November 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Sound support for Neuros OSD2 Davinci based board.
by Mark Brown 27 Nov '09

27 Nov '09

On Fri, Nov 27, 2009 at 06:14:26PM +0500, A. Porodko wrote: > Mark Brown wrote: > > I raised the same query when Andrey sent an off-list copy to me by > > mistake - he said that in future he intends to extend the driver to > > have additional features not present on the EVM (though for me that > > could always change the || later). > Understood. Shall I change if if to if || then? Honestly I'm not that bothered, I was mostly following up to the list so that readers and the archives could see the answer to the query (I guessed you may have mistakenly replied off list).

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[alsa-devel] [PATCH] ASoC: AIC23: Fixing infinite loop in resume path
by Anuj Aggarwal 27 Nov '09

27 Nov '09

This patch fixes two issues: a) Infinite loop in resume function b) Writes to non-existing registers in resume function Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8b..90a0264 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + for (reg = 0; reg < TLV320AIC23_RESET; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } -- 1.6.2.4

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[alsa-devel] [PATCH 2/3] ALSA: ARM: add Raumfeld audio support
by Daniel Mack 27 Nov '09

27 Nov '09

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/raumfeld.c | 416 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 427 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/raumfeld.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181..9c3bfd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b4..f3e08fd 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 0000000..45e3f1c --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,416 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel(a)caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg(a)slimlogic.co.uk> + * Richard Purdie <richard(a)openedhand.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/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/fixed.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/mfp-pxa3xx.h> +#include <mach/mfp-pxa300.h> +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET MFP_PIN_GPIO38 +#define GPIO_MCLK_RESET MFP_PIN_GPIO111 +#define GPIO_CODEC_RESET MFP_PIN_GPIO120 +#define GPIO_AUDIO_VA_ENABLE MFP_PIN_GPIO124 + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +/* Fixed regulator for AUDIO_VA */ + +static struct regulator_consumer_supply audio_va_consumer_supply = + REGULATOR_SUPPLY("va", "soc-audio.0"); + +struct regulator_init_data audio_va_initdata = { + .consumer_supplies = &audio_va_consumer_supply, + .num_consumer_supplies = 1, + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config audio_va_config = { + .supply_name = "audio_va", + .microvolts = 5000000, + .gpio = mfp_to_gpio(GPIO_AUDIO_VA_ENABLE), + .enable_high = 1, + .enabled_at_boot = 0, + .init_data = &audio_va_initdata, +}; + +static struct platform_device audio_va_device = { + .name = "reg-fixed-voltage", + .id = 0, + .dev = { + .platform_data = &audio_va_config, + }, +}; + +/* Dummy supplies for Codec's VD/VLC */ + +static struct regulator_consumer_supply audio_dummy_supplies[] = { + REGULATOR_SUPPLY("vd", "soc-audio.0"), + REGULATOR_SUPPLY("vlc", "soc-audio.0"), +}; + +struct regulator_init_data audio_dummy_initdata = { + .consumer_supplies = audio_dummy_supplies, + .num_consumer_supplies = ARRAY_SIZE(audio_dummy_supplies), + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, +}; + +static struct fixed_voltage_config audio_dummy_config = { + .supply_name = "audio_vd", + .microvolts = 3300000, + .gpio = -1, + .init_data = &audio_dummy_initdata, +}; + +static struct platform_device audio_supply_dummy_device = { + .name = "reg-fixed-voltage", + .id = 1, + .dev = { + .platform_data = &audio_dummy_config, + }, +}; + +static struct platform_device *regulator_devices[] = { + &audio_va_device, + &audio_supply_dummy_device, +}; + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(mfp_to_gpio(GPIO_MCLK_RESET), 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(mfp_to_gpio(GPIO_SPDIF_RESET), 1); + gpio_set_value(mfp_to_gpio(GPIO_CODEC_RESET), 1); + } else { + gpio_set_value(mfp_to_gpio(GPIO_MCLK_RESET), 0); + gpio_set_value(mfp_to_gpio(GPIO_SPDIF_RESET), 0); + gpio_set_value(mfp_to_gpio(GPIO_CODEC_RESET), 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_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; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_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; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static int raumfeld_ak4104_init(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .init = raumfeld_ak4104_init, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + platform_add_devices(regulator_devices, ARRAY_SIZE(regulator_devices)); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + platform_device_put(&audio_va_device); + platform_device_put(&audio_supply_dummy_device); + + gpio_free(mfp_to_gpio(GPIO_MCLK_RESET)); + gpio_free(mfp_to_gpio(GPIO_CODEC_RESET)); + gpio_free(mfp_to_gpio(GPIO_SPDIF_RESET)); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); -- 1.6.5.2

