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[alsa-devel] [PATCH 2b/4 v2] ASoC: add support for the sh7722 Migo-R board
by Guennadi Liakhovetski 29 Jan '10

29 Jan '10

Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978 codec, recording via external microphone and playback via headphones are implemented. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- v2 -> v3: 1. adjusted to the new wm8978 clocking scheme 2. added .hw_free() to switch off PLL diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 8072a6d..a86696b 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -55,4 +61,12 @@ config SND_FSI_DA7210 This option enables generic sound support for the FSI - DA7210 unit +config SND_SIU_MIGOR + tristate "SIU sound support on Migo-R" + depends on SH_MIGOR + select SND_SOC_SH4_SIU + select SND_SOC_WM8978 + help + This option enables sound support for the SH7722 Migo-R board + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index 1d0ec0a..8a5a192 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -16,7 +16,9 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o snd-soc-fsi-da7210-objs := fsi-da7210.o +snd-soc-migor-objs := migor.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o +obj-$(CONFIG_SND_SIU_MIGOR) += snd-soc-migor.o diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c new file mode 100644 index 0000000..3ccd9b3 --- /dev/null +++ b/sound/soc/sh/migor.c @@ -0,0 +1,222 @@ +/* + * ALSA SoC driver for Migo-R + * + * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> + * + * 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. + */ + +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/module.h> + +#include <asm/clock.h> + +#include <cpu/sh7722.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8978.h" +#include "siu.h" + +/* Default 8000Hz sampling frequency */ +static unsigned long codec_freq = 8000 * 512; + +static unsigned int use_count; + +/* External clock, sourced from the codec at the SIUMCKB pin */ +static unsigned long siumckb_recalc(struct clk *clk) +{ + return codec_freq; +} + +static struct clk_ops siumckb_clk_ops = { + .recalc = siumckb_recalc, +}; + +static struct clk siumckb_clk = { + .name = "siumckb_clk", + .id = -1, + .ops = &siumckb_clk_ops, + .rate = 0, /* initialised at run-time */ +}; + +static int migor_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->dai->codec_dai; + int ret; + unsigned int rate = params_rate(params); + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 13000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_DACCLK, 8); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8978_OPCLKRATE, rate * 512); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(rtd->dai->cpu_dai, SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + codec_freq = rate * 512; + /* + * This propagates the parent frequency change to children and + * recalculates the frequency table + */ + clk_set_rate(&siumckb_clk, codec_freq); + dev_dbg(codec_dai->dev, "%s: configure %luHz\n", __func__, codec_freq); + + ret = snd_soc_dai_set_sysclk(rtd->dai->cpu_dai, SIU_CLKB_EXT, + codec_freq / 2, SND_SOC_CLOCK_IN); + + if (!ret) + use_count++; + + return ret; +} + +static int migor_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + if (use_count) { + use_count--; + + if (!use_count) + snd_soc_dai_set_sysclk(codec_dai, WM8978_PLL, 0, + SND_SOC_CLOCK_IN); + } else { + dev_dbg(codec_dai->dev, "Unbalanced hw_free!\n"); + } + + return 0; +} + +static struct snd_soc_ops migor_dai_ops = { + .hw_params = migor_hw_params, + .hw_free = migor_hw_free, +}; + +static const struct snd_soc_dapm_widget migor_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Onboard Microphone", NULL), + SND_SOC_DAPM_MIC("External Microphone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone output connected to LHP/RHP, enable OUT4 for VMID */ + { "Headphone", NULL, "OUT4 VMID" }, + { "OUT4 VMID", NULL, "LHP" }, + { "OUT4 VMID", NULL, "RHP" }, + + /* On-board microphone */ + { "RMICN", NULL, "Mic Bias" }, + { "RMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Onboard Microphone" }, + + /* External microphone */ + { "LMICN", NULL, "Mic Bias" }, + { "LMICP", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "External Microphone" }, +}; + +static int migor_dai_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, migor_dapm_widgets, + ARRAY_SIZE(migor_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +/* migor digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link migor_dai = { + .name = "wm8978", + .stream_name = "WM8978", + .cpu_dai = &siu_i2s_dai, + .codec_dai = &wm8978_dai, + .ops = &migor_dai_ops, + .init = migor_dai_init, +}; + +/* migor audio machine driver */ +static struct snd_soc_card snd_soc_migor = { + .name = "Migo-R", + .platform = &siu_platform, + .dai_link = &migor_dai, + .num_links = 1, +}; + +/* migor audio subsystem */ +static struct snd_soc_device migor_snd_devdata = { + .card = &snd_soc_migor, + .codec_dev = &soc_codec_dev_wm8978, +}; + +static struct platform_device *migor_snd_device; + +static int __init migor_init(void) +{ + int ret; + + ret = clk_register(&siumckb_clk); + if (ret < 0) + return ret; + + /* Port number used on this machine: port B */ + migor_snd_device = platform_device_alloc("soc-audio", 1); + if (!migor_snd_device) { + ret = -ENOMEM; + goto epdevalloc; + } + + platform_set_drvdata(migor_snd_device, &migor_snd_devdata); + + migor_snd_devdata.dev = &migor_snd_device->dev; + + ret = platform_device_add(migor_snd_device); + if (ret) + goto epdevadd; + + return 0; + +epdevadd: + platform_device_put(migor_snd_device); +epdevalloc: + clk_unregister(&siumckb_clk); + return ret; +} + +static void __exit migor_exit(void) +{ + clk_unregister(&siumckb_clk); + platform_device_unregister(migor_snd_device); +} + +module_init(migor_init); +module_exit(migor_exit); + +MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski(a)gmx.de>"); +MODULE_DESCRIPTION("ALSA SoC Migor"); +MODULE_LICENSE("GPL v2");

