October 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] sound and sound-unstable git tree rebased
by Takashi Iwai 13 Oct '08

13 Oct '08

Hi, I rebased the whole sound GIT tree right now due to major conflicts with the upstream. Some commits order were changed due to manual pick-ups, but I think I picked all old commits. Please check whether anything missing. Also, the master branch is now pointing to the latest upstream, that is, ahead from 2.6.27. I revived the "devel" branch again for tracking the 2.6.27 tree. If you use 2.6.27 kernel but would like to track ALSA development, use this branch instead of master. Many topic branches vanished now because of the successful merges. There will be still topic/hda and topic/asoc branches, and occasionally more. The sound-unstable GIT tree is rebased as well. This is a usual thing, though... Sorry for inconvenience. Takashi

1 0

[alsa-devel] ALSA: ASoC codec: remove unused #include <version.h>
by Huang Weiyi 13 Oct '08

13 Oct '08

The files below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/ad1980.c sound/soc/codecs/wm8580.c sound/soc/codecs/wm8900.c This patch removes the said #include <version.h>. Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f..1397b8e 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe..a10edda 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d3..3b326c9 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

2 1

[alsa-devel] [PATCH] ASoC: update email address for Liam Girdwood
by Liam Girdwood 13 Oct '08

13 Oct '08

Update the contact information for Liam Girdwood in ASoC core and drivers as my old email address is no longer valid. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/oss/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 2 +- sound/soc/at91/at91-ssc.c | 2 +- sound/soc/at91/eti_b1_wm8731.c | 2 +- sound/soc/codecs/ac97.c | 3 +-- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8753.c | 3 +-- sound/soc/codecs/wm8753.h | 3 +-- sound/soc/codecs/wm9712.c | 3 +-- sound/soc/codecs/wm9713.c | 3 +-- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/soc-core.c | 5 ++--- sound/soc/soc-dapm.c | 5 ++--- 18 files changed, 21 insertions(+), 28 deletions(-) diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e..456a1b4 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * May 02, 2003 Liam Girdwood <lrg(a)slimlogic.co.uk> * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe..6e831af 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * May 2, 2003 Liam Girdwood <lrg(a)slimlogic.co.uk> * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index a5b1a79..1b61cc4 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -5,7 +5,7 @@ * Endrelia Technologies Inc. * * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 684781e..a22f349 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -10,7 +10,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96c..bd1ebdc 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d..d596f70 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44..be817b0 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde0..f96c04b 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b..ffb471e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d058..aba402b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1a8373d..fe95384 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b..e6ff692 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8..33851cd 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 2fb5829..e758034 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * lrg(a)slimlogic.co.uk * * 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 @@ -405,6 +405,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 9a70b00..8375bb5 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa75..1300cb5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + * Authors: 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 diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..462e635 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * with code, comments and ideas from :- * Richard Purdie <richard(a)openedhand.com> * @@ -1886,7 +1885,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 83fa9c4..efbd0b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * * 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 @@ -1541,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg(a)slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); -- 1.5.4.3

2 1

[alsa-devel] [PATCH] hda: corrected invalid mixer values
by Matthew Ranostay 13 Oct '08

13 Oct '08

Corrected invalid mixer index values on the 92hd71bxxx codec branch. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 1e7b6c1..c590655 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -1114,11 +1114,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),

2 1

[alsa-devel] [PATCH] hda: add mixers for analog mixer on 92hd75xx codecs
by Matthew Ranostay 11 Oct '08

11 Oct '08

Add support for mixers on the analog mixer on some 92hd75xx codecs, along with adding a 'Mixer' entry for it's connection on the dmux. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..1f18d32 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -320,8 +320,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[1] = { - 0x1c, +static hda_nid_t stac92hd71bxx_dmux_nids[2] = { + 0x1c, 0x1d, }; static hda_nid_t stac92hd71bxx_smux_nids[2] = { @@ -837,20 +837,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; -#define HD_DISABLE_PORTF 3 +#define HD_DISABLE_PORTF 2 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for node 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ @@ -859,10 +857,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect port 0d to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute dac0 input in audio mixer */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1083,6 +1077,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1095,8 +1090,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), */ - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -1616,7 +1620,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, + 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, 0x90a000f0, 0x01452050, 0x01452050, }; @@ -2961,7 +2965,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) /* labels for amp mux outputs */ static const char *stac92xx_amp_labels[3] = { - "Front Microphone", "Microphone", "Line In" + "Front Microphone", "Microphone", "Line In", }; /* create amp out controls mux on capable codecs */ @@ -4275,6 +4279,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = { #endif }; +static struct hda_input_mux stac92hd71bxx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x00 }, + { "Mixer", 0x01 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4289,6 +4303,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, + sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -4329,6 +4345,7 @@ again: /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; + spec->dinput_mux = &spec->private_dimux; /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; @@ -4346,12 +4363,13 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } - spec->aloopback_mask = 0x20; + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; /* GPIO0 High = EAPD */ @@ -4370,7 +4388,6 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); switch (spec->board_config) { case STAC_HP_M4: @@ -4391,6 +4408,10 @@ again: spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + if (spec->dinput_mux) + spec->private_dimux.num_items += + spec->num_dmics - + (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) {

