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

[alsa-devel] [PATCH 1/2] usbaudio:fix suspend/resume
by Oliver Neukum 11 Mar '11

11 Mar '11

From 0926f3aa059c42ccf096184654a9fb859c75f2e5 Mon Sep 17 00:00:00 2001 From: Oliver Neukum <oliver(a)neukum.org> Date: Fri, 11 Mar 2011 10:07:52 +0100 Subject: [PATCH 1/2] usbaudio:fix suspend/resume - ESHUTDOWN must be correctly handled - the optional interrupt endpoint's URB must be stopped and restarted Signed-off-by: Oliver Neukum <oneukum(a)suse.de> --- sound/usb/card.c | 12 ++++++++++-- sound/usb/mixer.c | 26 ++++++++++++++++++++++++-- sound/usb/mixer.h | 2 ++ 3 files changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index c0f8270..af76f97 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -586,6 +586,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct snd_usb_audio *chip = usb_get_intfdata(intf); struct list_head *p; struct snd_usb_stream *as; + struct usb_mixer_interface *mixer; if (chip == (void *)-1L) return 0; @@ -596,6 +597,10 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) as = list_entry(p, struct snd_usb_stream, list); snd_pcm_suspend_all(as->pcm); } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_usb_mixer_inactivate(mixer); + } } return 0; @@ -604,6 +609,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); + struct usb_mixer_interface *mixer; if (chip == (void *)-1L) return 0; @@ -611,9 +617,11 @@ static int usb_audio_resume(struct usb_interface *intf) return 0; /* * ALSA leaves material resumption to user space - * we just notify + * we just notify and restart the mixers */ - + list_for_each_entry(mixer, &chip->mixer_list, list) + snd_usb_mixer_activate(mixer); + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 85af605..da65130 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2058,8 +2058,9 @@ static void snd_usb_mixer_interrupt(struct urb *urb) { struct usb_mixer_interface *mixer = urb->context; int len = urb->actual_length; + int ustatus = urb->status; - if (urb->status != 0) + if (ustatus != 0) goto requeue; if (mixer->protocol == UAC_VERSION_1) { @@ -2100,12 +2101,33 @@ static void snd_usb_mixer_interrupt(struct urb *urb) } requeue: - if (urb->status != -ENOENT && urb->status != -ECONNRESET) { + if (ustatus != -ENOENT && ustatus != -ECONNRESET && ustatus != -ESHUTDOWN) { urb->dev = mixer->chip->dev; usb_submit_urb(urb, GFP_ATOMIC); } } +/* stop any bus activity of a mixer */ +void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + + +int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 26c636c..b4a2c81 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -52,5 +52,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); +void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); +int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); #endif /* __USBMIXER_H */ -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Add MAX9850 codec driver
by Christian Glindkamp 11 Mar '11