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[alsa-devel] [PATCH 1/1] ASoC: AM3517: ASoC driver not getting compiled
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the Makefile. Whereas the config option defined in Kconfig is SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517 was not getting compiled. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/omap/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..19283e5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- 1.6.2.4

2 1

[alsa-devel] [PATCH] ASoC: AC97: SMDK-WM9713: Convert notes from cset to sset
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> It's more robust when references are provided in control names rather than numid. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/smdk_wm9713.c | 23 +++++++---------------- 1 files changed, 7 insertions(+), 16 deletions(-) diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 6fa2c9d..24fd39f 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -29,24 +29,15 @@ static struct snd_soc_card smdk; /* Playback (HeadPhone):- - Headphone Playback Switch - On - $ amixer cset numid=4 1 - - Right Headphone Out Mux - Headphone - $ amixer cset numid=92 2 - Left Headphone Out Mux - Headphone - $ amixer cset numid=93 2 - - Right HP Mixer PCM Playback Switch - On - $ amixer cset numid=75 1 - Left HP Mixer PCM Playback Switch - On - $ amixer cset numid=81 1 + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute Capture (LineIn):- - Right Capture Source - Line - $ amixer cset numid=86 2 - Left Capture Source - Line - $ amixer cset numid=87 2 + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' */ static struct snd_soc_dai_link smdk_dai = { -- 1.6.2.5

3 2

[alsa-devel] [PATCH 1/1] ASoC: AIC23: Fixing writes to non-existing registers in resume function
by Anuj Aggarwal 29 Jan '10

29 Jan '10

Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23 register in resume function because of which register writes happen on some non-existing registers. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb..da589d8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: TWL4030: Modify codec default settings
by Peter Ujfalusi 29 Jan '10

29 Jan '10

Change the legacy default register configuration, which left some internal components on. Now we have either DAPM, or other ways to control these bits, so there is no need to enable them by default. The affected parts: Disable ADCL and ADCR Disable ARXL2 and ARXR2 analog PGA (playback) Disable APLL by default Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 74f0d65..e0106a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,12 +64,12 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ - 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ - 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x4a, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4a, /* REG_ARXR2_APGA_CTL (0x1C) */ 0x00, /* REG_ATX2ARXPGA (0x1D) */ 0x00, /* REG_BT_IF (0x1E) */ 0x00, /* REG_BTPGA (0x1F) */ @@ -99,7 +99,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ - 0x16, /* REG_APLL_CTL (0x3A) */ + 0x06, /* REG_APLL_CTL (0x3A) */ 0x00, /* REG_DTMF_CTL (0x3B) */ 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ -- 1.6.6.1

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[alsa-devel] [PATCHv2 05/10] ASoC: AC97: SMDK: Add wm9713 machine driver
by jassisinghbrar＠gmail.com 29 Jan '10

29 Jan '10

From: Jassi Brar <jassi.brar(a)samsung.com> This patch adds the common machine driver for SMDKs that have a WM9713 codec attched to the AC97 controller. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 8 +++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/smdk_wm9713.c | 97 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 107 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/smdk_wm9713.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index ad3690e..d1c6f93 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -115,3 +115,11 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC3X select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index b7411bd..1117678 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -28,6 +28,7 @@ snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c new file mode 100644 index 0000000..7dd933f --- /dev/null +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -0,0 +1,97 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.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/device.h> +#include <sound/soc.h> + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + Playback (HeadPhone):- + Headphone Playback Switch - On + $ amixer cset numid=4 1 + + Right Headphone Out Mux - Headphone + $ amixer cset numid=92 2 + Left Headphone Out Mux - Headphone + $ amixer cset numid=93 2 + + Right HP Mixer PCM Playback Switch - On + $ amixer cset numid=75 1 + Left HP Mixer PCM Playback Switch - On + $ amixer cset numid=81 1 + + Capture (LineIn):- + Right Capture Source - Line + $ amixer cset numid=86 2 + Left Capture Source - Line + $ amixer cset numid=87 2 +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- 1.6.2.5