2 2

[alsa-devel] ASoC updates
by Mark Brown 11 Oct '08

11 Oct '08

The following changes since commit f79f27b5e79b9ae1e39d8376fd2ed956e3e02430: Takashi Iwai (1): ALSA: ASoC - clean up Kconfig for TLV320AIC2 are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Jarkko Nikula (3): ASoC: OMAP: Add multilink support to McBSP DAI driver ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver Mark Brown (1): ASoC: Make TLV320AIC26 user-visible sound/soc/codecs/Kconfig | 2 +- sound/soc/omap/omap-mcbsp.c | 176 ++++++++++++++++++++++++++++++------------- sound/soc/omap/omap-mcbsp.h | 16 +++- 3 files changed, 134 insertions(+), 60 deletions(-)

2 5

[alsa-devel] Acer Aspire 6920 (hda ALC889): no headphone switching or capture
by Chase Cathcart 11 Oct '08

11 Oct '08

Hello, I have an Acer Aspire 6920 (not 6920g; ALC889/ICH8) on which I got sound output to work a while back (by upgrading to custom 2.6.26 kernel), with a couple problems: - When speakers/headphones are plugged into headphone jack, sound comes out both headphone jack and laptop speakers simultaneously. I have used hda-verb to see if I could switch the laptop speakers off: /usr/bin/hda-verb /dev/snd/hwC0D0 0x15 SET_PIN_WIDGET_CONTROL 0x00 (turns off headphones) /usr/bin/hda-verb /dev/snd/hwC0D0 0x15 SET_PIN_WIDGET_CONTROL 0x40 (turns on headphones/laptop speakers stay on) /usr/bin/hda-verb /dev/snd/hwC0D0 0x15 SET_PIN_WIDGET_CONTROL 0xc0 (supposedly some 'headphone mode', just seems to make the jack louder, laptop speakers still on) /usr/bin/hda-verb /dev/snd/hwC0D0 0x0c SET_AMP_GAIN_MUTE 0x7080 (mutes all) /usr/bin/hda-verb /dev/snd/hwC0D0 0x0c SET_AMP_GAIN_MUTE 0x7000 (unmutes all) The headphone switch in mixer does nothing. The Front switch in alsamixer mutes/unmutes all. Looking at patch_realtek.c, it seems all the alc883_acer_aspire_* code assumes that headphones are node 0x14 and laptop speakers are 0x15/0x16, which doesn't seem like the case here because everything seems to be at the whim of 0x15. Is there a verb I'm missing, should I try a different model parameter, or some other way of switching off the laptop speakers that isn't implemented by the driver? - I have no working capture devices. None of the built-in mic, mic jack, or line-in jack work. Since the amp-ins on 0x0b-0x0f all appear to be muted, I wonder if that has something to do with it, but I can't seem to find a verb to unmute them. I have hda-verb installed, and got the ALC889 Datasheet: ftp://202.65.194.211/pc/audio/ALC889_DataSheet_1.0.pdf , but I don't quite understand how to pass the verb parameters listed there on the command line. Hopefully the information I was able to provide isn't unhelpful, and if you need more information, please request. Thanks, Chase C. Here's my vitals (alsa-info.sh): !!################################ !!ALSA Information Script v 0.4.48 !!################################ !!Script ran on: Fri Oct 10 15:49:53 EDT 2008 !!Linux Distribution !!------------------ Ubuntu 8.04.1 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04.1" !!Kernel Information !!------------------ Kernel release: 2.6.27-200810-200810 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.17 Library version: 1.0.15 Utilities version: 1.0.15 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 22 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1025:0146 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-hda-intel: model=acer-aspire !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : acer-aspire,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889 Address: 0 Vendor Id: 0x10ec0889 Subsystem Id: 0x10250146 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x90 0x90] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x570]: 32000 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x03 0x03] [0x80 0x80] [0x9d 0x9d] [0x80 0x80] [0x80 0x80] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2d 0x2d] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5f0]: 32000 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x18561130: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Black DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99a3094e: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x4, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081003c: IN OUT HP EAPD Detect EAPD 0x0: Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=04, enabled=1 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x03 0x03] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a19c40: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02813441: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue DefAssociation = 0x4, Sequence = 0x1 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f 0x26* Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x08373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple DefAssociation = 0x2, Sequence = 0xd Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400280: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=28 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 12 0x18* 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 0x12 Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x3e, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x3e 0x3e] Connection: 2 0x25 0x0b Codec: LSI ID 1040 Address: 1 Vendor Id: 0x11c11040 Subsystem Id: 0x10250146 Revision Id: 0x100200 Modem Function Group: 0x1 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 2008-10-10 15:42 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 2008-10-10 15:42 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 2008-10-10 15:42 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 2008-10-10 15:42 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 2008-10-10 15:42 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 17 2008-10-10 15:42 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 26 2008-10-10 15:42 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 1 2008-10-10 15:42 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 2008-10-10 15:42 /dev/snd/timer !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) pcm.hda-intel { type hw card 0 } ctl.hda-intel { type hw card 0 } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC889 Digital [ALC889 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC889 Analog [ALC889 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 45 [70%] [-19.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [off] Front Right: Playback [off] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [2.00dB] [on] Front Right: Playback 64 [100%] [2.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 29 [94%] [9.00dB] [off] Front Right: Playback 29 [94%] [9.00dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 3 [10%] [-30.00dB] [on] Front Right: Playback 3 [10%] [-30.00dB] [on] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 3 [100%] Front Right: 3 [100%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Front Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Front Mic' !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Front Playback Volume' value.0 64 value.1 64 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 false value.1 false } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Line Playback Volume' value.0 29 value.1 29 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 false value.1 false } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Mic Playback Volume' value.0 3 value.1 3 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' iface MIXER name 'Mic Boost' value.0 3 value.1 3 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 true value.1 true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.13 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' value 'Front Mic' } control.14 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' comment.item.2 Line comment.item.3 CD iface MIXER name 'Input Source' index 1 value 'Front Mic' } control.15 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.16 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.17 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.19 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value false } control.20 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' iface MIXER name 'Master Playback Volume' value 45 } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.22 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } control.23 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 } } --endcollapse-- !!All Loaded Modules !!------------------ Module ipv6 af_packet rfcomm l2cap bluetooth i915 drm binfmt_misc acpi_cpufreq cpufreq_ondemand cpufreq_performance freq_table cpufreq_userspace cpufreq_conservative sbs sbshc arc4 ecb crypto_blkcipher cryptomgr snd_hda_intel snd_pcm_oss snd_mixer_oss snd_pcm uvcvideo snd_page_alloc snd_hwdep iwl3945 compat_ioctl32 snd_seq_oss rfkill snd_seq_midi snd_rawmidi videodev serio_raw firmware_class v4l1_compat psmouse mac80211 snd_seq_midi_event snd_seq cfg80211 snd_timer snd_seq_device atl1e snd soundcore battery video container ac button acer_wmi led_class mousedev wmi evdev usbhid hid sg sr_mod cdrom sd_mod pata_acpi ehci_hcd uhci_hcd usbcore thermal processor fan thermal_sys hwmon fuse ata_piix ata_generic ahci libata scsi_mod dock