11 Mar '11

This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Only 16 bit audio was tested while the codec was connected to an AT91SAM9G20 SSC in master mode. Signed-off-by: Christian Glindkamp <christian.glindkamp(a)taskit.de> --- I've all ready sent this patch some time ago, but it hung in the moderation queue. This is a slightly modified version. Unfortunately I do not have the hardware anymore to test suggested changes that alter function. sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 358 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 41 +++++ 4 files changed, 405 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345..51e9844 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -179,6 +180,9 @@ config SND_SOC_DMIC config SND_SOC_MAX98088 tristate +config SND_SOC_MAX9850 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507..f2efd1c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o @@ -102,6 +103,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c new file mode 100644 index 0000000..a8c1f95 --- /dev/null +++ b/sound/soc/codecs/max9850.c @@ -0,0 +1,358 @@ +/* + * max9850.c -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * + * Author: Christian Glindkamp <christian.glindkamp(a)taskit.de> + * + * Initial development of this code was funded by + * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/ + * + * 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/init.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9850.h" + +struct max9850_priv { + unsigned int sysclk; +}; + +/* max9850 register cache */ +static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { + 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +}; + +/* these registers are not used at the moment but provided for the sake of + * completeness */ +static int max9850_volatile_register(unsigned int reg) +{ + switch (reg) { + case MAX9850_STATUSA: + case MAX9850_STATUSB: + return 1; + default: + return 0; + } +} + +static const unsigned int max9850_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), + 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), + 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), + 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), +}; + +static const struct snd_kcontrol_new max9850_controls[] = { +SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), +SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1), +SOC_SINGLE("Mono", MAX9850_GENERAL_PURPOSE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new max9850_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0), +SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0), +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_INPUT("INL"), +SND_SOC_DAPM_INPUT("INR"), +SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0, + &max9850_mixer_controls[0], + ARRAY_SIZE(max9850_mixer_controls)), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", NULL, "DAC"}, + {"Output Mixer", "Line In Switch", "Line Input"}, + + /* outputs */ + {"Headphone Output", NULL, "Output Mixer"}, + {"HPL", NULL, "Headphone Output"}, + {"HPR", NULL, "Headphone Output"}, + {"OUTL", NULL, "Output Mixer"}, + {"OUTR", NULL, "Output Mixer"}, + + /* inputs */ + {"Line Input", NULL, "INL"}, + {"Line Input", NULL, "INR"}, + + /* supplies */ + {"DAC", NULL, "MCLK"}, +}; + +static int max9850_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + u64 lrclk_div; + u8 sf, da; + + /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ + sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1; + lrclk_div = (1 << 22); + lrclk_div *= params_rate(params); + lrclk_div *= sf; + do_div(lrclk_div, max9850->sysclk); + + snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f); + snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff); + + da = snd_soc_read(codec, MAX9850_DIGITAL_AUDIO); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + da |= 0x2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + da |= 0x3; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + + /* calculate mclk -> iclk divider */ + if (freq <= 13000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x0); + else if (freq <= 26000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x4); + else if (freq <= 40000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x8); + else + return -EINVAL; + + max9850->sysclk = freq; + return 0; +} + +static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 da = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da |= MAX9850_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + da |= MAX9850_DLY; + break; + case SND_SOC_DAIFMT_RIGHT_J: + da |= MAX9850_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + da |= MAX9850_BCINV | MAX9850_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + da |= MAX9850_BCINV; + break; + case SND_SOC_DAIFMT_NB_IF: + da |= MAX9850_INV; + break; + default: + return -EINVAL; + } + + /* set da */ + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN, + MAX9850_SHDN); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN, 0); + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max9850_dai_ops = { + .hw_params = max9850_hw_params, + .set_sysclk = max9850_set_dai_sysclk, + .set_fmt = max9850_set_dai_fmt, +}; + +static struct snd_soc_dai_driver max9850_dai = { + .name = "max9850-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9850_RATES, + .formats = MAX9850_FORMATS + }, + .ops = &max9850_dai_ops, +}; + +static int max9850_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* enable zero-detect */ + snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); + /* enable charge pump, disable everything else */ + snd_soc_write(codec, MAX9850_ENABLE, 0x30); + /* enable slew-rate control */ + snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40); + /* set slew-rate 125ms */ + snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); + + snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, + ARRAY_SIZE(max9850_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + snd_soc_add_controls(codec, max9850_controls, + ARRAY_SIZE(max9850_controls)); + + return 0; +} +static int max9850_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9850 = { + .probe = max9850_probe, + .remove = max9850_remove, + .set_bias_level = max9850_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max9850_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max9850_reg, + .volatile_register = max9850_volatile_register, +}; + +static int __devinit max9850_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9850_priv *max9850; + int ret; + + max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL); + if (max9850 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9850); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max9850, &max9850_dai, 1); + if (ret < 0) + kfree(max9850); + return ret; +} + +static __devexit int max9850_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max9850_i2c_id[] = { + { "max9850", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); + +static struct i2c_driver max9850_i2c_driver = { + .driver = { + .name = "max9850-codec", + .owner = THIS_MODULE, + }, + .probe = max9850_i2c_probe, + .remove = __devexit_p(max9850_i2c_remove), + .id_table = max9850_i2c_id, +}; + +static int __init max9850_init(void) +{ + return i2c_add_driver(&max9850_i2c_driver); +} +module_init(max9850_init); + +static void __exit max9850_exit(void) +{ + i2c_del_driver(&max9850_i2c_driver); +} +module_exit(max9850_exit); + +MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp(a)taskit.de>"); +MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h new file mode 100644 index 0000000..5268575 --- /dev/null +++ b/sound/soc/codecs/max9850.h @@ -0,0 +1,41 @@ +/* + * max9850.h -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * Author: Christian Glindkamp <christian.glindkamp(a)taskit.de> + * + * 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. + * + */ + +#ifndef _MAX9850_H +#define _MAX9850_H + +#define MAX9850_STATUSA 0x00 +#define MAX9850_STATUSB 0x01 +#define MAX9850_VOLUME 0x02 +#define MAX9850_GENERAL_PURPOSE 0x03 +#define MAX9850_INTERRUPT 0x04 +#define MAX9850_ENABLE 0x05 +#define MAX9850_CLOCK 0x06 +#define MAX9850_CHARGE_PUMP 0x07 +#define MAX9850_LRCLK_MSB 0x08 +#define MAX9850_LRCLK_LSB 0x09 +#define MAX9850_DIGITAL_AUDIO 0x0a + +#define MAX9850_CACHEREGNUM 11 + +/* MAX9850_ENABLE */ +#define MAX9850_SHDN (1<<7) + +/* MAX9850_DIGITAL_AUDIO */ +#define MAX9850_MASTER (1<<7) +#define MAX9850_INV (1<<6) +#define MAX9850_BCINV (1<<5) +#define MAX9850_DLY (1<<3) +#define MAX9850_RTJ (1<<2) + +#endif -- 1.7.2.3

