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

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de0070 to snd-hda
by Aaron Plattner 08 Jul '14

08 Jul '14

Vendor ID 0x10de0070 is used by a yet-to-be-named GPU chip. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- Another day, another Azalia codec ID. I verified that the patch_nvhdmi code is still needed for this chip. sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3e4417b0ddbe..1df5cf629a24 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3337,6 +3337,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, +{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -3394,6 +3395,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de0070"); MODULE_ALIAS("snd-hda-codec-id:10de0071"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- 2.0.1

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[alsa-devel] Integrate WM8974 codec with AM1808 processor
by Sangilikumar 08 Jul '14

08 Jul '14

Hello Engineers I'm working with TI's AM1808 ARM9 processor. My design in based on OMAP-L138 LCDK board. Here i replaced the TLV320AIC310 codec with WM8974 mono codec. Added the codec driver in board file and registered the sound card by static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "wm8974-hifi", .codec_name = "wm8974-codec.1-001a", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, }; But got some errors Sound card registered successfully but i got some errors wm8974-codec 1-001a: Failed to add route HPLOUT->Headphone Jack wm8974-codec 1-001a: dapm: unknown pin MONO_LOUT wm8974-codec 1-001a: dapm: unknown pin HPLCOM wm8974-codec 1-001a: dapm: unknown pin HPRCOM asoc: wm8974-hifi <-> davinci-mcasp.0 mapping ok ALSA device list: #0: DA850/OMAP-L138 EVM TCP cubic registered .... ALSA: Restoring mixer settings... Configuring network interfaces... Found hardware: "DA850_OMAP-L138" "" "" "" "" Hardware is initialized using a generic method /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #1 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #2 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #3 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #4 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #5 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #6 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #7 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #8 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #9 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #10 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #11 (No such file or directory) /usr/sbin/alsactl: set_control:1267: failed to obtain info for control #12 (No such file or directory) davinci_mdio davinci_mdio.0: resetting idled controller root@omapl138-lcdk:/aud# speaker-test speaker-test 1.0.24.2 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range asoc: machine hw_params failed from 8 to 2048 Using max buffer size 32768 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument root@omapl138-lcdk:/aud# Please let me know what are the changes i need to do in Linux files and drivers.

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[alsa-devel] [PATCH v2 0/7] ACPICA: Enable ACPICA prototypes for CONFIG_ACPI=n builds.
by Lv Zheng 08 Jul '14

08 Jul '14

This is the last series for ACPICA header inclusion cleanups. The whole story can be found below: 1. Unsafe inclusions: origin: https://lkml.org/lkml/2013/11/22/510 merged: https://lkml.org/lkml/2013/12/6/79 2. Header orders: origin: https://lkml.org/lkml/2014/4/8/57 merged: https://lkml.org/lkml/2014/5/12/111 3. Header stubs: origin: https://lkml.org/lkml/2014/12/18/149 merged: https://lkml.org/lkml/2014/4/29/716 This patchset contains patches that are parts of the last series sent before, so it is marked as v2 belonging to the same thread as this series: https://lkml.org/lkml/2014/12/18/149 Now we have cleaned up all mis-ordered ACPICA headers and implemented stubs for ACPICA prototypes (functions, global variables, macros), we finally can enable ACPICA headers for CONFIG_ACPI=n builds so that all ACPICA defined types are visible to all kernel source files. After exposing ACPICA types to all kernel sources, we can also sort the final wrong <acpi/acpi.h> inclusion out of <linux/sfi_acpi.h>. This patchset also includes a rule enforcement patch to avoid future inclusion mistakes. The patch set has passed a build/boot test on the following machines: Dell Inspiron Mini 1010 (i386) HP Compaq 8200 Elite SFF PC (x86-64) With the following kernel configuration items enabled: All drivers/acpi configurations All platform drivers All ACPI drivers 4 DRM drivers that implement ACPI opregion All buses with ACPI bindings All cpufreq configurations All xen configurations All ACPI table drivers They are tested twice with CONFIG_ACPI enabled/disabled. Also allyes/allno/allmod tests are done for ARCH=i386 and ARCH=x86_64. A boot test has been done to ARCH=x86_64 default configuration and it is confirmed that the ACPI devices are successfully created under /sys/bus/acpi/devices when CONFIG_ACPI is enabled. Lv Zheng (7): ACPICA: Linux: Add stub support for Linux specific variables and functions. ACPICA: Linux: Add stub implementation of ACPICA 64-bit mathematics. ACPICA: Linux: Add configuration item to indicate the architecture specific support. ACPICA: Linux: Allow ACPICA inclusion for CONFIG_ACPI=n builds. ACPI/SFI: Fix wrong <acpi/acpi.h> inclusion in SFI/ACPI wrapper - table definitions. ACPI: Cleanup useless ACPI inclusion. ACPI: Add support to force header inclusion rules for <acpi/acpi.h>. arch/ia64/Kconfig | 1 + arch/x86/Kconfig | 1 + drivers/acpi/Kconfig | 4 ++++ drivers/acpi/acpica/Makefile | 2 +- drivers/acpi/bus.c | 3 --- include/acpi/acpixf.h | 28 +++++++++++++++++----------- include/acpi/platform/aclinux.h | 12 ++++++++++++ include/acpi/platform/aclinuxex.h | 22 ++++++++++++++++++++++ include/linux/acpi.h | 6 +++--- include/linux/sfi_acpi.h | 3 --- sound/soc/intel/sst-haswell-dsp.c | 3 --- 11 files changed, 61 insertions(+), 24 deletions(-) -- 1.7.10