1 0

[alsa-devel] Fwd: Backported sbxfi driver (UNTESTED!)
by The Source 10 Oct '08

10 Oct '08

---------- Forwarded message ---------- From: The Source <thesourcehim(a)gmail.com> Date: 2008/10/10 Subject: Re: [alsa-devel] Backported sbxfi driver (UNTESTED!) To: Takashi Iwai <tiwai(a)suse.de> 2008/10/10 Takashi Iwai <tiwai(a)suse.de> At Fri, 10 Oct 2008 01:26:15 -0500, > Ted T. Logian wrote: > > > > > > > > On Fri, 2008-10-10 at 08:01 +0200, Takashi Iwai wrote: > > > At Thu, 09 Oct 2008 15:17:58 -0500, > > > Ted T. Logian wrote: > > > > > > > > Sorry for the multiple posts, but once I rmmod snd-sbxfi, running > mixer does > > > > fine. It seems if I have snd-sbxfi loaded and I run alsamixer, it > locks the > > > > entire system. > > > > > > So, do you mean loading the driver itself doesn't lock up? If so, > > > it's better than I expected. > > > > > > Did you run ever PCM playback/capture before that? This is more > > > dangerous :) > > > > > > Also, please give your hardware details, as there are different models > > > for X-Fi, and some are handled pretty differently. > > > > > > [BTW, please stop top-posting, and avoid HTML mails for ML. It's easy > > > to switch to normal (plain) mail mode on Gmail, just a click.] > > > > > > > > > thanks, > > > > > > Takashi > > > > > > I think perhaps from uninstalling pulseaudio I got further. I can use > > mixer now, and it even has a "Master" device, but nothing else... > > > > however, I do get a lock up later. > > > > I get this from running aplay, too, if this helps... > > Thanks! That's rather a stupid bug. > The patch is below. > > > Takashi > > diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c > index 8066bf4..8a0eea0 100644 > --- a/sound/pci/sbxfi/sbxfi.c > +++ b/sound/pci/sbxfi/sbxfi.c > @@ -831,7 +831,7 @@ static struct sbxfi_port *sbxfi_port_alloc(struct sbxfi > *chip, > port->src[0] = src; > port->src[1] = src + 1; > spin_lock_irq(&chip->port_lock); > - list_add(&chip->port_list, &port->list); > + list_add(&port->list, &chip->port_list); > spin_unlock_irq(&chip->port_lock); > return port; > } > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > Hello, I have tried the driver with the fix you provided, but I couldn't get anything. Kernel panic on any attempt to use sound or mixer. Also kernel panic several seconds after Xorg starts. Each crash causes either hang or reboot so I couldn't get any error messages. If there are some logs left, please tell me where they are and I'll send them to you. My card is X-Fi Platinum Fatality Champion Series.

