May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH]: ASoC: intel: make function stub static
by Randy Dunlap 24 May '16

24 May '16

From: Randy Dunlap <rdunlap(a)infradead.org> This function stub should have been 'static' in the original patch so that multiple uses of the header file (in different drivers) will not cause multiple function definitions. sound/soc/intel/boards/built-in.o: In function `sst_acpi_find_name_from_hid': (.text+0x560): multiple definition of `sst_acpi_find_name_from_hid' sound/soc/intel/atom/built-in.o:(.text+0x10610): first defined here ../scripts/Makefile.build:369: recipe for target 'sound/soc/intel/built-in.o' failed Fixes: f17131a93f43: add function stub when ACPI is not enabled Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Borislav Petkov <bp(a)alien8.de> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/common/sst-acpi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20160520.orig/sound/soc/intel/common/sst-acpi.h +++ linux-next-20160520/sound/soc/intel/common/sst-acpi.h @@ -20,7 +20,7 @@ #if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); #else -inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +static inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) { return NULL; }

2 1

Re: [alsa-devel] Add support for ALC295
by Takashi Iwai 24 May '16

24 May '16

On Tue, 24 May 2016 10:53:37 +0200, Kailang wrote: > > Hi Takashi, > > We have a new codec ALC295 that is brother and sister with ALC225. > Attach is the support patch. Applied now. Thanks. Takashi

1 0

[alsa-devel] [PATCH] MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski
by Krzysztof Kozlowski 24 May '16

24 May '16

Extend maintainer entry for Samsung SoC sound drivers with Krzysztof Kozlowski and Sylwester Nawrocki. The file pattern is duplicated in main Exynos ARM section so remove it from there. Cc: Kukjin Kim <kgene(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-samsung-soc(a)vger.kernel.org Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 243c0811a4d8..f0f4ca58d264 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1603,7 +1603,6 @@ F: drivers/*/*/*s3c2410* F: drivers/memory/samsung/* F: drivers/soc/samsung/* F: drivers/spi/spi-s3c* -F: sound/soc/samsung/* F: Documentation/arm/Samsung/ F: Documentation/devicetree/bindings/arm/samsung/ F: Documentation/devicetree/bindings/sram/samsung-sram.txt @@ -9857,7 +9856,9 @@ S: Maintained F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS +M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> M: Sangbeom Kim <sbkim73(a)samsung.com> +M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung/ -- 1.9.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 24 May '16

24 May '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.6 kernel. Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 6 files changed, 112 insertions(+), 69 deletions(-) -- 1.9.3

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: wm8985: add device tree binding for WM8985
by Petr Kulhavy 24 May '16

24 May '16

Add device tree binding for the WM8985 codec driver. Signed-off-by: Petr Kulhavy <petr(a)barix.com> Acked-by: Rob Herring <robh(a)kernel.org> --- v1: initial v2: no change Documentation/devicetree/bindings/sound/wm8985.txt | 36 ++++++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8985.txt diff --git a/Documentation/devicetree/bindings/sound/wm8985.txt b/Documentation/devicetree/bindings/sound/wm8985.txt new file mode 100644 index 000000000000..788d64766257 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8985.txt @@ -0,0 +1,36 @@ +Wolfson Microelectronics WM8985 and WM8758 audio codecs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8985" or "wlf,wm8758" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Pins on the device for linking into audio routes: + + * LIN : left microphone pre-amp negative input + * LIP : left microphone pre-amp positive input + * RIN : right microphone pre-amp negative input + * RIP : right microphone pre-amp positive input + * L2 : left line input / secondary pre-amp positive input + * R2 : right line input / secondary pre-amp positive input + * AUXL : left auxiliary input (WM8985 only) + * AUXR : right auxiliary input (WM8985 only) + + * HPL : left headphone / line output (the LOUT1 pin) + * HPR : right headphone / line output (the ROUT1 pin) + * SPKL : left headphone / line output (the LOUT2 pin) + * SPKR : right headphone / line output (the ROUT2 pin) + * Mic Bias : microphone bias + + +Example: + +wm8758: audio-codec@1a { + compatible = "wlf,wm8758"; + reg = <0x1a>; +}; -- 1.9.1

2 7

[alsa-devel] [PATCH v3 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 24 May '16

24 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver. Changes in v3: * merge the wm8985_xxx_snd_controls into wm8985_common_snd_controls and wm8985_specific_snd_controls in order to reduce the number of structures use the static initialization in snd_soc_codec_driver for the common part