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[alsa-devel] [PATCH v5] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 07 Jul '14

07 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v5 - Consolidated dai_fmt call back to write serial control register once - https://patchwork.kernel.org/patch/4474531/ .../devicetree/bindings/sound/tas2552.txt | 25 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 576 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 120 ++++ 6 files changed, 753 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..00223dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,25 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - One of: + "ti,tas2552" - TAS2552 + +- reg - I2C slave address + +Optional properties: + +- enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..65e7627 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..fe305d6 --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,576 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_LIMIT_LVL_CTRL, 0x0c}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct mutex mutex; + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + unsigned char regs[TAS2552_VBAT_DATA]; + struct gpio_desc *enable_gpio; + u8 power_state:1; +}; + +static int tas2552_power(struct tas2552_data *data, u8 power) +{ + int ret = 0; + + mutex_lock(&data->mutex); + + if (power) { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 1); + + data->power_state = 1; + } else { + if (data->enable_gpio) + gpiod_set_value(data->enable_gpio, 0); + + data->power_state = 0; + } + + mutex_unlock(&data->mutex); + return ret; +} + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg = 0x0; + + if (sw_shutdown) + cfg1_reg |= TAS2552_SWS_MASK; + else + cfg1_reg &= ~TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + u8 wclk_reg; + struct snd_soc_codec *codec = dai->codec; + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (params_rate(params)) { + case 8000: + wclk_reg = TAS2552_8KHZ; + break; + case 11025: + wclk_reg = TAS2552_11_12KHZ; + break; + case 16000: + wclk_reg = TAS2552_16KHZ; + break; + case 32000: + wclk_reg = TAS2552_32KHZ; + break; + case 22050: + case 24000: + wclk_reg = TAS2552_22_24KHZ; + break; + case 44100: + case 48000: + wclk_reg = TAS2552_44_48KHZ; + break; + case 96000: + wclk_reg = TAS2552_88_96KHZ; + break; + default: + dev_vdbg(codec->dev, "Substream sample rate is not found\n"); + return -EINVAL; + } + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_3, TAS2552_WCLK_MASK, wclk_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 serial_format; + u8 serial_control_mask = 0x00; + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_MASTER_MASK) + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + if (serial_control_mask) { + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + pm_runtime_put(codec->dev); + } + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + snd_soc_write(codec, TAS2552_PLL_CTRL_1, 0x10); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, 0x00); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, 0x00); + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + pm_runtime_put(codec->dev); + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg = 0x0; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg |= TAS2552_MUTE_MASK; + else + cfg1_reg &= ~TAS2552_MUTE_MASK; + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + pm_runtime_put(codec->dev); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 1); + tas2552_power(tas2552, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + pm_runtime_get_sync(codec->dev); + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + pm_runtime_put(codec->dev); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + tas2552_power(tas2552, 1); + tas2552_sw_shutdown(tas2552, 0); + + pm_runtime_set_active(codec->dev); + pm_runtime_set_autosuspend_delay(codec->dev, 1000); + pm_runtime_use_autosuspend(codec->dev); + pm_runtime_enable(codec->dev); + pm_runtime_mark_last_busy(codec->dev); + pm_runtime_put_sync_autosuspend(codec->dev); + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_SWS_MASK | TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + /* 0dB gain */ + snd_soc_write(codec, TAS2552_PGA_GAIN, 0x10); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + /** + * Data sheet indicates to write 0x0c to 0x0d during init but no + * additional information is given to what it means. + */ + snd_soc_write(codec, TAS2552_LIMIT_LVL_CTRL, 0x0c); + /** + * Data sheet indicates to write 0x20 to 0x0e during init but no + * additional information is given to what it means. + */ + + snd_soc_write(codec, TAS2552_LIMIT_RATE_HYS, 0x20); + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_PLL_ENABLE | TAS2552_LIM_EN); + + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + tas2552_power(tas2552, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + pm_runtime_put(codec->dev); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + pm_runtime_get_sync(codec->dev); + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + mutex_init(&data->mutex); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..107b237 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,120 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_LIMIT_LVL_CTRL 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +#endif -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: max98090: Fix build warning
by Thierry Reding 07 Jul '14

