August 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture
by Lothar Waßmann 05 Aug '13

05 Aug '13

When a sound capture/playback is terminated while a playback/capture is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other channel. Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active to prevent this. Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/codecs/sgtl5000.c | 16 +++++++++++++--- 1 files changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c8b9090..a4b2973 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: pcm1792a: Fix build with !OF
by Mark Brown 05 Aug '13

05 Aug '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm1792a.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3f83bf9..72cf835 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -169,13 +169,11 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .ops = &pcm1792a_dai_ops, }; -#ifdef CONFIG_OF static const struct of_device_id pcm1792a_of_match[] = { { .compatible = "ti,pcm1792a", }, { } }; MODULE_DEVICE_TABLE(of, pcm1792a_of_match); -#endif static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, @@ -231,7 +229,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = pcm1792a_of_match, + .of_match_table = of_match_ptr(pcm1792a_of_match), }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- 1.8.4.rc0

1 0

[alsa-devel] [RFC PATCH] ASoC: Add PCM1792A spi mode codec support
by Michael Trimarchi 05 Aug '13

05 Aug '13

Add PCM1792A spi mode codec support. This version implements only a subset of functionalities. Tested connect to a pandaboard ES device and based on recently pcm1681 codec. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../devicetree/bindings/sound/pcm1792a.txt | 18 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1792a.c | 245 ++++++++++++++++++++ sound/soc/codecs/pcm1792a.h | 26 +++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1792a.txt create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h diff --git a/Documentation/devicetree/bindings/sound/pcm1792a.txt b/Documentation/devicetree/bindings/sound/pcm1792a.txt new file mode 100644 index 0000000..b8e48e8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm1792a.txt @@ -0,0 +1,18 @@ +Texas Instruments pcm1792a DT bindings + +This driver supports the SPI bus. + +Required properties: + + - compatible: "ti,pcm1792a" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Examples: + + codec_spi: 1792a@0 { + compatible = "ti,pcm1792a"; + spi-max-frequency = <600000>; + }; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fb..45a9e3c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -55,6 +55,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -295,6 +296,9 @@ config SND_SOC_HDMI_CODEC config SND_SOC_PCM3008 tristate +config SND_SOC_PCM1792A + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd806..1f8ee0b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -43,6 +43,7 @@ snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -172,6 +173,7 @@ obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 0000000..99a4ee2 --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,245 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi <michael(a)amarulasolutions.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * 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/slab.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/spi/spi.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/of_device.h> + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .capture = { + .channels_min = 0, + .channels_max = 0, + }, + .ops = &pcm1792a_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); +#endif + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = pcm1792a_of_match, + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 0000000..7a83d1f --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * 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. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: core: init delayed_work for codec-codec links
by Richard Fitzgerald 05 Aug '13

05 Aug '13

If soc_probe_link_dais() finds a codec-codec link it skips creating a compress or pcm stream and links the DAIs together. But it must also init the delayed_work otherwise shutting down the DAI chain will fault when calling flush_delayed_work_sync() on the linked DAI. Pointing it to a dummy work callback is cleaner than taking special cases in the code to bypass the flush_delayed_work_sync(). Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4489c5b..bbe136c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -199,6 +199,10 @@ static ssize_t pmdown_time_set(struct device *dev, return count; } +static void dummy_delayed_work(struct work_struct *work) +{ +} + static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); #ifdef CONFIG_DEBUG_FS @@ -1428,6 +1432,8 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return ret; } } else { + INIT_DELAYED_WORK(&rtd->delayed_work, dummy_delayed_work); + /* link the DAI widgets */ play_w = codec_dai->playback_widget; capture_w = cpu_dai->capture_widget; -- 1.7.2.5

2 11

[alsa-devel] [PATCH] sound/sgtl5000: fix codec clock source transition to avoid clockless moment
by Oskar Schirmer 05 Aug '13

05 Aug '13

Powering down PLL before switching to a mode that does not use it is a bad idea. It would cause the SGTL5000 be without internal clock supply, especially on the I2C interface, which would make subsequent access to it fail. Thus, in case of not using PLL any longer, first set the mode control, then power down PLL. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/codecs/sgtl5000.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6c8a9e7..8ecb201 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -644,16 +644,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: dapm: Implement mixer input auto-disable
by Lars-Peter Clausen 05 Aug '13

