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

[alsa-devel] [PATCH 2/2] ASoC: ssm2602: add support for 11.025kHz and 22.5kHz sample rates
by Stefan Kristiansson 29 Sep '14

29 Sep '14

This adds the necessary values to the constraint list and register values to the coefficient table in order to configure the device for 11.025kHz and 22.5kHz sample rates. Signed-off-by: Stefan Kristiansson <stefan.kristiansson(a)saunalahti.fi> --- sound/soc/codecs/ssm2602.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bb..da95976 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -192,7 +192,7 @@ static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { }; static const unsigned int ssm2602_rates_11289600[] = { - 8000, 44100, 88200, + 8000, 11025, 22050, 44100, 88200, }; static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { @@ -237,6 +237,16 @@ static const struct ssm2602_coeff ssm2602_coeff_table[] = { {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)}, {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)}, + /* 11.025k */ + {11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)}, + {16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)}, + {12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)}, + + /* 22.05k */ + {11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)}, + {16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)}, + {12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)}, + /* 44.1k */ {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)}, {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)}, @@ -467,7 +477,8 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ +#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) -- 1.9.1

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 29 Sep '14

29 Sep '14

First time bug submitter, forgive me if I'm not following proper procedure. OS: Arch Linux 64-bit Kernel version: 3.16.3-1-ARCH What's happening: When the Master channel is lowered below 35% in alsamixer, all audio coming out of the C-Media USB sound card is muted. Additional info: PulseAudio is installed Output of 'lsusb': Bus 005 Device 002: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter Alsa hw ID name: 'Set' Output of 'cat /proc/asound/Set/usbmixer' http://ix.io/exb Any assistance in resolving the issue is greatly appreciated

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[alsa-devel] [PATCH v3 0/5] Beaglebone-Black HDMI audio
by Jyri Sarha 29 Sep '14

29 Sep '14

Changes since v2: - Change compatible property from "ti,gpio-clock" to "ti,gpio-gate-clock" - Some minor cleanups The code has a functional dependency to: http://www.mail-archive.com/linux-omap@vger.kernel.org/msg109264.html Without the above patch the audio card does not probe. The code has been rebased on top of Linux 3.17-rc5. The patches bellow, the above dependency, and couple of commits to add BBB HDMI audio support to omap2plus_defconfig can be pulled from: https://github.com/jsarha/linux.git linux-master-bbb-hdmi-audio Cheers, Jyri Jyri Sarha (5): clk: ti: add "ti,gpio-gate-clock" controlled clock drm/tilcdc: Add I2S HDMI audio config for tda998x ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: davinci: HDMI audio build for AM33XX and TDA998x ARM: dts: am335x-boneblack: Add HDMI audio support .../bindings/clock/ti/gpio-gate-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 6 +- arch/arm/boot/dts/am335x-boneblack.dts | 52 +++++ drivers/clk/ti/Makefile | 2 +- drivers/clk/ti/gpio.c | 202 ++++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/davinci-evm.c | 82 +++++++- 8 files changed, 395 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/ti/gpio-gate-clock.txt create mode 100644 drivers/clk/ti/gpio.c -- 1.7.9.5

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[alsa-devel] ASoC: How to connect a single CPU DAI to two codecs?
by Sergej Sawazki 28 Sep '14

28 Sep '14

Hello, I'm writing an ALSA SoC machine driver for a board that has two mono codecs (left and right audio channel). Each codec is controlled over I2C and has itsown address.Both codecs are connected to the same I2S CPU DAI (as shown below). +--------- (adr: 0x1a) -----+ | | | +--------> codec_1 ----> (left channel audio) | | I2C I2S ----+ | | | +--------> codec_2 ----> (right channel audio) | | +--------- (adr: 0x1b) -----+ Can someone give me some advice on how to setup the 'snd_soc_card' and 'snd_soc_dai_link' in this case? Do I need two dai_links with the same 'cpu_dai_name' and 'platform_name'? Are there examples for such a configuration? I didn't had much luck in finding one in the mainline? Any help would be highly appreciated. Thanks! Sergej

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[alsa-devel] [RFC][PATCH 00/13 v2] ALSA: Enhancement for Dice driver
by Takashi Sakamoto 28 Sep '14