1 0

[alsa-devel] [topic/asoc][DON'T PUSH][PATCH 0/3] ASoC: OMAP: McBSP DAI driver updates
by Jarkko Nikula 10 Oct '08

10 Oct '08

Here are updates to OMAP McBSP DAI driver adding support to OMAP2430 & OMAP34xx and adding support for multiple simultaneous DAI links. Please note that the first patch is not randconfig safe until necessary OMAP McBSP updates come into mainline so I rather don't push these yet until those changes are here. See http://marc.info/?l=linux-omap&m=122345107722469&w=2 -- Jarkko

3 8

[alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by Sebastian Schneider 09 Oct '08

09 Oct '08

Ok Takashi, just send me the code and give me any instructions I have to make to compile the card. I have one of the first X-FI Fatality cards for PCI, so I should be able to test it. -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

1 0

[alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by Ted T. Logian 09 Oct '08

09 Oct '08

I had to ask because oss4 has had emu20k1/x-fi support for a long time now. However, it has the obvious limitations of oss4 and also you cannot use usb microphones, so I can't use skype which I'd really like to do. Any ideas?

7 16

[alsa-devel] [PATCH] hda: VREF powerdown for headphones.
by Matthew Ranostay 09 Oct '08

09 Oct '08

Add support for powering down VREF on standard headphone insertion, also powers up the VREF on a headset insertion. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..2e866e0 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -38,6 +38,7 @@ #define NUM_CONTROL_ALLOC 32 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 +#define STAC_VREF_EVENT 0x40 enum { STAC_REF, @@ -3811,13 +3812,22 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) struct sigmatel_spec *spec = codec->spec; int idx = res >> 26 & 0x0f; - switch ((res >> 26) & 0x30) { + switch ((res >> 26) & 0x70) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ case STAC_PWR_EVENT: if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, idx); + break; + case STAC_VREF_EVENT: { + int data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + /* toggle VREF state based on GPIOx status */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, + !!(data & (1 << idx))); + break; + } } } @@ -4318,6 +4328,17 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ + switch (codec->subsystem_id) { + case 0x103c361a: + /* Enable VREF power saving on GPIO1 detect */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + spec->gpio_mask |= 0x02; + break; + } if ((codec->revision_id & 0xf) == 0 || (codec->revision_id & 0xf) == 1) { #ifdef SND_HDA_NEEDS_RESUME

2 1

[alsa-devel] [PATCH] hda: STAC_HP_M4
by Matthew Ranostay 09 Oct '08

09 Oct '08

Set the third microphone to a default config of a 'Line In' to be added the Input Source Mux's. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index b5c1449..5557116 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3605,7 +3605,12 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { - unsigned int pinctl = AC_PINCTL_IN_EN; + unsigned int pinctl = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + /* if PINCTL already set then skip */ + if (pinctl & AC_PINCAP_IN) + continue; + pinctl = AC_PINCTL_IN_EN; if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); @@ -4375,12 +4380,13 @@ again: switch (spec->board_config) { case STAC_HP_M4: spec->num_dmics = 0; - spec->num_smuxes = 1; + spec->num_smuxes = 0; spec->num_dmuxes = 0; /* enable internal microphone */ - snd_hda_codec_write_cache(codec, 0x0e, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + stac92xx_set_config_reg(codec, 0x0e, 0x01813040); + stac92xx_auto_set_pinctl(codec, 0x0e, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); break; default: spec->num_dmics = STAC92HD71BXX_NUM_DMICS;