07 Jul '14

From: Thierry Reding <treding(a)nvidia.com> The max98090_{suspend,resume}() functions are used for system sleep and therefore need to be guarded by CONFIG_PM_SLEEP rather than CONFIG_PM. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ca74f502a4cb..e806065d68cc 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2415,7 +2415,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); -- 2.0.1

2 1

[alsa-devel] [PATCH] ASoC: RT286: Fix silent at the beginning of stream
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch fix the issue that the output is almost silent at the beginning of starting a playback. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 20 +++++++++++++++++++- sound/soc/codecs/rt286.h | 5 +++++ 2 files changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e6f33ab..8103315 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -54,6 +54,7 @@ static struct reg_default rt286_index_def[] = { { 0x09, 0xd810 }, { 0x0a, 0x0060 }, { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, { 0x19, 0x0a17 }, { 0x20, 0x0020 }, @@ -62,6 +63,9 @@ static struct reg_default rt286_index_def[] = { { 0x4f, 0x50e9 }, { 0x50, 0x2c00 }, { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, }; #define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) @@ -902,14 +906,23 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_PREPARE: - if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); break; case SND_SOC_BIAS_STANDBY: snd_soc_write(codec, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); break; default: @@ -1152,6 +1165,11 @@ static int rt286_i2c_probe(struct i2c_client *i2c, /*Power down LDO2*/ regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + if (rt286->i2c->irq) { regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index 21c570f..b539b73 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -127,6 +127,7 @@ #define RT286_I2S_CTRL1 0x09 #define RT286_I2S_CTRL2 0x0a #define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d #define RT286_POWER_CTRL3 0x0f #define RT286_MIC1_DET_CTRL 0x19 #define RT286_MISC_CTRL1 0x20 @@ -135,6 +136,10 @@ #define RT286_CBJ_CTRL1 0x4f #define RT286_CBJ_CTRL2 0x50 #define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 /* SPDIF (0x06) */ #define RT286_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: add SOC_VALUE_ENUM_EXT
by Richard Fitzgerald 07 Jul '14

07 Jul '14

Adds an equivalent of SOC_ENUM_EXT for value enums Strictly speaking SOC_ENUM_EXT can also be used to define a value enum since the only difference is the get and set functions. But this doesn't look good in code because it is inconsistent with the normal control definitions. Adding a specific SOC_VALUE_ENUM_EXT is better for code clarity. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a5b4f6..1483920 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,6 +248,8 @@ .info = snd_soc_info_enum_double, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&xenum } +#define SOC_VALUE_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ + SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) #define SND_SOC_BYTES(xname, xbase, xregs) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: hda: Fix build warning
by Thierry Reding 07 Jul '14

07 Jul '14

From: Thierry Reding <treding(a)nvidia.com> The hda_tegra_disable_clocks() function is only used by the suspend and resume code, so it needs to be included in the #ifdef CONFIG_PM_SLEEP block to prevent the following warning: CC sound/pci/hda/hda_tegra.o sound/pci/hda/hda_tegra.c:238:13: warning: 'hda_tegra_disable_clocks' defined but not used [-Wunused-function] static void hda_tegra_disable_clocks(struct hda_tegra *data) ^ Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index dc4bb21ed17b..227990bc02e3 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -235,6 +235,7 @@ disable_hda: return rc; } +#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -242,7 +243,6 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } -#ifdef CONFIG_PM_SLEEP /* * power management */ -- 2.0.1