05 Aug '13

Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Rebased onto the latest asoc/topic/dapm --- include/sound/soc-dapm.h | 17 +++++++- include/sound/soc.h | 28 ++++++------ sound/soc/soc-dapm.c | 108 +++++++++++++++++++++++++++++++++++++++++------ 3 files changed, 125 insertions(+), 28 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3575721..c728d28 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -280,14 +280,26 @@ struct device; { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_AUTODISABLE(xname, reg, shift, max, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_AUTODISABLE(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_volsw, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ @@ -484,6 +496,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_in, /* link to DAI structure */ snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ + snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ }; enum snd_soc_dapm_subclass { diff --git a/include/sound/soc.h b/include/sound/soc.h index b1e1f96..6201c6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -30,13 +30,13 @@ /* * Convenience kcontrol builders */ -#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \ +#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert, xautodisable) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .rreg = xreg, .shift = shift_left, \ .rshift = shift_right, .max = xmax, .platform_max = xmax, \ - .invert = xinvert}) -#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ - SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert) + .invert = xinvert, .autodisable = xautodisable}) +#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, xautodisable) \ + SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert, xautodisable) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert}) @@ -52,7 +52,7 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ @@ -68,7 +68,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } #define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ @@ -97,7 +97,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .info = snd_soc_info_volsw, \ @@ -119,7 +119,7 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \ - max, invert) } + max, invert, 0) } #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -190,14 +190,14 @@ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ - SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) } + SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -206,7 +206,7 @@ .tlv.p = (tlv_array), \ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ - .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } + .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -216,7 +216,7 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \ - xmax, xinvert) } + xmax, xinvert, 0) } #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -1088,7 +1088,9 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift, invert; + unsigned int reg, rreg, shift, rshift; + unsigned int invert:1; + unsigned int autodisable:1; }; struct soc_bytes { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5f64c16..0944bc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -47,6 +47,15 @@ #define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; +static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *wsource, struct snd_soc_dapm_widget *wsink, + const char *control, + int (*connected)(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink)); +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -73,16 +82,18 @@ static int dapm_up_seq[] = { [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, [snd_soc_dapm_line] = 10, - [snd_soc_dapm_post] = 11, + [snd_soc_dapm_kcontrol] = 11, + [snd_soc_dapm_post] = 12, }; static int dapm_down_seq[] = { [snd_soc_dapm_pre] = 0, - [snd_soc_dapm_adc] = 1, - [snd_soc_dapm_hp] = 2, - [snd_soc_dapm_spk] = 2, - [snd_soc_dapm_line] = 2, - [snd_soc_dapm_out_drv] = 2, + [snd_soc_dapm_kcontrol] = 1, + [snd_soc_dapm_adc] = 2, + [snd_soc_dapm_hp] = 3, + [snd_soc_dapm_spk] = 3, + [snd_soc_dapm_line] = 3, + [snd_soc_dapm_out_drv] = 3, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_switch] = 5, [snd_soc_dapm_mixer_named_ctl] = 5, @@ -176,6 +187,7 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( struct dapm_kcontrol_data { unsigned int value; + struct snd_soc_dapm_widget *widget; struct list_head paths; struct snd_soc_dapm_widget_list *wlist; }; @@ -184,6 +196,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol) { struct dapm_kcontrol_data *data; + struct soc_mixer_control *mc; data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) { @@ -195,6 +208,39 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, INIT_LIST_HEAD(&data->paths); + switch (widget->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + mc = (struct soc_mixer_control *)kcontrol->private_value; + + if (mc->autodisable) { + struct snd_soc_dapm_widget template; + + memset(&template, 0, sizeof(template)); + template.reg = mc->reg; + template.mask = (1 << fls(mc->max)) - 1; + template.shift = mc->shift; + if (mc->invert) + template.off_val = mc->max; + else + template.off_val = 0; + template.on_val = template.off_val; + template.id = snd_soc_dapm_kcontrol; + template.name = kcontrol->id.name; + + data->widget = snd_soc_dapm_new_control(widget->dapm, + &template); + if (!data->widget) { + kfree(data); + return -ENOMEM; + } + } + break; + default: + break; + } + kcontrol->private_data = data; return 0; @@ -203,6 +249,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + kfree(data->widget); kfree(data->wlist); kfree(data); } @@ -246,6 +293,21 @@ static void dapm_kcontrol_add_path(const struct snd_kcontrol *kcontrol, struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); list_add_tail(&path->list_kcontrol, &data->paths); + + if (data->widget) { + snd_soc_dapm_add_path(data->widget->dapm, data->widget, + path->source, NULL, NULL); + } +} + +static bool dapm_kcontrol_is_powered(const struct snd_kcontrol *kcontrol) +{ + struct dapm_kcontrol_data *data = snd_kcontrol_chip(kcontrol); + + if (!data->widget) + return true; + + return data->widget->power; } static struct list_head *dapm_kcontrol_get_path_list( @@ -275,6 +337,9 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; + if (data->widget) + data->widget->on_val = value; + data->value = value; return true; @@ -515,6 +580,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_spk: case snd_soc_dapm_line: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: p->connect = 1; break; /* does affect routing - dynamically connected */ @@ -880,6 +946,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -975,6 +1042,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: return 0; default: break; @@ -1523,7 +1591,7 @@ static void dapm_widget_update(struct snd_soc_card *card) unsigned int wi; int ret; - if (!update) + if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; wlist = dapm_kcontrol_get_wlist(update->kcontrol); @@ -1668,6 +1736,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -2335,6 +2404,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_dai_in: case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: + case snd_soc_dapm_kcontrol: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2717,6 +2787,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -2724,17 +2795,24 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; + unsigned int val; if (snd_soc_volsw_is_stereo(mc)) dev_warn(codec->dapm.dev, "ASoC: Control '%s' is stereo, which is not supported\n", kcontrol->id.name); - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + if (dapm_kcontrol_is_powered(kcontrol)) + val = (snd_soc_read(codec, reg) >> shift) & mask; + else + val = dapm_kcontrol_get_value(kcontrol); + mutex_unlock(&card->dapm_mutex); + if (invert) - ucontrol->value.integer.value[0] = - max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = max - val; + else + ucontrol->value.integer.value[0] = val; return 0; } @@ -2775,11 +2853,14 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (invert) val = max - val; - mask = mask << shift; - val = val << shift; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + dapm_kcontrol_set_value(kcontrol, val); + + mask = mask << shift; + val = val << shift; + change = snd_soc_test_bits(codec, reg, mask, val); if (change) { update.kcontrol = kcontrol; @@ -3179,6 +3260,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_clock_supply: + case snd_soc_dapm_kcontrol: w->power_check = dapm_supply_check_power; break; default: -- 1.8.0

