January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4]: ASoC SST v2: add mid asoc drivers
by Koul, Vinod 07 Jan '11

07 Jan '11

Hi, This patchset adds the asoc drivers for mid devices. These drivers along with sst driver form the audio subsystem in these class of devices. The current asoc driver supports the Intel(R) Medfield platform and sn95031 codec. This patchset addresses the comments received in previous v1 patches. This current patch set adds only sn95031 codec driver, the sst platform driver and Medfield machine driver. More codec drivers will be added eventually. This tisn95031 codec driver adds only playback path. Capture path and jack detection are todo. The Platform driver is currently dependent on the SST driver which is in the staging. Brief description of these patches with changes from previous v1: 1/4: tisn95031 Codec driver with config changes Fixed code format issues. Renamed codec to tisn95031. Cleaned up update_bits usage, fixed mode alias 2/4: SST Platform driver Cleaned up this, added locking. Renamed aptly to sst_platform. Fixed mode alias 3/4: Machine driver Moved device creation to core. Only add soc-audio here. Fixed mode alias, renamed to Medfield machine driver as this is for Medfield only 4/4: Make file and kconfig changes for machine and platform driver ~Harsha & Vinod

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[alsa-devel] Alsa hda-analyzer and Lenovo G560
by WH Bouterse 07 Jan '11

07 Jan '11

This may be the wrong list and I apologize ! A change in run hda-analyze script today 01/06/11 has rendered internal mic inoperable again for Lenovo G560 It has been necessary to run script upon each reboot to gain use of the internal mic. This was the known problem being worked on. Once again if I sent this to the wrong place I apologize and please direct me to the right list. WH Bouterse -

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[PATCH] ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx
by David Henningsson 07 Jan '11

07 Jan '11

BugLink: http://bugs.launchpad.net/bugs/689036 Many new Lenovos need the ideapad quirk. Also, since the auto parser for this chip is far from optimal, the regression risk is low (although not zero). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 9 +-------- 1 files changed, 1 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 74291f7..dc92e93 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3109,16 +3109,9 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { CXT5066_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), - SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G series (AMD)", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; -- 1.7.1 --------------000804070405060405080004--

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Remove NEO1973(GTA01) audio driver
by Seungwhan Youn 07 Jan '11

07 Jan '11

This patch remove to support NEO1973(GTA01) audio driver, because we don't need to support audio driver that is not supported in linux kernel. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/samsung/Kconfig | 9 - sound/soc/samsung/Makefile | 2 - sound/soc/samsung/neo1973_wm8753.c | 706 ------------------------------------ 3 files changed, 0 insertions(+), 717 deletions(-) delete mode 100644 sound/soc/samsung/neo1973_wm8753.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a6a6b5f..6cd865b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -34,15 +34,6 @@ config SND_SAMSUNG_SPDIF config SND_SAMSUNG_I2S tristate -config SND_SOC_SAMSUNG_NEO1973_WM8753 - tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01 - select SND_S3C24XX_I2S - select SND_SOC_WM8753 - help - Say Y if you want to add support for SoC audio on smdk2440 - with the WM8753. - config SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753 tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 705d4e8..11de33a 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -19,7 +19,6 @@ obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o -snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o @@ -37,7 +36,6 @@ snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o -obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_SOC_SAMSUNG_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c deleted file mode 100644 index c7a2451..0000000 --- a/sound/soc/samsung/neo1973_wm8753.c +++ /dev/null @@ -1,706 +0,0 @@ -/* - * neo1973_wm8753.c -- SoC audio for Neo1973 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.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. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/tlv.h> - -#include <asm/mach-types.h> -#include <asm/hardware/scoop.h> -#include <mach/regs-clock.h> -#include <mach/regs-gpio.h> -#include <mach/hardware.h> -#include <linux/io.h> -#include <mach/spi-gpio.h> - -#include <plat/regs-iis.h> - -#include "../codecs/wm8753.h" -#include "lm4857.h" -#include "dma.h" -#include "s3c24xx-i2s.h" - -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - -static struct snd_soc_card neo1973; -static struct i2c_client *i2c; - -static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_hifi_ops = { - .hw_params = neo1973_hifi_hw_params, - .hw_free = neo1973_hifi_hw_free, -}; - -static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); - if (ret < 0) - return ret; - - /* configure and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_voice_ops = { - .hw_params = neo1973_voice_hw_params, - .hw_free = neo1973_voice_hw_free, -}; - -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - } - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Call Mic", NULL), -}; - - -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Call Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 II. It is missing logic to detect hp/mic insertions and logic - * to re-route the audio in such an event. - */ -static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; - - pr_debug("Entered %s\n", __func__); - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "OUT4"); - snd_soc_dapm_nc_pin(dapm, "LINE1"); - snd_soc_dapm_nc_pin(dapm, "LINE2"); - - /* Add neo1973 specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); - - /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); - if (err < 0) - return err; - - /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(dapm, dapm_routes, - ARRAY_SIZE(dapm_routes)); - - snd_soc_dapm_sync(dapm); - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "bluetooth-dai", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .platform_name = "samsung-audio", - .cpu_dai_name = "s3c24xx-i2s", - .codec_dai_name = "wm8753-hifi", - .codec_name = "wm8753-codec.0-0x1a", - .init = neo1973_wm8753_init, - .ops = &neo1973_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .platform_name = "samsung-audio", - .cpu_dai_name = "bluetooth-dai", - .codec_dai_name = "wm8753-voice", - .codec_name = "wm8753-codec.0-0x1a", - .ops = &neo1973_voice_ops, -}, -}; - -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), -}; - -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } -}; - -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, -}; - -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - if (!machine_is_neo1973_gta01()) { - printk(KERN_INFO - "Only GTA01 hardware supported by ASoC driver\n"); - return -ENODEV; - } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_snd_device, &neo1973); - ret = platform_device_add(neo1973_snd_device); - - if (ret) { - platform_device_put(neo1973_snd_device); - return ret; - } - - ret = i2c_add_driver(&lm4857_i2c_driver); - - if (ret != 0) - platform_device_unregister(neo1973_snd_device); - - return ret; -} - -static void __exit neo1973_exit(void) -{ - pr_debug("Entered %s\n", __func__); - - i2c_del_driver(&lm4857_i2c_driver); - platform_device_unregister(neo1973_snd_device); -} - -module_init(neo1973_init); -module_exit(neo1973_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); -MODULE_LICENSE("GPL"); -- 1.6.2.5