2 1

[alsa-devel] [PATCH v2] ASoC: rt5640: add ASRC support
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5640 series codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 79 ++++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5640.h | 3 ++ 2 files changed, 78 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efd..494b1b4 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1020,9 +1020,45 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_asrc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(w->codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (rt5640->asrc_en[RT5640_AIF1] || + rt5640->asrc_en[RT5640_AIF2]) { + snd_soc_update_bits(w->codec, + RT5640_DUMMY1, 0x70, 0x70); + snd_soc_update_bits(w->codec, + RT5640_JD_CTRL, 0x3, 0x3); + snd_soc_write(w->codec, RT5640_ASRC_1, 0x9b00); + snd_soc_write(w->codec, RT5640_ASRC_2, 0xf800); + } + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, RT5640_ASRC_1, 0x0); + snd_soc_write(w->codec, RT5640_ASRC_2, 0x0); + snd_soc_update_bits(w->codec, + RT5640_JD_CTRL, 0x3, 0); + snd_soc_update_bits(w->codec, + RT5640_DUMMY1, 0x70, 0); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY_S("ASRC Function", 1, SND_SOC_NOPM, + 0, 0, rt5640_asrc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* Input Side */ /* micbias */ SND_SOC_DAPM_SUPPLY("LDO2", RT5640_PWR_ANLG1, @@ -1276,6 +1312,9 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { + {"I2S1", NULL, "ASRC Function"}, + {"I2S2", NULL, "ASRC Function"}, + {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1636,10 +1675,17 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, rt5640->lrck[dai->id] = params_rate(params); pre_div = rl6231_get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); - if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", - rt5640->lrck[dai->id], dai->id); - return -EINVAL; + if ((rt5640->bclk_ratio[dai->id] != 64 && + rt5640->bclk_ratio[dai->id] != 32) || pre_div < 0) { + if (rt5640->sysclk < rt5640->lrck[dai->id] * 384) { + dev_err(codec->dev, "Unsupported clock setting"); + return -EINVAL; + } + dev_info(codec->dev, "Use ASRC\n"); + rt5640->asrc_en[dai->id] = true; + pre_div = 0; + } else { + rt5640->asrc_en[dai->id] = false; } frame_size = snd_soc_params_to_frame_size(params); if (frame_size < 0) { @@ -1698,6 +1744,17 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rt5640_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640->asrc_en[dai->id] = false; + + return 0; +} + static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; @@ -1864,6 +1921,16 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return 0; } +static int rt5640_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + rt5640->bclk_ratio[dai->id] = ratio; + + return 0; +} + static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1995,9 +2062,11 @@ static int rt5640_resume(struct snd_soc_codec *codec) static const struct snd_soc_dai_ops rt5640_aif_dai_ops = { .hw_params = rt5640_hw_params, + .hw_free = rt5640_hw_free, .set_fmt = rt5640_set_dai_fmt, .set_sysclk = rt5640_set_dai_sysclk, .set_pll = rt5640_set_dai_pll, + .set_bclk_ratio = rt5640_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5640_dai[] = { @@ -2214,6 +2283,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, } rt5640->hp_mute = 1; + rt5640->bclk_ratio[RT5640_AIF1] = 64; + rt5640->bclk_ratio[RT5640_AIF2] = 64; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, rt5640_dai, ARRAY_SIZE(rt5640_dai)); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 58ebe96..63b465d 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -2095,6 +2095,9 @@ struct rt5640_priv { int pll_out; bool hp_mute; + + bool asrc_en[RT5640_AIFS]; + int bclk_ratio[RT5640_AIFS]; }; #endif -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/3] ASoC: RT286: remove test code
by bardliao＠realtek.com 07 Jul '14

07 Jul '14

From: Bard Liao <bardliao(a)realtek.com> Remove test code. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index acfba9c..7c5f9d0 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -993,7 +993,6 @@ static int rt286_probe(struct snd_soc_codec *codec) codec->dapm.bias_level = SND_SOC_BIAS_OFF; rt286->codec = codec; - rt286->i2c->irq = 0; if (rt286->i2c->irq) { snd_soc_update_bits(codec, RT286_IRQ_CTRL, 0x2, 0x2); -- 1.8.1.1.439.g50a6b54

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