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[alsa-devel] [PATCH 1/2] ASoC: PXA: z2: Mute internal speaker when headphones are connected
by Vasily Khoruzhick 11 Mar '11

11 Mar '11

Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/pxa/z2.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 3ceaef6..b0b6c8d 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -95,6 +95,11 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, + { + .pin = "Ext Spk", + .mask = SND_JACK_HEADPHONE, + .invert = 1 + }, }; /* Headset jack detection gpios */ -- 1.7.4.1

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Re: [alsa-devel] [Device-drivers-devel] [PATCH] Add driver for Analog Devices ADAU1701 SigmaDSP
by Mark Brown 11 Mar '11

11 Mar '11

On Fri, Mar 11, 2011 at 02:54:42AM -0500, Cai, Cliff wrote: > 1.The default firmware is used when the chip is just powered-up, > Once driver gets the control,it should load the custom firmware to replace > The default one, > 2.we don't need to worry about this case,the driver which needs loading firmware > Has to be compiled as a module,and of cause,we load it after the filesystem has been mounted. Forcing the driver to be build modular isn't great for usability either, though - it's not very general and some systems don't use modules at all for various reasons (eg, simplicity or difficulties in keeping the fs and the kernel partitions in sync). I'm not sure this gels well with the approach you're taking with the driver just passing the firmware through. On the one hand you don't want the driver to know anything about the firmware or join up the firmware handling with the audio subsystem but on the other hand the driver is requiring that a firmware be loaded as soon as the driver is even though it can provide some very basic functionality without it. It'd seem better to just let the user load firmware at their leisure.

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[alsa-devel] [PATCH] ALSA: HDA: Remove over-initialization of mux in ALC269VB
by David Henningsson 11 Mar '11

11 Mar '11

Takashi, would you like to give this one a thought to make sure I haven't missed any case where this verb is needed. For both models (amic and dmic), the verb is re-issued, and the autoparser would either add a mux, or at least set it to a correct start. The stable-cc (2.6.38+) is set that way because we tried it on top of 2.6.35 and it didn't work on the particular machine. I don't know about 2.6.36 or 2.6.37. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH] intel8x0m: append 'm' to "r_intel8x0"
by Paul Bolle 11 Mar '11

11 Mar '11

Appending an 'm' will distinguish it from a similar struct in intel8x0.c Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/pci/intel8x0m.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index e16b3e9..ae31655 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1108,7 +1108,7 @@ struct ich_reg_info { static int __devinit snd_intel8x0m_create(struct snd_card *card, struct pci_dev *pci, unsigned long device_type, - struct intel8x0m ** r_intel8x0) + struct intel8x0m **r_intel8x0m) { struct intel8x0m *chip; int err; @@ -1124,7 +1124,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, }; struct ich_reg_info *tbl; - *r_intel8x0 = NULL; + *r_intel8x0m = NULL; if ((err = pci_enable_device(pci)) < 0) return err; @@ -1237,7 +1237,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); - *r_intel8x0 = chip; + *r_intel8x0m = chip; return 0; } -- 1.7.4.1