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Re: [alsa-devel] [RFC/PATCH v7 01/12] media: Media device node support
by Greg KH 07 Jan '11

07 Jan '11

A: No. Q: Should I include quotations after my reply? http://daringfireball.net/2007/07/on_top On Thu, Jan 06, 2011 at 07:24:27PM -0500, Andy Walls wrote: > Why, yes, there is a standard: > > http://pubs.opengroup.org/onlinepubs/009695399/functions/xsh_chap02_03.html > > A somewhat verbose description of the errnos is in section 2.3. Ah, so perhaps -ENXIO is the correct thing to return with here. thanks, greg k-h

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Re: [alsa-devel] [RFC/PATCH v7 01/12] media: Media device node support
by Greg KH 07 Jan '11

07 Jan '11

On Fri, Jan 07, 2011 at 12:27:11AM +0100, Hans Verkuil wrote: > On Thursday, January 06, 2011 23:19:12 Greg KH wrote: > > <snip> > > > > > > +static ssize_t media_read(struct file *filp, char __user *buf, > > > > > + size_t sz, loff_t *off) > > > > > +{ > > > > > + struct media_devnode *mdev = media_devnode_data(filp); > > > > > + > > > > > + if (!mdev->fops->read) > > > > > + return -EINVAL; > > > > > + if (!media_devnode_is_registered(mdev)) > > > > > + return -EIO; > > > > > > > > How could this happen? > > > > > > This can happen when a USB device is disconnected for instance. > > > > But what's to keep that from happening on the next line as well? > > Nothing. > > > That > > doesn't seem like a check you can ever be sure about, so I wouldn't do > > it at all. > > Actually, there is a reason why this was done for v4l (and now the media > API): typically, once a USB disconnect happens V4L drivers will call > video_unregister_device() which calls device_unregister. Afterwards the > device node should reject any new file operations except for release(). > > Obviously, this check can be done in the driver as well, but doing this > check in the V4L core has the advantage of 1) consistent return codes and > 2) drivers no longer have to check. > > Of course, since the disconnect can happen at any time drivers still need > to be able to handle errors from the USB subsystem due to disconnects, but > that is something they always have to do. > > > > > > > And are you sure -EIO is correct? > > > > > > -ENXIO is probably better (I always confuse that with -ENODEV). > > I wondered why V4L uses -EIO and I think it is related to the V4L2 specification > of the read() function: > > EIO > I/O error. This indicates some hardware problem or a failure to communicate with > a remote device (USB camera etc.). > > Well, I guess a disconnect can be seen as a failure to communicate :-) > > I think that ENODEV is much better. After all, there is no device > anymore after a disconnect. > > Is there some standard for this? I've always thought that -ENODEV was good for when the device disappeared. thanks, greg k-h