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[alsa-devel] ugly sound output (intel-hda) [was: mmotm 2009-11-24-16-47 uploaded]
by Jiri Slaby 27 Nov '09

27 Nov '09

On 11/25/2009 01:47 AM, akpm(a)linux-foundation.org wrote: > The mm-of-the-moment snapshot 2009-11-24-16-47 has been uploaded to Hi, when I boot into that kernel, I hear, since the boot phase, short scratches repeatedly every (just a guess) 100 ms. # cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xffa74000 irq 29 # cat /proc/asound/card0/codec#2 Codec: SigmaTel STAC9274D Address: 2 Function Id: 0x1 Vendor Id: 0x83847621 Subsystem Id: 0x00000100 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0e, stepsize=0x05, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=3, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Analog Loopback: 0x00 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Front Playback Volume", index=0, device=0 Control: name="Front Playback Switch", index=0, device=0 Device: name="STAC92xx Analog", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Side Playback Volume", index=0, device=0 Control: name="Side Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Center Playback Volume", index=0, device=0 Control: name="Center Playback Switch", index=0, device=0 Control: name="LFE Playback Volume", index=0, device=0 Control: name="LFE Playback Switch", index=0, device=0 Control: name="Swap Center/LFE Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=4 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Surround Playback Volume", index=0, device=0 Control: name="Surround Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=2 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 Control: name="Headphone Playback Switch", index=0, device=0 Amp-Out caps: N/A Amp-Out vals: [0x67 0x67] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Device: name="STAC92xx Analog", type="Audio", device=0 Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1b Processing caps: benign=0, ncoeff=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1c Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x1d Processing caps: benign=0, ncoeff=0 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02214020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=01, enabled=1 Connection: 3 0x02 0x03 0x06* Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Control: name="Front Mic Jack Mode", index=0, device=0 Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a19080: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x8, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Connection: 3 0x02* 0x03 0x06 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Control: name="Line Jack Mode", index=0, device=0 Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0181304e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0xe Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Control: name="Mic Jack Mode", index=0, device=0 Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19040: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Connection: 1 0x04 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x503301f0: [N/A] CD at Int N/A Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x50a001f0: [N/A] Mic at Int N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Control: name="Mux Capture Volume", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 9 0x0e* 0x12 0x0f 0x0b 0x0c 0x0d 0x0a 0x10 0x11 Node 0x18 [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 Amp-In caps: N/A Amp-In vals: [0x08 0x08] Connection: 1 0x15 Node 0x19 [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x16 Node 0x1a [Audio Selector] wcaps 0x300103: Stereo Amp-In Control: name="Capture Volume", index=2, device=0 Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x17 Node 0x1b [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Connection: 3 0x18* 0x13 0x14 Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=1, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x19* 0x13 0x14 Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Control: name="Capture Switch", index=2, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 3 0x1a* 0x13 0x14 Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="STAC92xx Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x1f [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Delay: 3 samples Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Device: name="STAC92xx Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x22 Node 0x21 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01442170: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x7, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 5 0x1e* 0x1f 0x1b 0x1c 0x1d Node 0x22 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x81c42090: [Fixed] SPDIF In at Ext Rear Conn = RCA, Color = Grey DefAssociation = 0x9, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x23 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x24 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=127 Connection: 5 0x02 0x03 0x04 0x05 0x06 It's a regression against, I think, 2009-11-13-19-59. thanks, -- js

2 5

[alsa-devel] [PATCH] ASoC: AM3517: Fix AIC23 suspend/resume hang
by Anuj Aggarwal 27 Nov '09

27 Nov '09

System hang is observed While trying to do suspend. It was because of an infinite loop in the AIC23 resume path which was trying to restore AIC23 register values from the register cache. This patch fixes the problem by correcting the resume path and properly activating/deactivating the digital interface while doing the suspend / off transitions. Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- sound/soc/codecs/tlv320aic23.c | 16 +++------------- 1 files changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 6b24d8b..cabe60c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -565,13 +565,12 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* everything off except vref/vmid, */ - tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + /* Activate the digital interface */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ + /* Deactivate the digital interface */ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); - tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); break; } codec->bias_level = level; @@ -615,7 +614,6 @@ static int tlv320aic23_suspend(struct platform_device *pdev, struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -625,14 +623,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; - u16 reg; - - /* Sync reg_cache with the hardware */ - for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { - u16 val = tlv320aic23_read_reg_cache(codec, reg); - tlv320aic23_write(codec, reg, val); - } tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); -- 1.6.2.4