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[alsa-devel] [PATCH v3 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 05 Aug '13

05 Aug '13

Changelog: v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 62 + .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1311 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 227 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1781 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] ASoC: omap-mcbsp 192000 freq
by Michael Trimarchi 05 Aug '13

05 Aug '13

Hi I'm working on having pcm1792a and omap platform. Right now I have already sent and rfc for the codec and I have a local build that change this file too to support 192000. What I don't understand is the meaning of: .period_bytes_min = 32, What is the relation between fifo_size and period_bytes_min? because at this frequency 32 bytes are not so much and the fifo of the port are >= 512 bytes. Is it possible to increase the period_bytes_min without violating the hw constraints? Michael .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, >From f1e6fcfd81e6642ba5695c848ac6aa729e7467aa Mon Sep 17 00:00:00 2001 From: Michael Trimarchi <michael(a)amarulasolutions.com> Date: Mon, 5 Aug 2013 12:26:35 +0200 Subject: [PATCH] OMAP MCBSP can support 192000 frame rate. Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7483efb..d285ddd 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -39,7 +39,7 @@ #include "mcbsp.h" #include "omap-mcbsp.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_192000) #define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \ xhandler_get, xhandler_put) \ -- 1.7.9.5

1 0

[alsa-devel] (no subject)
by Chih-Chung Chang 05 Aug '13

05 Aug '13

This is a revised patch which also checks if the mic is external.

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - Fix jack gating when auto_{mute, mic} is suppressed.
by Chih-Chung Chang 05 Aug '13

05 Aug '13

The snd_hda_jack_set_gating_jack() call didn't work when auto_{mute,mic} is suppressed because (1) am_entry is not filled with nid of the mic pin. (2) The jacks are not created (by snd_hda_jack_detect_enable_callback) before the snd_hda_jack_set_gating_jack call. Now we use the first input pin nid directly, and create the jack if it doesn't exist yet. Signed-off-by: Chih-Chung Chang <chihchung(a)chromium.org> diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..aa02c0f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -247,8 +247,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); + struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); if (!gated || !gating) return -EINVAL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..2b64de5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3258,11 +3258,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PROBE) { - if (snd_BUG_ON(!spec->gen.am_entry[1].pin || - !spec->gen.autocfg.hp_pins[0])) + int mic_pin = spec->gen.autocfg.inputs[0].pin; + int hp_pin = spec->gen.autocfg.hp_pins[0]; + + if (snd_BUG_ON(!mic_pin || !hp_pin)) return; - snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin, - spec->gen.autocfg.hp_pins[0]); + snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin); } } -- 1.8.3

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