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

[alsa-devel] Wolfson 5102 on Beaglebone Black using ALSA
by Clark Dunson 15 Jun '16

15 Jun '16

Hey everybody! We have been working on integrating the Wolfson 5102 codec with the Beaglebone Black. We’ve made pretty decent progress, got the arizona code in kernel 4.4 registering the wm5102 as ALSA sound card (both player and recorder), have McAsp0 as 24.576 MHz clock source, and have 12 IRQs showing up in /proc/interrupts. We have been working with sound{compatible = "simple-audio-card”}, but now have hit our first real question. sysclk is hard coded to ‘0’ under simple-audio-card, and we had to hack the kernel and change that to ‘1’ to achieve ALSA sound card registration. We’d all like for this work to proceed without having to hack the kernel, and wondered if one of you might recommend another approach or selection for device tree sound{compatible} that might work out better? Thanks! Clark

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[alsa-devel] [PATCH] ASoC: dapm: Fix to return correct path list in is_connected_ep.
by jeeja.kp＠intel.com 15 Jun '16

15 Jun '16

From: Jeeja KP <jeeja.kp(a)intel.com> In is_connected_ep, when custom_stop_condition is true, need to return the correct paths instead of con which is 0. Fixes: 6742064aef7f('ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets') Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-dapm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index db781f6..3c3f027 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1092,8 +1092,10 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); - if (custom_stop_condition && custom_stop_condition(widget, dir)) - return con; + if (custom_stop_condition && custom_stop_condition(widget, dir)) { + widget->endpoints[dir] = 1; + return widget->endpoints[dir]; + } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); -- 2.5.0

2 1

[alsa-devel] [PATCH 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Clemens Gruber 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. [I rebased Eric's patches from Feb 2015, fixed a few warnings, URLs, etc. and squashed two patches into one. - Clemens] Clemens Gruber (1): ASoC: sgtl5000: Remove misleading comment Eric Nelson (5): ASoC: sgtl5000: Fix regulator support ASoC: sgtl5000: Write all default registers ASoC: sgtl5000: Initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: Disable internal PLL early ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 421 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 321 deletions(-) -- 2.8.3

4 17

[alsa-devel] [PATCH RFC 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Eric Nelson 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. Patch 1 moves the regulators to the I2C device and initializes them before I2C activity. Patch 2 addresses an issue found during development of the patch and is somewhat unrelated to regulators. It is included for discussion Patch 3 removes the root cause of patch 2 Patch 4 adds return value checks when setting the LINREG and ANA_POWER registers Patch 5 addresses the cause of follow-on failures and the source of the error in patch 3. Patch 6 removes the naive use of the supplies to lower power consumption and replaces it with something more modest. The patch set is being sent as an RFC for discussion, and should probably be squashed into two or three patches covering the regulator updates (patches 1, 3-5 and perhaps 6), initialization error handling (patch 2). Eric Nelson (6): ASoC: sgtl5000: fix regulator support ASoC: sgtl5000: write all default registers ASoC: sgtl5000: initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: check return values ASoC: sgtl5000: disable internal PLL early ASoC: sgtl5000: Don't disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 417 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 317 deletions(-) -- 1.9.1

3 28

[alsa-devel] [PATCH v2] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Change numerical value of ADAU17X1_CLK_SRC_PLL_AUTO to 0, since that is what the simple-card driver will call snd_soc_dai_set_sysclk() with when the CODEC node has a clock property. So this is really the only way to make this work with the simple-card driver. --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..bf04b7e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -22,13 +22,18 @@ enum adau17x1_pll_src { }; enum adau17x1_clk_src { + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/usb/mixer_maps.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1f8fb0d9..9038b2e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin(a)bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; + /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; + /* just guess */ static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..e9fbc3b 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -24,11 +24,16 @@ enum adau17x1_pll_src { enum adau17x1_clk_src { ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

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[alsa-devel] [PATCH] ALSA: seq_oss: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Also, removed some unnecessary comments. Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/core/seq/oss/seq_oss_synth.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index b16dbef..cd0e0eb 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -70,11 +70,11 @@ struct seq_oss_synth { static int max_synth_devs; static struct seq_oss_synth *synth_devs[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS]; static struct seq_oss_synth midi_synth_dev = { - -1, /* seq_device */ - SYNTH_TYPE_MIDI, /* synth_type */ - 0, /* synth_subtype */ - 16, /* nr_voices */ - "MIDI", /* name */ + .seq_device = -1, + .synth_type = SYNTH_TYPE_MIDI, + .synth_subtype = 0, + .nr_voices = 16, + .name = "MIDI", }; static DEFINE_SPINLOCK(register_lock); -- 1.9.1

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Re: [alsa-devel] Support Dell headset mode for ALC891
by Takashi Iwai 15 Jun '16

15 Jun '16

On Wed, 08 Jun 2016 09:17:12 +0200, Kailang wrote: > > Hi Takashi, > > This patch is supported Dell headset mode for ALC891. > It is only support I-phone type headset. > I think this function is only support for DELL. > So, headset mode and platform support is merged. > > This patch is test passed by Ubuntu team. > > Many Thanks. (snip) > Date: Tue, 7 Jun 2016 11:31:34 +0800 > Subject: [PATCH] ALSA: hda/realtek - ALC891 headset mode for Dell > > New headset mode of ALC891 for Dell. > > Signed-off-by: Kailang Yang <kailang(a)realtek.com> Could you put more comments in the changelog, as what you gave in the mail? > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7960316..9dc641e 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3718,6 +3718,9 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > case 0x10ec0295: > alc_process_coef_fw(codec, coef0225); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); > + break; > } > codec_dbg(codec, "Headset jack set to unplugged mode.\n"); > } > @@ -3805,6 +3808,8 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > alc_process_coef_fw(codec, coef0293); > snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); > case 0x10ec0662: No break between 0867 and 0662? thanks, Takashi

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[alsa-devel] [alsa-lib][PATCH 00/10 v2] ctl: add APIs for control element set
by Takashi Sakamoto 15 Jun '16

15 Jun '16

Hi, This patchset is revised version of below one: [alsa-devel] [alsa-lib][PATCH 0/6] control: add APIs for control element set http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104795.h… In this time, I add a program to test the feature. Changes: - Use a model which consists of element set/element/member/value. - Add an description about the design of ALSA ctl feature. - Obsolete old APIs due to the lack of const semantics. - Add an description about threshold level feature and topology framework. - Improve comments to threshold level APIs. - Add a test program. Before starting any discussions, please read description of the model, to stand on the same basis of understanding. I note that execution of the test program causes an abort to PulseAudio. I can see this line below. pulseaudio: hcontrol.c:745: snd_hctl_handle_event: Assertion `res >= 0 && dir == 0' failed. I guess that quick addition/removal of control element set causes this assertion, but I have little time to investigate the cause. I wish to get helps of ALSA developers in PulseAudio side. If you need an practical example of the feature, please test one of Python 3 samples in my 'alsa-gi'. https://github.com/takaswie/alsa-gi/ This works without alsa-lib, by using kernel/userspace interface directly. Therefore, this patchet doesn't affect the implementation. Takashi Sakamoto (10): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs ctl: deprecate APIs to add an element set with a single element pcm: use new APIs to add a control element set for softvol plugin ctl: add explaination about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 36 +- src/control/control.c | 829 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1216 insertions(+), 222 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

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