2 1

[PATCH] ARM: OMAP: Hack add some OMAP3 McBSP muxing
by Jarkko Nikula 09 Oct '08

09 Oct '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- arch/arm/mach-omap2/mux.c | 52 +++++++++++++++++++++++++++++++++ arch/arm/plat-omap/include/mach/mux.h | 29 ++++++++++++++++++ 2 files changed, 81 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-omap2/mux.c b/arch/arm/mach-omap2/mux.c index dacb41f..1dff09e 100644 --- a/arch/arm/mach-omap2/mux.c +++ b/arch/arm/mach-omap2/mux.c @@ -261,6 +261,58 @@ static struct pin_config __initdata_or_module omap34xx_pins[] = { * mux-mode | [active-mode | off-mode] */ +/* Primary McBSP multiplexing */ +MUX_CFG_34XX("Y21_3430_MCBSP1_CLKR", 0x18c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AA21_3430_MCBSP1_FSR", 0x18e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("Y21_3430_MCBSP1_DX", 0x190, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("U21_3430_MCBSP1_DR", 0x192, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("T21_3430_MCBSP1_CLKS", 0x194, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("K26_3430_MCBSP1_FSX", 0x196, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("W21_3430_MCBSP1_CLKX", 0x198, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("P21_3430_MCBSP2_FSX", 0x13c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("N21_3430_MCBSP2_CLKX", 0x13e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("R21_3430_MCBSP2_DR", 0x140, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("M21_3430_MCBSP2_DX", 0x142, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AF6_3430_MCBSP3_DX", 0x16c, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AE6_3430_MCBSP3_DR", 0x16e, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AF5_3430_MCBSP3_CLKX", 0x170, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AE5_3430_MCBSP3_FSX", 0x172, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AE1_3430_MCBSP4_CLKX", 0x184, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AD1_3430_MCBSP4_DR", 0x186, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AD2_3430_MCBSP4_DX", 0x188, + OMAP34XX_MUX_MODE0) +MUX_CFG_34XX("AC1_3430_MCBSP4_FSX", 0x18a, + OMAP34XX_MUX_MODE0) + +MUX_CFG_34XX("AF10_3430_MCBSP5_CLKX", 0x5d8, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AE11_3430_MCBSP5_DR", 0x5e4, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AH9_3430_MCBSP5_FSX", 0x5e6, + OMAP34XX_MUX_MODE1) +MUX_CFG_34XX("AF13_3430_MCBSP5_DX", 0x5e8, + OMAP34XX_MUX_MODE1) + /* 34xx I2C */ MUX_CFG_34XX("K21_34XX_I2C1_SCL", 0x1ba, OMAP34XX_MUX_MODE0 | OMAP34XX_PIN_INPUT_PULLUP) diff --git a/arch/arm/plat-omap/include/mach/mux.h b/arch/arm/plat-omap/include/mach/mux.h index f4362b8..c0cfa73 100644 --- a/arch/arm/plat-omap/include/mach/mux.h +++ b/arch/arm/plat-omap/include/mach/mux.h @@ -677,6 +677,35 @@ enum omap24xx_index { }; enum omap34xx_index { + /* Primary McBSP multiplexing */ + Y21_3430_MCBSP1_CLKR, + AA21_3430_MCBSP1_FSR, + Y21_3430_MCBSP1_DX, + U21_3430_MCBSP1_DR, + T21_3430_MCBSP1_CLKS, + K26_3430_MCBSP1_FSX, + W21_3430_MCBSP1_CLKX, + + P21_3430_MCBSP2_FSX, + N21_3430_MCBSP2_CLKX, + R21_3430_MCBSP2_DR, + M21_3430_MCBSP2_DX, + + AF6_3430_MCBSP3_DX, + AE6_3430_MCBSP3_DR, + AF5_3430_MCBSP3_CLKX, + AE5_3430_MCBSP3_FSX, + + AE1_3430_MCBSP4_CLKX, + AD1_3430_MCBSP4_DR, + AD2_3430_MCBSP4_DX, + AC1_3430_MCBSP4_FSX, + + AF10_3430_MCBSP5_CLKX, + AE11_3430_MCBSP5_DR, + AH9_3430_MCBSP5_FSX, + AF13_3430_MCBSP5_DX, + /* 34xx I2C */ K21_34XX_I2C1_SCL, J21_34XX_I2C1_SDA, -- 1.5.6.5 --Multipart=_Thu__2_Apr_2009_19_56_53_+0300_OCGZiXCSMdXq4.P8--

1 0

[alsa-devel] programming AC'97
by joan vicient 09 Oct '08

09 Oct '08

Hi, I'm an electronic engineering student and I'm having some troubles with my project wich I have to develop to become an engineer. The project itself is based on my laptop (ICH6 with the AC97 soundcard). I'm trying to develop a sound generator with the AC97. I've been working hard but there is something wrong and I'm not sure what is it. I think is the "Buffer Descriptor List Base Addres" or maybe the codec initialization, not sure... I am looking at the "AC'97 program model" and alsa-driver to understand how does the driver configure the soundcard so I can do the same but I don't understand the source code at all... So... somebody out there with some skills can help me? Thank you very much!!

1 0

[alsa-devel] [PATCH 0/9] ASoC Blackfin supporting
by Bryan Wu 09 Oct '08

09 Oct '08

Hi Mark, After your last review, Cliff and me updated this patch series: - fix coding style issue - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

6 24

Re: [alsa-devel] Will there ever be emu20k1/x-fi support in Alsa?
by The Source 09 Oct '08

09 Oct '08

So, should anyone start to port OSS driver or should we wait for specs?

