24000000 + +static struct snd_soc_dai_link snow_dai[] = { + { + .name = "Primary", + .stream_name = "Primary", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static int snow_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + int ret; + + /* Set the MCLK rate for the codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + FIN_PLL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Select I2S Bus clock to set RCLK and BCLK */ + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_card snow_snd = { + .name = "Snow-I2S", + .dai_link = snow_dai, + .num_links = ARRAY_SIZE(snow_dai), + + .late_probe = snow_late_probe, +}; + +static int snow_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snow_snd; + struct device_node *i2s_node, *codec_node; + int i, ret; + + i2s_node = of_parse_phandle(pdev->dev.of_node, + "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Property 'i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + codec_node = of_parse_phandle(pdev->dev.of_node, + "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(snow_dai); i++) { + snow_dai[i].codec_of_node = codec_node; + snow_dai[i].cpu_of_node = i2s_node; + snow_dai[i].platform_of_node = i2s_node; + } + + card->dev = &pdev->dev; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + return ret; +} + +static const struct of_device_id snow_of_match[] = { + { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98095", }, + {}, +}; + +static struct platform_driver snow_driver = { + .driver = { + .name = "snow-audio", + .owner = THIS_MODULE, + .of_match_table = snow_of_match, + }, + .probe = snow_probe, +}; + +module_platform_driver(snow_driver); + +MODULE_DESCRIPTION("ALSA SoC Audio machine driver for Snow"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

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[alsa-devel] [PATCH v2] ASoC: fsl_spdif: Fix clock source for rxclk rate measurement
by Nicolin Chen 29 Apr '14

29 Apr '14

The rxclk rate actually uses sysclk, ipg clock for example, as its reference clock to calculate it. But the driver currently doesn't pass a correct clock source. So fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Changelog v2: * Use fixed name for sysclk fetching. sound/soc/fsl/fsl_spdif.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e2836b3..e7be6b3 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -81,6 +81,7 @@ struct fsl_spdif_priv { struct clk *txclk[SPDIF_TXRATE_MAX]; struct clk *rxclk; struct clk *coreclk; + struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -767,7 +768,7 @@ static int spdif_get_rxclk_rate(struct fsl_spdif_priv *spdif_priv, clksrc = (phaseconf >> SRPC_CLKSRC_SEL_OFFSET) & 0xf; if (srpc_dpll_locked[clksrc] && (phaseconf & SRPC_DPLL_LOCKED)) { /* Get bus clock from system */ - busclk_freq = clk_get_rate(spdif_priv->rxclk); + busclk_freq = clk_get_rate(spdif_priv->sysclk); } /* FreqMeas_CLK = (BUS_CLK * FreqMeas) / 2 ^ 10 / GAINSEL / 128 */ @@ -1147,6 +1148,13 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } + /* Get system clock for rx clock rate calculation */ + spdif_priv->sysclk = devm_clk_get(&pdev->dev, "rxtx5"); + if (IS_ERR(spdif_priv->sysclk)) { + dev_err(&pdev->dev, "no sys clock (rxtx5) in devicetree\n"); + return PTR_ERR(spdif_priv->sysclk); + } + /* Get core clock for data register access via DMA */ spdif_priv->coreclk = devm_clk_get(&pdev->dev, "core"); if (IS_ERR(spdif_priv->coreclk)) { -- 1.8.4

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[alsa-devel] [PATCH 1/5] ASoC: Add support for machine specific trigger callback
by Stefan Roese 29 Apr '14

29 Apr '14

From: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Machine specific trigger callback allows to do final stream start/stop related operations in a machine driver after setting up the codec, DMA and DAI. One example could be clock management for linked streams case where machine driver can start/stop synchronously the linked streams. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Stefan Roese <sr(a)denx.de> --- sound/soc/soc-pcm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2cedf09..a3a7021 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -819,6 +819,13 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; } + + if (rtd->dai_link->ops && rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); + if (ret < 0) + return ret; + } + return 0; } -- 1.9.1

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[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 29 Apr '14

29 Apr '14

Should I be posting somewhere else. If so can someone let me know where?

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[alsa-devel] [PATCH] ASoC: core: Fix component_list corruption when unloading modules
by Jarkko Nikula 29 Apr '14

29 Apr '14

This fixes module unload regressions introduced by commits 98e639fb8a3e ("ASoC: Track which components have been registered with snd_soc_register_component()") and b37f1d123c69 ("ASoC: Let snd_soc_platform subclass snd_soc_component"). First commit causes component_list to be corrupted when removing codec and second when removing platform. Reason for both is that components associated with platform or codec are never removed from the list because for them registered_as_component field in struct snd_soc_component is always false. Now list becomes corrupted when snd_soc_unregister_platform() or snd_soc_unregister_codec() frees the platform or codec structure and where the associated struct snd_soc_component is embedded. Fix these by moving component unregistration and cleanup to a new local function __snd_soc_unregister_component() that takes component as its argument. Since component is known for platforms and codecs the __snd_soc_unregister_component() can be called directly and snd_soc_unregister_component() takes care to find and unregister only components that were registered using snd_soc_register_component(). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-core.c | 25 +++++++++++++++---------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f18112a32541..62c3980fad5b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4061,6 +4061,18 @@ int snd_soc_register_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_register_component); +static void __snd_soc_unregister_component(struct snd_soc_component *cmpnt) +{ + snd_soc_unregister_dais(cmpnt); + + mutex_lock(&client_mutex); + list_del(&cmpnt->list); + mutex_unlock(&client_mutex); + + dev_dbg(cmpnt->dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); + kfree(cmpnt->name); +} + /** * snd_soc_unregister_component - Unregister a component from the ASoC core * @@ -4076,14 +4088,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(cmpnt); - - mutex_lock(&client_mutex); - list_del(&cmpnt->list); - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); - kfree(cmpnt->name); + __snd_soc_unregister_component(cmpnt); } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); @@ -4183,7 +4188,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); */ void snd_soc_remove_platform(struct snd_soc_platform *platform) { - snd_soc_unregister_component(platform->dev); + __snd_soc_unregister_component(&platform->component); mutex_lock(&client_mutex); list_del(&platform->list); @@ -4388,7 +4393,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - snd_soc_unregister_component(dev); + __snd_soc_unregister_component(&codec->component); mutex_lock(&client_mutex); list_del(&codec->list); -- 1.9.2

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