28 Sep '14

This patchset is to update my previous series. http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076730.html The previous series includes some bugs, as a result, playback/capture is not successfull. I've tested with actual device (TC Electronic, Impact Twin) and fix so as Dice driver drives the device. Currently this driver supports: - up to 192kHz - PCM playback/capture - MIDI playback/capture - Duplex streams with synchronization (with snd-firewire-lib) This driver doesn't support: - Synchronization between multiple devices on the same bus The rest of issue: - Handling bus reset Especially, just after powering on. Currently this driver often fails to probe device. I require more time to investigate. A workasound is to execute modprobe when the device is powered. Takashi Sakamoto (13): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc functionality into a file ALSA: dice: Add new functions for limitation of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for PCM capture ALSA: dice: Add support for MIDI capture/playback sound/firewire/Kconfig | 3 - sound/firewire/Makefile | 4 +- sound/firewire/dice-interface.h | 371 -------- sound/firewire/dice.c | 1511 -------------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/dice/dice-interface.h | 371 ++++++++ sound/firewire/dice/dice-midi.c | 173 ++++ sound/firewire/dice/dice-pcm.c | 389 ++++++++ sound/firewire/dice/dice-proc.c | 252 ++++++ sound/firewire/dice/dice-stream.c | 426 +++++++++ sound/firewire/dice/dice-transaction.c | 370 ++++++++ sound/firewire/dice/dice.c | 364 ++++++++ sound/firewire/dice/dice.h | 189 ++++ 14 files changed, 2728 insertions(+), 1888 deletions(-) delete mode 100644 sound/firewire/dice-interface.h delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c create mode 100644 sound/firewire/dice/dice-interface.h create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h -- 1.9.1

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[alsa-devel] [PATCH] ASoC:kirkwood: Don't raise an error when no DAI format
by Jean-Francois Moine 28 Sep '14

28 Sep '14

The two DAIs of the kirkwood controller have a unique PCM format. The simple-card sets the audio hardware definitions of all CPU DAIs. The PCM format is defined only when it is present in the DT. This patch prevents the controller to raise an error when the DT audio card definition by the simple card contains the PCM format of one CPU DAI only. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0704cd6..26d5f85 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -45,6 +45,8 @@ static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned long value; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case 0: + return 0; /* already done (simple-card) */ case SND_SOC_DAIFMT_RIGHT_J: mask = KIRKWOOD_I2S_CTL_RJ; break; -- 2.1.1

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[alsa-devel] [PATCH 1/2] ASoC: max98090: Enable both DMIC channels also when using mono configuration
by Jarkko Nikula 28 Sep '14

28 Sep '14

According to MAX98090 specification "Digital microphone clock (DMC) is enabled once both data channels are enabled.". Therefore both digital microphone data channels must be enabled also when using mono microphone configuration. Fix this by moving "DMICL_ENA" and "DMICR_ENA" supply widgets from "DMICL" and "DMICR" inputs to "DMIC Mux" in order to enable both data channels whenever there is active mono or stereo digital microphone input path. Use of "DMICL" and "DMICR" inputs are retained for informative source and in case the driver would find use for exact digital microphone configuration in the future. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- for-next, only byt-max98090 uses DMIC and I noticed this after disabling unused codec pins. --- sound/soc/codecs/max98090.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f2a3f30a5d9f..d519294f57c7 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,8 +1311,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, - {"DMICL", NULL, "DMICL_ENA"}, - {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1370,6 +1368,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, + {"DMIC Mux", "DMIC", "DMICL_ENA"}, + {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- 2.1.0

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[alsa-devel] [PATCH] ASoC: core: fix possible ZERO_SIZE_PTR pointer dereferencing error.
by Xiubo Li 28 Sep '14

28 Sep '14

Since we cannot make sure the 'params->num_regs' will always be none zero here, and then if it equals to zero, the kmemdup() will return ZERO_SIZE_PTR, which equals to ((void *)16). So this patch fix this with just doing the zero check before calling kmemdup(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7b5a7ce..263514a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3051,7 +3051,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; -- 2.1.0.27.g96db324

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[alsa-devel] [PATCH] ASoC: ssm4567: Add driver for Analog Devices SSM4567 amplifier
by Anatol Pomozov 27 Sep '14