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[alsa-devel] [PATCH] ASoC: cs4270: fix dynamic initialization of register cache
by Timur Tabi 06 Jan '11

06 Jan '11

The CS4270 codec driver dynamically initializes its register cache, rather than using a hard-coded array. Somwhere along the conversion to multi-compaonent, this feature broke. Because the register cache is more deeply ingrained into ASoC itself, the initialization of the cache has to happen earlier, in the I2C probe function. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 68 ++++++++++++++++---------------------------- 1 files changed, 25 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 3a582ca..4d36c0e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -114,6 +114,8 @@ static const char *supply_names[] = { struct cs4270_private { enum snd_soc_control_type control_type; void *control_data; + u8 reg_cache[CS4270_NUMREGS]; + struct snd_soc_codec_driver codec_drv; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ unsigned int slave_mode; @@ -263,38 +265,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, } /** - * cs4270_fill_cache - pre-fill the CS4270 register cache. - * @codec: the codec for this CS4270 - * - * This function fills in the CS4270 register cache by reading the register - * values from the hardware. - * - * This CS4270 registers are cached to avoid excessive I2C I/O operations. - * After the initial read to pre-fill the cache, the CS4270 never updates - * the register values, so we won't have a cache coherency problem. - * - * We use the auto-increment feature of the CS4270 to read all registers in - * one shot. - */ -static int cs4270_fill_cache(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache; - struct i2c_client *i2c_client = codec->control_data; - s32 length; - - length = i2c_smbus_read_i2c_block_data(i2c_client, - CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache); - - if (length != CS4270_NUMREGS) { - dev_err(codec->dev, "i2c read failure, addr=0x%x\n", - i2c_client->addr); - return -EIO; - } - - return 0; -} - -/** * cs4270_read_reg_cache - read from the CS4270 register cache. * @codec: the codec for this CS4270 * @reg: the register to read @@ -554,14 +524,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) codec->control_data = cs4270->control_data; - /* The I2C interface is set up, so pre-fill our register cache */ - - ret = cs4270_fill_cache(codec); - if (ret < 0) { - dev_err(codec->dev, "failed to fill register cache\n"); - return ret; - } - /* Disable auto-mute. This feature appears to be buggy. In some * situations, auto-mute will not deactivate when it should, so we want * this feature disabled by default. An application (e.g. alsactl) can @@ -707,7 +669,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) * Assign this variable to the codec_dev field of the machine driver's * snd_soc_device structure. */ -static struct snd_soc_codec_driver soc_codec_device_cs4270 = { +static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .probe = cs4270_probe, .remove = cs4270_remove, .suspend = cs4270_soc_suspend, @@ -757,12 +719,32 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } + /* Initialize the register cache. We do this dynamically because we + * don't know what the default values are. + */ + ret = i2c_smbus_read_i2c_block_data(i2c_client, + CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, + cs4270->reg_cache); + if (ret != CS4270_NUMREGS) { + dev_err(&i2c_client->dev, "i2c read failure, addr=0x%x\n", + i2c_client->addr); + return -EIO; + } + + /* We need to create a new copy of the snd_soc_codec_driver structure + * for each CS4270 because the .reg_cache_default field is different + * for each one. + */ + memcpy(&cs4270->codec_drv, &soc_codec_device_cs4270, + sizeof(soc_codec_device_cs4270)); + cs4270->codec_drv.reg_cache_default = cs4270->reg_cache; + i2c_set_clientdata(i2c_client, cs4270); cs4270->control_data = i2c_client; cs4270->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c_client->dev, - &soc_codec_device_cs4270, &cs4270_dai, 1); + ret = snd_soc_register_codec(&i2c_client->dev, &cs4270->codec_drv, + &cs4270_dai, 1); if (ret < 0) kfree(cs4270); return ret; -- 1.7.3.4

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Re: [alsa-devel] [PATCH 07/11] ASoC: tegra: Add tegra-das driver
by Stephen Warren 06 Jan '11

