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

[alsa-devel] [PATCH] fm801: convert struct description to kernel-doc
by Andy Shevchenko 19 May '14

19 May '14

Just move field descriptions to the struct description in the kernel-doc format. There is no functional change. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/fm801.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index e8910d0..529f5f4 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -158,21 +158,27 @@ MODULE_PARM_DESC(radio_nr, "Radio device numbers"); #define FM801_GPIO_GS3 (1<<15) #define FM801_GPIO_GS(x) (1<<(12+(x))) -/* - +/** + * struct fm801 - describes FM801 chip + * @port: I/O port number + * @multichannel: multichannel support + * @secondary: secondary codec + * @secondary_addr: address of the secondary codec + * @tea575x_tuner: tuner access method & flags + * @ply_ctrl: playback control + * @cap_ctrl: capture control */ - struct fm801 { int irq; - unsigned long port; /* I/O port number */ - unsigned int multichannel: 1, /* multichannel support */ - secondary: 1; /* secondary codec */ - unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner access method & flags */ + unsigned long port; + unsigned int multichannel: 1, + secondary: 1; + unsigned char secondary_addr; + unsigned int tea575x_tuner; - unsigned short ply_ctrl; /* playback control */ - unsigned short cap_ctrl; /* capture control */ + unsigned short ply_ctrl; + unsigned short cap_ctrl; unsigned long ply_buffer; unsigned int ply_buf; -- 2.0.0.rc2

2 1

[alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add sound card driver for Snow board
by Tushar Behera 19 May '14

19 May '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 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 | 129 ++++++++++++++++++++++ 4 files changed, 158 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..6f3fd02 --- /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 I2S0 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..e1a4b45 --- /dev/null +++ b/sound/soc/samsung/snow.c @@ -0,0 +1,129 @@ +/* + * 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[] = { + { /* Playback DAI i/f */ + .name = "Playback", + .stream_name = "Playback", + .codec_dai_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, { /* Capture DAI i/f */ + .name = "Capture", + .stream_name = "Capture", + .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

3 13

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: Add check during device suspend
by Tushar Behera 19 May '14

19 May '14

Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause() unconditinally during device suspend. In case where DMA controller doesn't support PAUSE/RESUME functionality, this call is not able to stop the DMA controller. In this scenario, audio playback doesn't resume after device resume. Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes the issue. It has been tested with audio playback on Samsung platform having PL330 DMA controller which doesn't support PAUSE/RESUME. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- sound/core/pcm_dmaengine.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 94d0873..76cbb9e 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -182,6 +182,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; int ret; switch (cmd) { @@ -196,6 +197,11 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) dmaengine_resume(prtd->dma_chan); break; case SNDRV_PCM_TRIGGER_SUSPEND: + if (runtime->info & SNDRV_PCM_INFO_PAUSE) + dmaengine_pause(prtd->dma_chan); + else + dmaengine_terminate_all(prtd->dma_chan); + break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dmaengine_pause(prtd->dma_chan); break; -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] dma: pl330: Add support for DMA_PAUSE command
by Lars-Peter Clausen 19 May '14

19 May '14

On 05/15/2014 02:01 PM, Tushar Behera wrote: > On 14 May 2014 17:54, Lars-Peter Clausen <lars(a)metafoo.de> wrote: >> On 05/14/2014 02:07 PM, Tushar Behera wrote: >>> >>> On 14 May 2014 17:29, Jassi Brar <jassisinghbrar(a)gmail.com> wrote: >>>> >>>> On Wed, May 14, 2014 at 8:53 AM, Tushar Behera <tushar.behera(a)linaro.org> >>>> wrote: >>>>> >>>>> While playing back audio, pmc_dmaengine requests the DMA channel to >>>>> stop DMA transmission through DMA_PAUSE command. >>>>> >>>>> Currently PL330 driver doesn't support DMA pause command, leaving >>>>> the DMA state inconsistent when the system resumes. Instead, it would >>>>> be better to terminate the DMA transfer during suspend and restart >>>>> again during resume. >>>>> >>>>> Tested with audio playback across a suspend-resume cycle. >>>>> >>>> What is pmc_dmaengine? How does DMA_PAUSE help, when there is no >>>> DMA_RESUME? >>>> >>> >>> Sorry, it is a typo. >>> >>> sound/core/pcm_dmaengine.c:snd_dmaengine_pcm_trigger() --> >>> dmaengine_pause() is called during system suspend. >> >> >> It is only called if the DMA driver has support for pausing and resuming DMA >> transfers. Or at least that is the intention. >> >> - Lars > > During suspend, snd_dmaengine_pcm_trigger():SNDRV_PCM_TRIGGER_SUSPEND > is called which unconditionally calls dmaengine_pause(). Should we > update snd_dmaengine_pcm_trigger() to check for DMA pause/resume > support and call dmaengine_pause() or dmaengine_terminate_all() > accordingly? As far as I understand it we do not have to do anything for TRIGGER_SUSPEND if we do not set the SNDRV_PCM_INFO_RESUME flag. It looks like TRIGGER_SUSPEND is called unconditionally during suspend. But since the error code is ignored it should be fine if we just call dmaengine_pause() and that return -ENOSYS or similar. Are you seeing an actual issue that you are trying to fix with your patch? - Lars