2 4

[alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Chuang Eric 24 May '16

24 May '16

Hi, Let me illustrate the system first. FE1 -----+----> BE | FE2 -----+ When there is multi-FE stream to a single BE, the BE will receive SND_SOC_DAPM_STREAM_STOP if one of the FE shutdown. However, the BE is stilled used by another, this will let BE cpu_dai and codec_dai think they are inactive. The later power check will go wrong. I have check the latest linux version, and didn't find any related change. The following is my change to address this, and will re-send a formal patch if it's ok. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e8088..9c2d159 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -163,6 +163,13 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + /* don't sent stop event if this BE is used by other FE */ + if (event == SND_SOC_DAPM_STREAM_STOP && + be->dpcm[dir].users >= 1) { + pr_warn("kc, %s(), be->dai_link->name %s skip stop event\n", __func__, be->dai_link->name); + continue; + } + snd_soc_dapm_stream_event(be, dir, event); } Sincerely, Kai Chieh

1 0

[alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by srv_kaichieh.chaung 24 May '16

24 May '16

Hi, Let me illustrate the system first. FE1 -----+----> BE | FE2 -----+ When there is multi-FE stream to a single BE, the BE will receive SND_SOC_DAPM_STREAM_STOP if one of the FE shutdown. However, the BE is stilled used by another, this will let BE cpu_dai and codec_dai think they are inactive. The later power check will go wrong. I have check the latest linux version, and didn't find any related change. The following is my change to address this, and will re-send a formal patch if it's ok. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e8088..9c2d159 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -163,6 +163,13 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + /* don't sent stop event if this BE is used by other FE */ + if (event == SND_SOC_DAPM_STREAM_STOP && + be->dpcm[dir].users >= 1) { + pr_warn("kc, %s(), be->dai_link->name %s skip stop event\n", __func__, be->dai_link->name); + continue; + } + snd_soc_dapm_stream_event(be, dir, event); } Sincerely, Kai Chieh

1 0

[alsa-devel] [PATCH v2] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 24 May '16

24 May '16

There are multi codec devices on the RK3399 platform, we can use this patch support and control these codecs. --- Changes in v2: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> .../sound/rockchip-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../rockchip/rockchip_max98357a_rt5514_da7219.c | 333 ++++++++++++++++++++ 4 files changed, 361 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..5928384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rockchip-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rockchip-sound"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..d02aa94 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_ROCKCHIP_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip boards using the MAX98357A/RT5514/DA7219" + depends on SND_SOC_ROCKCHIP && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support for SoC audio on Rockchip + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..ca97489 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rockchip-max98357a-rt5514-da7219-objs := rockchip_max98357a_rt5514_da7219.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98357A_RT5514_DA7219) += snd-soc-rockchip-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..818c47e --- /dev/null +++ b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c @@ -0,0 +1,333 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rockchip-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rockchip-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + for (i = 0; i < card->num_links; i++) { + cpu_node = of_parse_phandle(pdev->dev.of_node, + "rockchip,cpu", i); + if (!cpu_node) { + dev_err(&pdev->dev, "Property[%d] 'rockchip,cpu' missing or invalid\n", i); + return -EINVAL; + } + + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rockchip-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

2 2

[alsa-devel] [PATCH 43/54] MAINTAINERS: Add file patterns for sound device tree bindings
by Geert Uytterhoeven 23 May '16

23 May '16

Submitters of device tree binding documentation may forget to CC the subsystem maintainer if this is missing. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- Please apply this patch directly if you want to be involved in device tree binding documentation for your subsystem. --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 139824579fa97c2c..9d81eff6577ff886 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10728,6 +10728,7 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported +F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* -- 1.9.1

2 2

[alsa-devel] [PATCH 0056/1529] Fix typo
by Andrea Gelmini 23 May '16

23 May '16

Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- Documentation/sound/alsa/soc/machine.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 74056db..6bf2d20 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -3,7 +3,7 @@ ASoC Machine Driver The ASoC machine (or board) driver is the code that glues together all the component drivers (e.g. codecs, platforms and DAIs). It also describes the -relationships between each componnent which include audio paths, GPIOs, +relationships between each component which include audio paths, GPIOs, interrupts, clocking, jacks and voltage regulators. The machine driver can contain codec and platform specific code. It registers -- 2.8.2.534.g1f66975

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: fix build failure
by Sudip Mukherjee 23 May '16

23 May '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/kirkwood/snd-soc-kirkwood.ko] undefined! To satisfy the dependency CONFIG_SND_KIRKWOOD_SOC should depend on HAS_DMA. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- build log at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/132180911 sound/soc/kirkwood/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 132bb83..bc3c7b5 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,7 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST + depends on HAS_DMA help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 23 May '16

23 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver.