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[alsa-devel] Problem with ASUSTeK Computer Inc. Device 8398
by Bráulio B O Bhavamitra 27 Nov '09

27 Nov '09

Hello I'm using Ubuntu Karmic on a Asus T91 netbook. The sounds works except for the internal MIC so I have installed linux-backports-modules-alsa-karmic-generic (linux-backports-modules-alsa-2.6.31-15-generic). Now the problem is that unpredictably the sound stop working, that is, the apps play but no sound is really played. Looking at dmesg reveals lots of errors (see below). lspci output is also attached. Should I provice more information? Any devel version I can use/test? Regards, bráulio $ dmesg [ 240.404023] ALSA hda_intel.c:699: azx_get_response timeout, switching to polling mode: last cmd=0x015f0900 [ 241.408023] ALSA hda_intel.c:723: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x015f0900 [ 241.408097] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x15f0900 [ 241.408116] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1470740 [ 241.408198] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1470740 [ 241.408281] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x18f0900 [ 241.408303] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x2370105 [ 241.408386] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x23a02b [ 241.408469] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x23902b [ 241.408490] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1539000 [ 241.408573] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x1539000 [ 241.408656] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x33a01f [ 241.408677] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x33901f [ 241.408760] ALSA hda_intel.c:647: spurious response 0x0:0x0, last cmd=0x835014 (continues) $ lspci -v 00:1b.0 Audio device: Intel Corporation System Controller Hub (SCH Poulsbo) HD Audio Controller (rev 07) Subsystem: ASUSTeK Computer Inc. Device 8398 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at f3f38000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

1 0

[alsa-devel] [PATCH 1/2 v2] ASoC: sh: fsi: Add runtime PM support
by Kuninori Morimoto 27 Nov '09

27 Nov '09

This patch add support runtime PM. Driver callbacks for Runtime PM are empty because the device registers are always re-initialized after pm_runtime_get_sync(). The Runtime PM functions replaces the clock framework module stop bit handling in this driver. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o add pm_runtime_disable if failed sound/soc/sh/fsi.c | 40 ++++++++++++++++++++++++++-------------- 1 files changed, 26 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index e1a3d1a..e2f70ac 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -559,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -674,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -872,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -903,14 +901,9 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_iounmap; - } + pm_suspend_ignore_children(&pdev->dev, true); + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -935,6 +928,7 @@ exit_free_irq: free_irq(irq, master); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -947,7 +941,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -957,9 +951,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, -- 1.6.3.3

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[alsa-devel] [PATCH] opti-miro: fix OOPS if hardware is not detected
by Krzysztof Helt 27 Nov '09

27 Nov '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> If a hardware is not detected there is a kernel crash due to not initialized snd_miro->aci pointer. This pointer is initialized after detection of the opti (miro) chip. This bug was introduced by patches to expose ACI mikser outside the snd-miro driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 40b64cd..e374869 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1229,7 +1229,8 @@ static void snd_card_miro_free(struct snd_card *card) struct snd_miro *miro = card->private_data; release_and_free_resource(miro->res_aci_port); - miro->aci->aci_port = 0; + if (miro->aci) + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } -- 1.6.4 ---------------------------------------------------------------------- Wygraj pobyt w Alpach dla ca�ej rodziny Kliknij >>> http://link.interia.pl/f2446

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[alsa-devel] [PATCH 0/5] PCM mmap (temporary) fixes for non-coherent architectures
by Takashi Iwai 27 Nov '09

27 Nov '09

Hi, this is a patchset to fix Oopses from ALSA PCM core with mmap on MIPS and PPC non-coherent architectures. This contains also a clean-up for ARM mmap. In this patch series, I don't try to port dma_mmap_coherent() to these architectures yet, but just put ugly ifdefs in the ALSA core side. Understand that this is a first step forward to a more cleaner solution, and the purpose right now is just to fix up long-standing Oops. I'll try to push dma_mmap_coherent() patches for the next kernel once again after this gets merged. Since the changes are minimal, I'd like to put them in ASAP for 2.6.33. Please review and give feedbacks if any problem is found. Thanks, Takashi

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[alsa-devel] audio pm on omap - resume tik sound
by Ryan 27 Nov '09

27 Nov '09

Hello, When audio resumes from suspend i hear a tik sound on my speakers. Why is this sound produced and is there a fix somewhere for this. Regards, ryan

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