27 Sep '14

Analog Devices SSM4567 is a boost class-D audio amplifier. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/ssm4567.txt | 15 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm4567.c | 350 +++++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ssm4567.txt create mode 100644 sound/soc/codecs/ssm4567.c diff --git a/Documentation/devicetree/bindings/sound/ssm4567.txt b/Documentation/devicetree/bindings/sound/ssm4567.txt new file mode 100644 index 0000000..ec3d9e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ssm4567.txt @@ -0,0 +1,15 @@ +Analog Devices SSM4567 audio amplifier + +This device supports I2C only. + +Required properties: + - compatible : Must be "adi,ssm4567" + - reg : the I2C address of the device. This will either be 0x34 (LR_SEL/ADDR connected to AGND), + 0x35 (LR_SEL/ADDR connected to IOVDD) or 0x36 (LR_SEL/ADDR open). + +Example: + + ssm4567: ssm4567@34 { + compatible = "adi,ssm4567"; + reg = <0x34>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8ab1547..da45b5a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -94,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2518 if I2C select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C + select SND_SOC_SSM4567 if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C @@ -546,6 +547,9 @@ config SND_SOC_SSM2602_I2C select SND_SOC_SSM2602 tristate +config SND_SOC_SSM4567 + tristate "Analog Devices ssm4567 amplifier driver support" + config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index afba944..5dce451 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -95,6 +95,7 @@ snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o +snd-soc-ssm4567-objs := ssm4567.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o @@ -266,6 +267,7 @@ obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o +obj-$(CONFIG_SND_SOC_SSM4567) += snd-soc-ssm4567.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c new file mode 100644 index 0000000..fb92248 --- /dev/null +++ b/sound/soc/codecs/ssm4567.c @@ -0,0 +1,350 @@ +/* + * SSM4567 amplifier audio driver + * + * Copyright 2014 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * + * Based on code copyright/by: + * Copyright 2013 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define SSM4567_REG_POWER_CTRL 0x00 +#define SSM4567_REG_AMP_SNS_CTRL 0x01 +#define SSM4567_REG_DAC_CTRL 0x02 +#define SSM4567_REG_DAC_VOLUME 0x03 +#define SSM4567_REG_SAI_CTRL_1 0x04 +#define SSM4567_REG_SAI_CTRL_2 0x05 +#define SSM4567_REG_SAI_PLACEMENT_1 0x06 +#define SSM4567_REG_SAI_PLACEMENT_2 0x07 +#define SSM4567_REG_SAI_PLACEMENT_3 0x08 +#define SSM4567_REG_SAI_PLACEMENT_4 0x09 +#define SSM4567_REG_SAI_PLACEMENT_5 0x0a +#define SSM4567_REG_SAI_PLACEMENT_6 0x0b +#define SSM4567_REG_BATTERY_V_OUT 0x0c +#define SSM4567_REG_LIMITER_CTRL_1 0x0d +#define SSM4567_REG_LIMITER_CTRL_2 0x0e +#define SSM4567_REG_LIMITER_CTRL_3 0x0f +#define SSM4567_REG_STATUS_1 0x10 +#define SSM4567_REG_STATUS_2 0x11 +#define SSM4567_REG_FAULT_CTRL 0x12 +#define SSM4567_REG_PDM_CTRL 0x13 +#define SSM4567_REG_MCLK_RATIO 0x14 +#define SSM4567_REG_BOOST_CTRL_1 0x15 +#define SSM4567_REG_BOOST_CTRL_2 0x16 +#define SSM4567_REG_SOFT_RESET 0xff + +/* POWER_CTRL */ +#define SSM4567_POWER_APWDN_EN BIT(7) +#define SSM4567_POWER_BSNS_PWDN BIT(6) +#define SSM4567_POWER_VSNS_PWDN BIT(5) +#define SSM4567_POWER_ISNS_PWDN BIT(4) +#define SSM4567_POWER_BOOST_PWDN BIT(3) +#define SSM4567_POWER_AMP_PWDN BIT(2) +#define SSM4567_POWER_VBAT_ONLY BIT(1) +#define SSM4567_POWER_SPWDN BIT(0) + +/* DAC_CTRL */ +#define SSM4567_DAC_HV BIT(7) +#define SSM4567_DAC_MUTE BIT(6) +#define SSM4567_DAC_HPF BIT(5) +#define SSM4567_DAC_LPM BIT(4) +#define SSM4567_DAC_FS_MASK 0x7 +#define SSM4567_DAC_FS_8000_12000 0x0 +#define SSM4567_DAC_FS_16000_24000 0x1 +#define SSM4567_DAC_FS_32000_48000 0x2 +#define SSM4567_DAC_FS_64000_96000 0x3 +#define SSM4567_DAC_FS_128000_192000 0x4 + +struct ssm4567 { + struct regmap *regmap; +}; + +static const struct reg_default ssm4567_reg_defaults[] = { + { SSM4567_REG_POWER_CTRL, 0x81 }, + { SSM4567_REG_AMP_SNS_CTRL, 0x09 }, + { SSM4567_REG_DAC_CTRL, 0x32 }, + { SSM4567_REG_DAC_VOLUME, 0x40 }, + { SSM4567_REG_SAI_CTRL_1, 0x00 }, + { SSM4567_REG_SAI_CTRL_2, 0x08 }, + { SSM4567_REG_SAI_PLACEMENT_1, 0x01 }, + { SSM4567_REG_SAI_PLACEMENT_2, 0x20 }, + { SSM4567_REG_SAI_PLACEMENT_3, 0x32 }, + { SSM4567_REG_SAI_PLACEMENT_4, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_5, 0x07 }, + { SSM4567_REG_SAI_PLACEMENT_6, 0x07 }, + { SSM4567_REG_BATTERY_V_OUT, 0x00 }, + { SSM4567_REG_LIMITER_CTRL_1, 0xa4 }, + { SSM4567_REG_LIMITER_CTRL_2, 0x73 }, + { SSM4567_REG_LIMITER_CTRL_3, 0x00 }, + { SSM4567_REG_STATUS_1, 0x00 }, + { SSM4567_REG_STATUS_2, 0x00 }, + { SSM4567_REG_FAULT_CTRL, 0x30 }, + { SSM4567_REG_PDM_CTRL, 0x40 }, + { SSM4567_REG_MCLK_RATIO, 0x11 }, + { SSM4567_REG_BOOST_CTRL_1, 0x03 }, + { SSM4567_REG_BOOST_CTRL_2, 0x00 }, + { SSM4567_REG_SOFT_RESET, 0x00 }, +}; + + +static bool ssm4567_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_BOOST_CTRL_2: + return true; + default: + return false; + } + +} + +static bool ssm4567_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_POWER_CTRL ... SSM4567_REG_SAI_PLACEMENT_6: + case SSM4567_REG_LIMITER_CTRL_1 ... SSM4567_REG_LIMITER_CTRL_3: + case SSM4567_REG_FAULT_CTRL ... SSM4567_REG_BOOST_CTRL_2: + /* The datasheet states that soft reset register is read-only, + * but logically it is write-only. */ + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static bool ssm4567_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SSM4567_REG_BATTERY_V_OUT: + case SSM4567_REG_STATUS_1 ... SSM4567_REG_STATUS_2: + case SSM4567_REG_SOFT_RESET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(ssm4567_vol_tlv, -7125, 2400); + +static const struct snd_kcontrol_new ssm4567_snd_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", SSM4567_REG_DAC_VOLUME, 0, + 0xff, 1, ssm4567_vol_tlv), + SOC_SINGLE("Master Playback Switch", SSM4567_REG_DAC_CTRL, 6, 1, 1), + SOC_SINGLE("DAC Low Power Mode Switch", SSM4567_REG_DAC_CTRL, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM4567_REG_POWER_CTRL, 2, 1), + + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route ssm4567_routes[] = { + { "OUT", NULL, "DAC" }, +}; + +static int ssm4567_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 ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int dacfs; + + if (rate >= 8000 && rate <= 12000) + dacfs = SSM4567_DAC_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + dacfs = SSM4567_DAC_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + dacfs = SSM4567_DAC_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + dacfs = SSM4567_DAC_FS_64000_96000; + else if (rate >= 64000 && rate <= 96000) + dacfs = SSM4567_DAC_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + dacfs = SSM4567_DAC_FS_128000_192000; + else + return -EINVAL; + + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_FS_MASK, dacfs); +} + +static int ssm4567_mute(struct snd_soc_dai *dai, int mute) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + val = mute ? SSM4567_DAC_MUTE : 0; + return regmap_update_bits(ssm4567->regmap, SSM4567_REG_DAC_CTRL, + SSM4567_DAC_MUTE, val); +} + +static int ssm4567_set_power(struct ssm4567 *ssm4567, bool enable) +{ + int ret = 0; + + if (!enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, SSM4567_POWER_SPWDN); + regcache_mark_dirty(ssm4567->regmap); + } + + regcache_cache_only(ssm4567->regmap, !enable); + + if (enable) { + ret = regmap_update_bits(ssm4567->regmap, + SSM4567_REG_POWER_CTRL, + SSM4567_POWER_SPWDN, 0x00); + regcache_sync(ssm4567->regmap); + } + + return ret; +} + +static int ssm4567_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct ssm4567 *ssm4567 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = ssm4567_set_power(ssm4567, true); + break; + case SND_SOC_BIAS_OFF: + ret = ssm4567_set_power(ssm4567, false); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +#define SSM4567_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ssm4567_dai_ops = { + .hw_params = ssm4567_hw_params, + .digital_mute = ssm4567_mute, +}; + +static struct snd_soc_dai_driver ssm4567_dai = { + .name = "ssm4567-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SSM4567_FORMATS, + }, + .ops = &ssm4567_dai_ops, +}; + +static struct snd_soc_codec_driver ssm4567_codec_driver = { + .set_bias_level = ssm4567_set_bias_level, + .idle_bias_off = true, + + .controls = ssm4567_snd_controls, + .num_controls = ARRAY_SIZE(ssm4567_snd_controls), + .dapm_widgets = ssm4567_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ssm4567_dapm_widgets), + .dapm_routes = ssm4567_routes, + .num_dapm_routes = ARRAY_SIZE(ssm4567_routes), +}; + +static const struct regmap_config ssm4567_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .max_register = SSM4567_REG_SOFT_RESET, + .readable_reg = ssm4567_readable_reg, + .writeable_reg = ssm4567_writeable_reg, + .volatile_reg = ssm4567_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ssm4567_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ssm4567_reg_defaults), +}; + +static int ssm4567_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ssm4567 *ssm4567; + int ret; + + ssm4567 = devm_kzalloc(&i2c->dev, sizeof(*ssm4567), GFP_KERNEL); + if (ssm4567 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ssm4567); + + ssm4567->regmap = devm_regmap_init_i2c(i2c, &ssm4567_regmap_config); + if (IS_ERR(ssm4567->regmap)) + return PTR_ERR(ssm4567->regmap); + + ret = regmap_write(ssm4567->regmap, SSM4567_REG_SOFT_RESET, 0x00); + if (ret) + return ret; + + ret = ssm4567_set_power(ssm4567, false); + if (ret) + return ret; + + return snd_soc_register_codec(&i2c->dev, &ssm4567_codec_driver, + &ssm4567_dai, 1); +} + +static int ssm4567_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm4567_i2c_ids[] = { + { "ssm4567", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm4567_i2c_ids); + +static struct i2c_driver ssm4567_driver = { + .driver = { + .name = "ssm4567", + .owner = THIS_MODULE, + }, + .probe = ssm4567_i2c_probe, + .remove = ssm4567_i2c_remove, + .id_table = ssm4567_i2c_ids, +}; +module_i2c_driver(ssm4567_driver); + +MODULE_DESCRIPTION("ASoC SSM4567 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); -- 2.1.0.rc2.206.gedb03e5