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 23 May '16

23 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver. Changes in v2: * use chip type enum instead of chip structure * do not copy the complete controls, widget, routes but only the differences and add widgets dynamically using add_widgets()

2 7

[alsa-devel] [PATCH] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 23 May '16

23 May '16

There are multi codec devices on the RK3399 platform, we can use this patch support and control these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- .../sound/rockchip-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../rockchip/rockchip_max98357a_rt5514_da7219.c | 339 ++++++++++++++++++++ 4 files changed, 367 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..5928384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rockchip-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rockchip-sound"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..d02aa94 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_ROCKCHIP_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip boards using the MAX98357A/RT5514/DA7219" + depends on SND_SOC_ROCKCHIP && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support for SoC audio on Rockchip + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..ca97489 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rockchip-max98357a-rt5514-da7219-objs := rockchip_max98357a_rt5514_da7219.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98357A_RT5514_DA7219) += snd-soc-rockchip-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..50c0d35 --- /dev/null +++ b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c @@ -0,0 +1,339 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rockchip-sound" + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = 12288000; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rockchip-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + for (i = 0; i < card->num_links; i++) { + cpu_node = of_parse_phandle(pdev->dev.of_node, + "rockchip,cpu", i); + if (!cpu_node) { + dev_err(&pdev->dev, "Property[%d] 'rockchip,cpu' missing or invalid\n", i); + return -EINVAL; + } + + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rockchip-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

1 0

[alsa-devel] journalctl suggests I report Bug to ALSA developers
by P.W. Pulto 21 May '16

21 May '16

Dear ALSA developers, Having a problem with my Sennheiser DW Pro 2 Dect Headset, I looked in the journalctl. It suggested I report a bug to the ALSA developers. I have no idea if this would be a "kernel driver bug" or an "other ALSA bug". Hopefully I'm at the right address. The problem occurs when I take the headset off the base station and a connection between base station and headset is established. My ALSA-info: ------------------- http://www.alsa-project.org/db/?f=2eaeea040fef7cc2476c239e1fec90dd2bf4eea8 The journalctl output: ----------------------------- journalctl --no-pager --since "25 min ago" May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Hopefully this is sufficient information. Obviously you welcome to ask for missing info! Kind regards, P.W. Pulto -- Like us on Facebook <https://www.facebook.com/studiopulto> dhr. P.W. Pulto Studio Pulto vzw Jef Vandermeulenstraat 37 9050 Gentbrugge Bélgica tel: 0032 9 220 73 40 e-mail: pwpulto(a)studiopulto.org http://www.studiopulto.org IBAN.: BE03 4487 6576 5184 BIC: KREDBEBB ondern nr.: BE 0470 427 828

1 0

[alsa-devel] [PATCH 1/1] ASoC: pxa: Fix module autoload for platform drivers
by Andrea Adami 21 May '16

21 May '16

These platform drivers are lacking MODULE_ALIAS so module autoloading doesn't work. Tested on corgi and poodle with kernel 4.4. Signed-off-by: Andrea Adami <andrea.adami(a)gmail.com> --- sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-pcm.c | 1 + sound/soc/pxa/mmp-sspa.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + 8 files changed, 8 insertions(+) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index ec522e9..b6cb995 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -133,3 +133,4 @@ module_platform_driver(mmp_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("ALSA SoC Brownstone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 5c8f9db..d1661fa 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver); MODULE_AUTHOR("Robert Jarzmik (rjarzmik(a)free.fr)"); MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 51e790d..96df9b2 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("MMP Soc Audio DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index eca60c2..ca8b23f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("MMP SSPA SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4e74d95..bcc81e9 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver); MODULE_AUTHOR("Marek Vasut <marek.vasut(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index da03fad..3cad990 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f3de615..9615e6d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 9f39039..410d48b 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v7 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Scott Branden 20 May '16