1 0

[alsa-devel] dv6627el & audio problems
by yelo_3 09 Oct '08

09 Oct '08

Hello, I want to share my problems with you. I'm running ubuntu intrepid, with these packages versions: linux 2.6.27-6-generic alsa 1.0.17.dfsg-2ubuntu1 I have an hp dv6627el with this audio card model: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) and this is my alsa page: http://www.alsa-project.org/db/?f=457f3992bcf0b34b51772a730071ce18deea75b2 I don't use any additional module option other that the one I will specify. I have 2 problems: 1) mute led does not work. if I use the option power_save (for isntance=3) the mute led lights up (red) after 3 seconds of audio inactivity (and after having closed the audio software), even if the volume is not muted. If I keep the audio software open, the led remains "off" (blue). if instead I don't use the option power_save (so power_save=0) the led never lights up. 2) when using headphones, the speakers are normally muted, but if I hit the mute button twice (or use the volume control applet to mute and unmute), the speakers are activated. Thank you very much for you help. Nicolò

1 0

[alsa-devel] [PATCH] [ALSA] Allow setting codec register with debugfs filesystem
by Troy Kisky 08 Oct '08

08 Oct '08

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg will set register 0x06 to a value of 0x59 Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> Mark, I didn't base it on dapm's pop_time directory because that hasn't made it to my testing kernel yet. I cannot test directly from the alsa git tree. Thanks for the feedback. diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f616b0e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -516,6 +516,9 @@ struct snd_soc_device { struct delayed_work delayed_work; struct work_struct deferred_resume_work; void *codec_data; +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_root; +#endif }; /* runtime channel data */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad38113..156d11a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -962,10 +962,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +1000,112 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_DEBUG_FS +static int code_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t code_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(devdata, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t code_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = code_reg_open_file, + .read = code_reg_read_file, + .write = code_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *debugfs_codec_reg; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + debugfs_codec_reg = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!debugfs_codec_reg) + goto exit2; + + socdev->debugfs_root = root; + return; +exit2: + debugfs_remove(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove_recursive(socdev->debugfs_root); + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1217,6 +1319,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_codec_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1240,6 +1343,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #endif mutex_lock(&codec->mutex); + soc_cleanup_codec_debugfs(socdev); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; -- 1.5.4.3

2 1

[alsa-devel] [PATCH] ASoC: Check for machine type in GTA01 machine driver
by Mark Brown 08 Oct '08

08 Oct '08

Since there are now multiple OpenMoko platforms it is more important to check that the machine driver is running on the correct system. This was orgininally generated as part of the initial GTA02 machine port. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/neo1973_wm8753.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c602674..87ddfef 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -24,6 +24,7 @@ #include <sound/soc-dapm.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> @@ -677,6 +678,12 @@ static int __init neo1973_init(void) DBG("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; -- 1.5.6.5

2 1

Re: [alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 08 Oct '08

08 Oct '08

Not yet. Would please share what you have done to make it work ? Also when I tried to strace aplay , it keeps looking for a </share/alsa/cards/tlv320aic23.conf> file which is not there in the alsa-lib package. Any idea what goes in that file ? Regards Jesslyn On Tue, Oct 7, 2008 at 7:16 PM, Arun KS <arunks(a)mistralsolutions.com> wrote: > Jesslyn Abdul Salam wrote: > >> I got it working. I2C_BOARD_INFO macro was wrongly intialized. Instead of >> tlv320aic23 i had written tlvaic23. Thanks to Arun for pointing this error >> >> > Can you hear audio playing? > Because i had done some hack to make it working in 5912osk. > I want to know you are not facing it or not. > > Arun > > Jesslyn Abdul Salam >> >> >> On Tue, Oct 7, 2008 at 5:36 PM, Jesslyn Abdul Salam < >> jesslyn.abdulsalam(a)gmail.com> wrote: >> >> >> >>> ---------- Forwarded message ---------- >>> From: Jesslyn Abdul Salam <jesslyn.abdulsalam(a)gmail.com> >>> Date: Tue, Oct 7, 2008 at 5:36 PM >>> Subject: Re: [alsa-devel] AIC23 SOC Driver on OSK5912 >>> To: Arun KS <arunks(a)mistralsolutions.com> >>> >>> >>> Yes, it is there. but it shows as "tlv320aic23b" instead of "tlv320aic" >>> Will that be the reason ???? >>> >>> >>> >>> >>> On Tue, Oct 7, 2008 at 5:32 PM, Arun KS <arunks(a)mistralsolutions.com >>> >wrote: >>> >>> >>> >>>> On Tue, Oct 7, 2008 at 5:24 PM, Mark Brown <broonie(a)sirena.org.uk> >>>> wrote: >>>> >>>> >>>>> On Tue, Oct 07, 2008 at 05:18:55PM +0530, Jesslyn Abdul Salam wrote: >>>>> >>>>> >>>>> >>>>>> I have been trying to use the AIC23 ALSA SoC driver for OSK5912 for >>>>>> the >>>>>> >>>>>> >>>>> past >>>> >>>> >>>>> few days. I got the driver patch from the linux-omap/alsa-devel mailing >>>>>> lists. The driver got compiled along just fine. But when the kernel >>>>>> >>>>>> >>>>> boots, >>>> >>>> >>>>> no sound cards have been listed. >>>>>> >>>>>> >>>>> CCing in Arun KS <arunks(a)mistralsolutions.com> who contributed that >>>>> support. >>>>> >>>>> >>>>> >>>>>> Advanced Linux Sound Architecture Driver Version 1.0.17. >>>>>> ASoC version >>>>>> 0.13.2 >>>>>> AIC23 Audio Codec >>>>>> 0.1 >>>>>> MCLK = 12000000 [12000000], usecount = >>>>>> 0 >>>>>> ALSA device >>>>>> list: >>>>>> No soundcards >>>>>> found. >>>>>> Any one have tested this before ? >>>>>> >>>>>> >>>>> Can you check i2c? >>>> >>>> # cat /sys/bus/i2c/devices/1-001b/name >>>> tlv320aic23 >>>> >>>> >>>> >>>>> _______________________________________________ >>>>> Alsa-devel mailing list >>>>> Alsa-devel(a)alsa-project.org >>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>>> >>>>> >>>>> >>>> >>> >>> >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >> >> >> > > > Please do not print this email unless it is absolutely necessary. Spread > environmental awareness. > > > -------------------------------------------------------DISCLAIMER------------------------------------------------------ > The information transmitted herewith is confidential and proprietary > information intended only for use by the individual or entity to which it is > addressed. If the reader of this message is not the intended recipient, you > are hereby notified that any review, retransmission, dissemination, > distribution, copying or other use of, or taking of any action in reliance > upon this information is strictly prohibited. If you have received this > communication in error, please contact the sender and delete the material > from your computer. > > -------------------------------------------------------------------------------------------------------------------------------- > > > >