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[alsa-devel] [PATCH - hda intel conextant 1/1] Fixed microphone mixer for HP Compaq Presario F700 Added conextant patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea
by knprox＠mi.rr.com 28 Jan '10

28 Jan '10

From: Ken Prox <kprox(a)users.sourceforge.net> Signed-off-by: Ken Prox <kprox(a)users.sourceforge.net> --- pci/hda/patch_conexant.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 51 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 947785f..6142d75 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -1723,6 +1723,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_f700_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1813,6 +1829,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_f700_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1832,6 +1874,7 @@ enum { CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ + CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_MODELS }; @@ -1840,6 +1883,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", + [CXT5051_F700] = "hp 700" }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1849,6 +1893,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; @@ -1899,8 +1944,14 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; + case CXT5051_F700: + spec->init_verbs[0] = cxt5051_f700_init_verbs; + spec->mixers[0] = cxt5051_f700_mixers; + spec->no_auto_mic = 1; + break; } + return 0; } -- 1.6.2.5

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 28 Jan '10

28 Jan '10

The following changes since commit 0d34e91596ef537c2893a031f0483014bb82adf3: Guennadi Liakhovetski (1): ASoC: add a WM8978 codec driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Kuninori Morimoto (1): ASoC: fsi: Add spin lock operation for accessing shared area Mark Brown (3): ASoC: Remove version display from WM9713 ASoC: Add TLV information and additional volumes to WM9713 ASoC: Set codec->dev for AC97 devices Peter Ujfalusi (1): ASoC: TWL4030: Modify codec default settings sound/soc/codecs/twl4030.c | 8 +++--- sound/soc/codecs/wm9713.c | 64 ++++++++++++++++++++++++++++++------------- sound/soc/sh/fsi.c | 29 ++++++++++++++++++-- sound/soc/soc-core.c | 1 + 4 files changed, 75 insertions(+), 27 deletions(-)

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[alsa-devel] [PATCH] ASoC: fsi: Add spin lock operation for accessing shared area
by Kuninori Morimoto 28 Jan '10

28 Jan '10

fsi_master_xxx function should be protected by spin lock, because it are used from both FSI-A and FSI-B. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 29 ++++++++++++++++++++++++++--- 1 files changed, 26 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5f9f269..ebf3588 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -110,6 +110,7 @@ struct fsi_master { struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; + spinlock_t lock; }; /************************************************************************ @@ -168,30 +169,51 @@ static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) static int fsi_master_write(struct fsi_master *master, u32 reg, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_write((u32)(master->base + reg), data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_write((u32)(master->base + reg), data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static u32 fsi_master_read(struct fsi_master *master, u32 reg) { + u32 ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return 0; - return __fsi_reg_read((u32)(master->base + reg)); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_read((u32)(master->base + reg)); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } static int fsi_master_mask_set(struct fsi_master *master, u32 reg, u32 mask, u32 data) { + int ret; + unsigned long flags; + if ((reg < MREG_START) || (reg > MREG_END)) return -1; - return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_lock_irqsave(&master->lock, flags); + ret = __fsi_reg_mask_set((u32)(master->base + reg), mask, data); + spin_unlock_irqrestore(&master->lock, flags); + + return ret; } /************************************************************************ @@ -929,6 +951,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsib.base = master->base + 0x40; master->fsib.master = master; + spin_lock_init(&master->lock); pm_runtime_enable(&pdev->dev); pm_runtime_resume(&pdev->dev); -- 1.6.3.3

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[alsa-devel] [PATCH]Change the AZX_MAX_PCMS to 10
by Wei Ni 28 Jan '10

28 Jan '10

Hi, Takashi When I develop hdmi audio driver on the GT21x, I found a bug in the had_intel.c In hda_codec.c, it has define "[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },", it support up to device 9 for HDMI. But in hda_intel.c, it only define AZX_MAX_PCMS as 8. So the GT21x have 4 hdmi codecs, when run azx_attach_pcm_stream(), it will show error "Invalid PCM device number 8", and "... number 9", and return "-EINVAL". We should change the AZX_MAX_PCMS to 10. I attach the patch file, please check it. Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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