3 6

[alsa-devel] [PATCH 0/13] make return of 0 explicit
by Julia Lawall 19 May '14

19 May '14

Sometimes a local variable is used as a return value in a case where the return value is always 0. The result is more apparent if this variable is just replaced by 0. This is done by the following semantic patch, although some cleanups may be needed. (http://coccinelle.lip6.fr/) // <smpl> @r exists@ local idexpression ret; identifier reti; expression e; position p; @@ ret@reti = 0; ... when != $ret = e\|ret &= e\|ret |= e\|ret += e\|ret -= e\|ret *= e\|ret ^= e$ when != $++ret\|--ret\|ret++\|ret--$ when != &ret return ret;@p @bad1 exists@ expression e != 0; local idexpression r.ret; position r.p; @@ $ret = e\|ret &= e\|ret |= e\|ret += e\|ret -= e\|ret *= e\|ret ^= e\|++ret\|--ret\|ret++\|ret--\|&ret$ ... return ret;@p @bad2 exists@ identifier r.reti; position r.p; identifier f; type T; @@ f(...,T reti,...) { ... return reti;@p } @change depends on !bad1 && !bad2@ position r.p; local idexpression r.ret; identifier f; @@ f(...) { <+... return -ret +0 ;@p ...+> } @rewrite@ local idexpression r.ret; expression e; position p; identifier change.f; @@ f(...) { <+... $ret@p = e\|&ret@p$ ...+> } @depends on change@ local idexpression r.ret; position p != rewrite.p; identifier change.f; @@ f(...) { <+... ( break; | ret = 0; ... when exists ret@p ... when any | - ret = 0; ) ...+> } @depends on change@ identifier r.reti; type T; constant C; identifier change.f; @@ f(...) { ... when any -T reti = C; ... when != reti when strict } @depends on change@ identifier r.reti; type T; identifier change.f; @@ f(...) { ... when any -T reti; ... when != reti when strict } // </smpl> The first four rules detect cases where only 0 reaches a return. The remaining rules clean up any variable initializations or declarations that are made unnecessary by this change.

2 2

[alsa-devel] [PATCH] ASoC: Intel: Add Baytrail byt-max98090 machine driver
by Jarkko Nikula 19 May '14