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Re: [alsa-devel] [PATCH v3 1/5] clk: ti: add "ti, gpio-gate-clock" controlled clock
by Tomi Valkeinen 27 Sep '14

27 Sep '14

On 19/09/14 16:12, Nishanth Menon wrote: > On 09/19/2014 08:07 AM, Tomi Valkeinen wrote: >> On 16/09/14 23:40, Jyri Sarha wrote: >>> The added ti,gpio-gate-clock is a basic clock that can be enabled and >>> disabled trough a gpio output. The DT binding document for the clock >>> is also added. For EPROBE_DEFER handling the registering of the clock >>> has to be delayed until of_clk_get() call time. >>> >>> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> >>> --- >>> .../bindings/clock/ti/gpio-gate-clock.txt | 21 ++ >>> drivers/clk/ti/Makefile | 2 +- >>> drivers/clk/ti/gpio.c | 202 ++++++++++++++++++++ >>> 3 files changed, 224 insertions(+), 1 deletion(-) >>> create mode 100644 Documentation/devicetree/bindings/clock/ti/gpio-gate-clock.txt >>> create mode 100644 drivers/clk/ti/gpio.c >> >> Why is this a TI clock? Sounds like a generic one to me. > > Like thread: https://lkml.org/lkml/2014/9/5/284 ? Right, I should've read the earlier versions before making any smart comments =). Tomi

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