3 3

[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 08 Oct '08

08 Oct '08

The following changes since commit 691ac84a77b1aebb7e09b4924b4b9c07a0f4f66a: Takashi Iwai (1): Merge branch 'asoc-fixes' into topic/asoc are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai This is all 2.6.28ish material. Arun KS (1): ASoC: Add custom SOC_SINGLE_TLV for tlv320aic23 codec Jarkko Nikula (2): ASoC: tlv320aic3x: Use uniform tlv320aic naming ASoC: Remove unused AUDIO_NAME define from codec drivers Mark Brown (4): ASoC: Add WM8753 SPI support ASoC: Add WM8510 SPI support ASoC: Implement WM8510 bias level control ASoC: Make WM8510 microphone input a DAPM mixer Richard Zhao (1): ALSA: ASoC: add new param mux to dapm_mux_update_power sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ssm2602.c | 1 - sound/soc/codecs/tlv320aic23.c | 54 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.c | 5 +- sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wm8510.c | 109 ++++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm8510.h | 1 + sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 72 +++++++++++++++++++++++++- sound/soc/codecs/wm8753.h | 1 + sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/soc-dapm.c | 8 ++-- 15 files changed, 221 insertions(+), 37 deletions(-)

2 9

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: dynamic detection of MIDI interfaces in uaxx-quirk
by pedro.lopez.cabanillas＠gmail.com 08 Oct '08

08 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> The MIDI interfaces have to be detected dynamically for Edirol devices ua-700, ua-25 and ua4-fx. This patch reverses the wrong changes made by my other patch in uaxx-quirk. Signed-off-by: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6e70ba4..bbd70d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3023,6 +3023,31 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, alts = &iface->altsetting[1]; altsd = get_iface_desc(alts); + if (altsd->bNumEndpoints == 2) { + static const struct snd_usb_midi_endpoint_info ua700_ep = { + .out_cables = 0x0003, + .in_cables = 0x0003 + }; + static const struct snd_usb_audio_quirk ua700_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &ua700_ep + }; + static const struct snd_usb_midi_endpoint_info uaxx_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk uaxx_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &uaxx_ep + }; + if (chip->usb_id == USB_ID(0x0582, 0x002b)) + return snd_usb_create_midi_interface(chip, iface, + &ua700_quirk); + else + return snd_usb_create_midi_interface(chip, iface, + &uaxx_quirk); + } + if (altsd->bNumEndpoints != 1) return -ENXIO; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 85532cf..69689e7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -870,11 +870,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1216,11 +1212,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 @@ -1375,11 +1367,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = -1 -- 1.5.2.4

2 1

[alsa-devel] ASoC WM8510 Mic PGA problem
by ext-jukka.hynninen＠vaisala.com 08 Oct '08

08 Oct '08

Hi all, I'm facing trouble with programmable mic gain stage and WM8510 ASoC driver. First of all, the DAPM control "Mic PGA Switch" seems to be operating just the opposite way the codec datasheet suggests. I inverted the control with expected result, PGA now almost works. When I start capturing, the "Mic PGA" control is instantly turned off again. If I manually enable the control while the capture is running, it works just fine. Any ideas? Am I missing some DAPM routings? -Jukka

3 11

Re: [alsa-devel] [PATCH] Sigmatel stac92hd71b7 low HP volume
by Takashi Iwai 08 Oct '08