19 May '14

Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec. Jack detect code from Kevin Strasser <kevin.strasser(a)intel.com>, GPIO resolving from Mika Westerberg <mika.westerberg(a)linux.intel.com> and fixes and cleanups from Liam Girdwood <liam.r.girdwood(a)linux.intel.com>. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/Kconfig | 10 ++ sound/soc/intel/Makefile | 2 + sound/soc/intel/byt-max98090.c | 252 +++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-acpi.c | 1 + 4 files changed, 265 insertions(+) create mode 100644 sound/soc/intel/byt-max98090.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3c81b3891209..d6c3d840ed82 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -49,3 +49,13 @@ config SND_SOC_INTEL_BYT_RT5640_MACH help This adds audio driver for Intel Baytrail platform based boards with the RT5640 audio codec. + +config SND_SOC_INTEL_BYT_MAX98090_MACH + tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_MAX98090 + select VLV2_PLAT_CLK + help + This adds audio driver for Intel Baytrail platform based boards + with the MAX98090 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 0db4e2f336dc..4bfca79a42ba 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -23,6 +23,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o # Machine support snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o +snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c new file mode 100644 index 000000000000..4246b2c81201 --- /dev/null +++ b/sound/soc/intel/byt-max98090.c @@ -0,0 +1,252 @@ +/* + * Intel Baytrail SST MAX98090 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope 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/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/max98090.h" + +struct byt_max98090_private { + struct snd_soc_jack jack; +}; + +static const struct snd_soc_dapm_widget byt_max98090_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route byt_max98090_audio_map[] = { + {"IN34", NULL, "Headset Mic"}, + {"IN34", NULL, "MICBIAS"}, + {"MICBIAS", NULL, "Headset Mic"}, + {"DMICL", NULL, "Int Mic"}, + {"Headphone", NULL, "HPL"}, + {"Headphone", NULL, "HPR"}, + {"Ext Spk", NULL, "SPKL"}, + {"Ext Spk", NULL, "SPKR"}, +}; + +static const struct snd_kcontrol_new byt_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Ext Spk", + .mask = SND_JACK_LINEOUT, + }, + { + .pin = "Int Mic", + .mask = SND_JACK_LINEIN, + }, +}; + +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, + .debounce_time = 200, + }, + { + .name = "mic-gpio", + .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN, + .debounce_time = 200, + }, +}; + +static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = runtime->card; + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); + struct snd_soc_jack *jack = &drv->jack; + struct gpio_desc *mic_desc; + struct gpio_desc *hp_desc; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_dai_set_sysclk(runtime->codec_dai, + M98090_REG_SYSTEM_CLOCK, + 25000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "Can't set codec clock %d\n", ret); + return ret; + } + + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphone"); + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + + snd_soc_dapm_sync(dapm); + + /* + * ASoC still uses legacy GPIOs so we look both GPIOs using + * descriptors here, convert them to numbers and release the + * acquired descriptors. Once ASoC switches over to GPIO descriptor + * APIs we can pass them directly. + */ + hp_desc = gpiod_get_index(card->dev->parent, NULL, 0); + if (IS_ERR(hp_desc)) + return 0; + mic_desc = gpiod_get_index(card->dev->parent, NULL, 1); + if (IS_ERR(mic_desc)) { + gpiod_put(hp_desc); + return 0; + } + + hs_jack_gpios[0].gpio = desc_to_gpio(hp_desc); + hs_jack_gpios[1].gpio = desc_to_gpio(mic_desc); + + gpiod_put(mic_desc); + gpiod_put(hp_desc); + + /* Enable jack detection */ + ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, M98090_IJDET_MASK, + 1 << M98090_IJDET_SHIFT); + + snd_soc_jack_report(jack, SND_JACK_LINEOUT | SND_JACK_LINEIN, + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_LINEIN); + + ret = snd_soc_jack_add_gpios(jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; +} + +static struct snd_soc_dai_link byt_max98090_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "baytrail-pcm-audio", + .codec_dai_name = "HiFi", + .codec_name = "i2c-193C9890:00", + .platform_name = "baytrail-pcm-audio", + .init = byt_max98090_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card byt_max98090_card = { + .name = "byt-max98090", + .dai_link = byt_max98090_dais, + .num_links = ARRAY_SIZE(byt_max98090_dais), + .dapm_widgets = byt_max98090_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets), + .dapm_routes = byt_max98090_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map), + .controls = byt_max98090_controls, + .num_controls = ARRAY_SIZE(byt_max98090_controls), +}; + +#ifdef CONFIG_PM_SLEEP +static const struct dev_pm_ops byt_max98090_pm_ops = { + .suspend = snd_soc_suspend, + .resume = snd_soc_resume, +}; + +#define BYT_MAX98090_PM_OPS (&byt_max98090_pm_ops) +#else +#define BYT_MAX98090_PM_OPS NULL +#endif + +static int byt_max98090_probe(struct platform_device *pdev) +{ + int ret_val = 0; + struct byt_max98090_private *drv; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC); + if (!drv) { + dev_err(&pdev->dev, "allocation failed\n"); + return -ENOMEM; + } + + /* register the soc card */ + byt_max98090_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&byt_max98090_card, drv); + ret_val = snd_soc_register_card(&byt_max98090_card); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &byt_max98090_card); + + return ret_val; +} + +static int byt_max98090_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + struct byt_max98090_private *drv = snd_soc_card_get_drvdata(soc_card); + + snd_soc_jack_free_gpios(&drv->jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver byt_max98090_driver = { + .probe = byt_max98090_probe, + .remove = byt_max98090_remove, + .driver = { + .name = "byt-max98090", + .owner = THIS_MODULE, + .pm = BYT_MAX98090_PM_OPS, + }, +}; +module_platform_driver(byt_max98090_driver) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-max98090"); diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 18aee77f8d4a..42edc6f4fc4a 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -247,6 +247,7 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { static struct sst_acpi_mach baytrail_machines[] = { { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, {} }; -- 2.0.0.rc0

2 2

[alsa-devel] This patch is for hearing sound separate on channels
by asd 18 May '14

18 May '14

--- a/hda_codec.c +++ b/hda_codec.c @@ -5375,6 +5375,7 @@ int chs = substream->runtime->channels; struct hda_spdif_out *spdif; int i; + //mout->no_share_stream = 1; mout->no_share_stream = 1; mutex_lock(&codec->spdif_mutex);

1 0

[alsa-devel] [BUG]ABit aa8xe sound card not selected
by Stefano Statuti 18 May '14

18 May '14

hello, I have opened this ticket on ubuntu-alsa launchpad _https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1319291 Raymond (superquad-vortrex2) ask me to contact you. ----quote Try specify model=6stack-dig https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 147b:1045 send email to alsa-devel mailing list to add pci quirk since Bios does not setup pin default Default 0x00000000: [Jack] Line Out at Ext N/A Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x00000000 0x1f 0x00000000 ----- whit Ubuntu 12.04 I don't have this problem bye

2 1

[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

2 1

[alsa-devel] [PATCH] ALSA: ASoC: Fix WM9713 mic 2 preamp volume
by Matt Reimer 17 May '14

17 May '14

Fix WM9713's mic 2 preamp volume. The previous definition specified the Mic A pre-amp input select bits (aka "Mic A Source") instead. Signed-off-by: Matt Reimer <mreimer(a)sdgsystems.com> --- sound/soc/codecs/wm9713.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index acea892..5f69b52 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -134,7 +134,7 @@ SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv), -SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv), +SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 8, 3, 0, mic_tlv), SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), -- 1.7.5.4

2 1