06 Jan '11

Mark Brown wrote: > On Fri, Dec 17, 2010 at 02:41:26PM -0700, Stephen Warren wrote: > > > /* > > * Terminology: > > * DAS: Digital audio switch (HW module controlled by this driver) > > * DAP: Digital audio port (port/pins on Tegra device) > > * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) > > * > > * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific > > * DAC, or another DAP. When DAPs are connected, one must be the master and > > * one the slave. Each DAC allows selection of a specific DAP for input, to > > * cater for the case where N DAPs are connected to 1 DAC for broadcast > > * output. > > * > > * This driver is dumb; no attempt is made to ensure that a valid routing > > * configuration is programmed. > > */ > > Reading this it's essentially just exporting point to point switches > within the DAS block. This really feels like it ought to be represented > as a CODEC driver to give runtime flexibility to userspace to control > the mux - it looks like the AP part of the system streams data into this > block which then has very flexible switching on to the externally > connected audio devices. Hmm. I don't see any examples of anything like Tegra's DAS in the existing ASoC code-base. So, I'm slightly unsure how to recast the DAS driver as a codec. In particular, in the machine driver, would I do something like: static struct snd_soc_dai_link harmony_i2s_das_link = { .name = "WM8903", .stream_name = "WM8903 PCM", .cpu_dai_name = "tegra-i2s-dai.0", .platform_name = "tegra-pcm-audio", .codec_name = "tegra-das.0", .codec_dai_name = "tegra-das.0-in", .ops = &harmony_i2s_das_ops, }; static struct snd_soc_dai_link harmony_das_wm8903_link = { .cpu_dai_name = "tegra-das.0-out", .codec_name = "wm8903-codec.0-001a", .codec_dai_name = "wm8903-hifi", .ops = &harmony_das_codec_ops, }; static struct snd_soc_dai_link *harmony_links[] = { & harmony_i2s_das_link, & harmony_das_wm8903_link, }; static struct snd_soc_card snd_soc_harmony = { .name = "tegra-harmony", .dai_link = &harmony_links, .num_links = ARRAY_SIZE(harmony_links), }; And then re-write the tegra-das driver to expose two DAIs somehow? Wouldn't that end up exposing two streams to user-space, one for each entry in harmony_links[]? -- nvpublic

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[alsa-devel] [PATCH 0/2] ASoC: Override codec compress_type from the machine driver
by Dimitris Papastamos 06 Jan '11

06 Jan '11

This patch adds support for overriding the compress_type of the codec driver from the machine driver. This patch depends on the previous series of patches that I've submitted with Subject "ASoC: Laying the groundwork for compress_type overriding". Dimitris Papastamos (2): ASoC: soc-cache: Use reg_def_copy instead of reg_cache_default ASoC: soc-core: Allow machine drivers to override compress_type include/sound/soc.h | 2 + sound/soc/soc-cache.c | 36 +++++++++++++------- sound/soc/soc-core.c | 87 +++++++++++++++++++++++++++++++++++++++++++----- 3 files changed, 104 insertions(+), 21 deletions(-) -- 1.7.3.2

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[alsa-devel] CX20585 - selecting wrong digital output
by Andy Robinson 06 Jan '11

06 Jan '11

I opened bug 0005235 in the bug tracking system regarding no output from my laptop's (Asus U50F) mini-TOSLINK jack, even though ALSA was detecting and providing a SPDIF output device. Using the alsa-info output, somebody was able to diagnose the issue, and I was able to confirm the problem. It looks like the CX20585 has two digital outputs: Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Since the CX20585 uses the CXT5066 patch in ALSA, node 0x21 is used for the SPDIF device. However, at least in the case of my laptop, that digital output is connected to a pin complex which is N/A: Pin Default 0x400001f0: [N/A] Line Out at Ext N/A However, node 0x12 IS connected to the optical out jack: Pin Default 0x034511f0: [Jack] SPDIF Out at Ext Left So, in this case, ALSA should use node 0x12 instead of node 0x21 for SPDIF out. I confirmed this with the latest snapshot by changing the digital output from 0x21 to 0x12, and that solved the problem. If there are no volunteers that want to take a look at a "real" solution for this :-), I might take a crack at it myself, but I'm not all that familiar with the driver right now. There seems to be different ways this could be handled: (1) Introduce a model that would use 0x12 instead of 0x21 as the digital output (2) Check the pin complexes associated with outputs 0x12 and 0x21 to determine which one is actually connected to a jack, and use that output as the SPDIF device instead of always choosing 0x21 (3) What if both digital outputs are connected to a valid jack? Should ALSA then create devices for both? As of now, it's not even creating a device for 0x12 even though it's a valid audio output. Any comments or suggestions are welcome.

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