20 May '16

Looks good Simran. Mark, anything else for us to do before this driver can be accepted upstream? On 16-05-17 12:46 PM, Simran Rai wrote: > From: Simran <ssimran(a)lbrmn-ssimran-lnxub64.ric.broadcom.com> > > Hi, > > This patchset contains audio support for Broadcom's Cygnus SoC. > It contains DT bindings and core audio driver. The audio driver supports > both capture and playback of Audio PCM samples over I2S/TDM interface and > provides playback support over SPDIF interface. > > This patchset is derived from a previously submitted patchset: > http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html > > This patchset has been tested on Cygnus wireless audio bcm958305K board. > It is based on v4.6-rc1 and is available from github: > > repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v7 > > Changes from v6: > - DT bindings acknowledged by Rob Herring > Changes from v5: > - Fix code style, e.g. change "if" statements to "switch" statements > - Fix SPDIF output enable register field > - Set BUFFER_PAIR_ENABLE for both mono, stereo and TDM modes > - Reflect PCM bit formats for SPDIF, SSP and TDM as supported by > hardware > Changes from v4: > - Fix power suspend function and add power resume function > - Move clock initialization code from audio driver to clock framework > Changes from v3: > - Fix the subject lines to match the style for the subsystem > Changes from v2: > - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. > - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using > standard kernel trace infrastructure. > - Fix interrupt handler. Acknowledge only those interrupts that are > handledby ISR. > - Modify configure_vco() and the pll_macro_entry() struct to make it > better readable. The functionality did not change. > - Remove casts on macros > - Remove surround sound channel grouping from the driver. > Changes from v1: > - Address code review comments. Fix print format of type size_t and > pointer. > > Simran Rai (3): > ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio > ASoC: cygnus: Add Cygnus audio DAI driver > ASoC: cygnus: Add Cygnus audio DMA driver > > .../bindings/sound/brcm,cygnus-audio.txt | 67 + > sound/soc/bcm/Kconfig | 9 + > sound/soc/bcm/Makefile | 5 + > sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ > sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++++ > sound/soc/bcm/cygnus-ssp.h | 139 ++ > 6 files changed, 2610 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt > create mode 100644 sound/soc/bcm/cygnus-pcm.c > create mode 100644 sound/soc/bcm/cygnus-ssp.c > create mode 100644 sound/soc/bcm/cygnus-ssp.h >

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headphone noise on Dell XPS 13 9360
by Kai-Heng Feng 20 May '16

20 May '16

The headphone has noise when playing sound or switching microphone sources. It uses the same codec on XPS 13 9350, but with different subsystem ID. Applying the fixup can solve the issue. Also, changing the model name to better differentiate models. v2: Reorder by device ID. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- 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 002f153..10ca900 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5466,8 +5466,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.8.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headphone noise on Dell XPS 13 9360
by Kai-Heng Feng 20 May '16

20 May '16

The headphone has noise when playing sound or switching microphone sources. It uses the same codec on XPS 13 9350, but with different subsystem ID. Applying the fixup can solve the issue. Also, changing the model name to better differentiate models. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- 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 002f153..e6af865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5466,7 +5466,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.8.1

2 1

[alsa-devel] [PATCH v4 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 19 May '16

19 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 4 of the patch: * minor correction in the binding example ("interrupt names") * re-added index note to the "mpu" property.

4 18

[alsa-devel] [PATCH 1/1] ASoC: core: Do not print an actual error when deferring probe
by Alexander Stein 19 May '16

19 May '16

In case there is no DAI (yet), do not print an error, this might happen a lot of times. Print a notice instead. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..352f7c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1001,7 +1001,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, cpu_dai_component.dai_name = dai_link->cpu_dai_name; rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { - dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); goto _err_defer; } @@ -1013,7 +1013,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); goto _err_defer; } @@ -1042,7 +1042,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->platform = platform; } if (!rtd->platform) { - dev_err(card->dev, "ASoC: platform %s not registered\n", + dev_notice(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } -- 2.7.3

4 12

[alsa-devel] [PATCH] ASoC: ak4642: Enable cache usage to fix crashes on resume
by Mark Brown 19 May '16

19 May '16

The ak4642 driver is using a regmap cache sync to restore the configuration of the chip on resume but (as Peter observed) does not actually define a register cache which means that the resume is never going to work and we trigger asserts in regmap. Fix this by enabling caching. Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/ak4642.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1ee8506c06c7..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { -- 2.8.1

3 4

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Add channel constraints for refcap
by Yong Zhi 19 May '16

19 May '16

Add constraint for ref DMIC to match with the topology firmware config. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 7cc7251..463f0e5 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -339,8 +339,22 @@ static struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int skylake_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); -- 1.9.1

2 2

[alsa-devel] [PATCH V2 1/2] dt: binding: ASOC: add dt-binding doc for pcm5102a
by kernel＠martin.sperl.org 18 May '16

18 May '16

From: Martin Sperl <kernel(a)martin.sperl.org> Add dt-binding documentation for the TI pcm5102a DAC. Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> --- Documentation/devicetree/bindings/sound/pcm5102a.txt | 13 +++++++++++++ 1 file changed, 13 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 0000000..87d72be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODEC + +This device is a i2s only device. + +Required properties: + +- compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; -- 2.1.4

2 2