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[alsa-devel] Fix for recording issue of ASUS N82JV
by Chih-Wei Huang 11 Mar '11

11 Mar '11

Hi, This small patch fixes the unable to record issue on ASUS N82JV. Thanks. Tested OK on android-x86 2.2 & 2.3 with kernel 2.6.38-rc4. Regards,

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[alsa-devel] [PATCH] Four patches to fixup surround internal speakers on Realtek 88x
by David Henningsson 10 Mar '11

10 Mar '11

I spend last Friday together with Bartłomiej Żogała fixing up a long standing issue with Lenovo Y530, which has 4+1 internal speakers on a Realtek 888. And we all want that supported by the auto parser, don't we? So here comes the patches. Would be nice to have in 2.6.38. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 10 Mar '11

10 Mar '11

The following changes since commit f9c540456ab662a3b5d79c1fa93c51305a97fdf3: ASoC: Remove warnings in ep93xx-i2s.c (2011-03-07 17:32:02 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (4): ASoC: Manage mode and rate bits correctly for CS4271 CODEC. ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC. ASoC: Improve EP93xx I2S clocks management. ASoC: Enable 192kHz sample rate for EP93xx. Dan Carpenter (1): ASoC: sgtl5000: use after free in ldo_regulator_register() Mark Brown (16): ASoC: Add DAPM widget and path data to CODEC driver structure ASoC: Use data based init for WM9081 DAPM ASoC: Fix section mismatch warnings in WM8994 ASoC: Provide CODEC clocking operations and API calls ASoC: Convert WM9081 SYSCLK configuration to be device wide ASoC: Refactor symmetric_rates check to reduce indentation ASoC: Warn rather than set a silly constraint when we can't do symmetry ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew() ASoC: Simplify WM9081 speaker startup by using widgets for spaker output ASoC: Fix broken bitfield definitions in WM8978 ASoC: Use the correct DAPM context when cleaning up final widget set ASoC: Fix double addition of prefixes due to widget prefixing Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Fix typo in late revision WM8994 DAC2R name ASoC: Ensure WM8958 gets all WM8994 late revision widgets Merge branch 'for-2.6.38' into for-2.6.39 include/sound/soc.h | 20 +++++- sound/soc/codecs/cs4271.c | 83 ++++++++++++++-------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/wm8994-tables.c | 2 +- sound/soc/codecs/wm8994.c | 10 +++- sound/soc/codecs/wm8994.h | 2 +- sound/soc/codecs/wm9081.c | 51 ++++--------- sound/soc/ep93xx/edb93xx.c | 16 ++-- sound/soc/ep93xx/ep93xx-i2s.c | 23 +++--- sound/soc/ep93xx/ep93xx-pcm.c | 4 +- sound/soc/soc-core.c | 141 ++++++++++++++++++++++++++++++-------- sound/soc/soc-dapm.c | 45 +++++++++++-- 12 files changed, 274 insertions(+), 127 deletions(-)

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 10 Mar '11

10 Mar '11

The following changes since commit ffd6eae2a0d18ca4a741615292a9c9ce904307fb: ASoC: AM3517: Update codec name after multi-component update (2011-03-09 11:48:55 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Mark Brown (4): ASoC: Fix broken bitfield definitions in WM8978 ASoC: Use the correct DAPM context when cleaning up final widget set ASoC: Fix typo in late revision WM8994 DAC2R name ASoC: Ensure WM8958 gets all WM8994 late revision widgets The DAPM context fix is essential for multi-component DAPM, without it we will try to apply updates from the end of DAPM sequences to the wrong device which isn't terribly clever. On other systems the change will be a noop. The other three updates are device specific ones, the WM8994/58 ones are slightly more invasive but are critical to WM8958. There may be one more update (for mini2440) coming later. sound/soc/codecs/wm8978.c | 14 ++++++++------ sound/soc/codecs/wm8994.c | 10 +++++++++- sound/soc/soc-dapm.c | 2 +- 3 files changed, 18 insertions(+), 8 deletions(-)

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