08 Oct '08

[please don't drop Cc to alsa-devel ML] At Mon, 6 Oct 2008 12:48:43 +0200, Klaas De Craemer wrote: > > Hello, > > I'm sure that the audioquality will degrade somewhat because of the additional amplification, > but before, the HP-out was not usable at all. I had to use an external ampli just to use my > headphones, even after turning all volume sliders to 100%. Then again, when removing the HP, > the volume though the internal speakers was way too loud... > > I'm not sure what you mean with the output pin and headphone control, but I have uploaded the > output of the codegraph tool before and after my modification here: > http://users.vtk.be/~s0159120/codec_original.svg > http://users.vtk.be/~s0159120/codec_final.svg > > Below is the output of alsa-info with the patched module > Klaas Thanks. So, if I understand correctly, when the pin 0x0a is connected directly from DAC0 0x10, the output level is too low even if you set the all volumes maximal. In the alsa-info.sh output, I see the following: > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] which means that you decrease the output level intentionally. Is the HP output still too low even if you set to the higher value? Takashi > > ----- > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.51 > !!################################ > > !!Script ran on: Mon Oct 6 12:42:48 CEST 2008 > > > !!Linux Distribution > !!------------------ > > Debian GNU/Linux lenny/sid \n \l > > > !!Kernel Information > !!------------------ > > Kernel release: 2.6.26-5.slh.6-sidux-686 > Operating System: GNU/Linux > Architecture: i686 > Processor: unknown > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.16 > Library version: > Utilities version: 1.0.16 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xf6fdc000 irq 21 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) > > > !!Advanced information - PCI Vendor/Device/Susbsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:293e (rev 03) > Subsystem: 1028:024f > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd-atiixp-modem: index=-2 > snd-intel8x0m: index=-2 > snd-via82xx-modem: index=-2 > snd-pcsp: index=-2 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : 0 > id : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > model : > <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> > position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > single_cmd : N > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: IDT 92HD71B7X > Address: 0 > Vendor Id: 0x111d76b2 > Subsystem Id: 0x1028024f > Revision Id: 0x100302 > 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=0x00, stepsize=0x00, mute=1 > Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 > GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 > IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 > IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 > Node 0x0a [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x081c: OUT HP Detect > Pin Default 0x0421101f: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=30, enabled=1 > Connection: 3 > 0x10 0x11 0x17* > Node 0x0b [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x04a11221: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0c [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x0e [Pin Complex] wcaps 0x400081: Stereo > Pincap 0x081724: IN Detect > Vref caps: HIZ 50 GRD 80 > Pin Default 0x23a1902e: [Jack] Mic at Sep Left > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0xe > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=00, enabled=0 > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Pincap 0x0814: OUT Detect > Pin Default 0x23014250: [Jack] Line Out at Sep Left > Conn = 1/8, Color = Green > DefAssociation = 0x5, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Connection: 3 > 0x10* 0x11 0x17 > Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x5e 0x5e] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L > Amp-Out caps: N/A > Amp-Out vals: [0x3d 0x3d] > Converter: stream=5, channel=0 > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x12 [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 0x13 [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 0x14 [Pin Complex] wcaps 0x400100: Mono > Pincap 0x0810: OUT > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x16 > Node 0x15 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x10* 0x11 0x17 > Node 0x16 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x15 > Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x1f 0x1f] [0x1f 0x1f] [0x97 0x97] [0x97 0x97] [0x97 0x97] > Connection: 5 > 0x10 0x11 0x27 0x1a 0x1b > Node 0x18 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x90a000f0: [Fixed] Mic at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x19 [Pin Complex] wcaps 0x40000d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x20: IN > Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x01 0x01] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 3 > 0x0b* 0x0c 0x0e > Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x00 0x00] > Connection: 4 > 0x1a* 0x17 0x18 0x19 > Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x80 0x80] > Connection: 4 > 0x1b* 0x17 0x18 0x19 > Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Connection: 1 > 0x24 > Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital > Pincap 0x0810010: OUT EAPD > EAPD 0x0: > Pin Default 0x40f000f6: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x6 > Pin-ctls: 0x00: > Power: setting=D0, actual=D0 > Connection: 2 > 0x24* 0x25 > Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x0810: OUT > Pin Default 0x40f000f7: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x7 > Pin-ctls: 0x00: > Connection: 1 > 0x25 > Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital > 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 0x22 [Audio Output] wcaps 0x40211: Stereo Digital > 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 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x24 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x21* 0x1c 0x1d > Node 0x25 [Audio Selector] wcaps 0x300101: Stereo > Connection: 3 > 0x22* 0x1c 0x1d > Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > Node 0x27 [Pin Complex] wcaps 0x400000: Mono > Pincap 0x0820: IN > Pin Default 0x40f000f0: [N/A] Other at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Pin-ctls: 0x00: > Node 0x28 [Volume Knob Widget] wcaps 0x600000: Mono > Volume-Knob: delta=1, steps=127, direct=1, val=127 > Connection: 2 > 0x10* 0x11 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw---- 1 root audio 116, 7 okt 6 12:40 /dev/snd/controlC0 > crw-rw---- 1 root audio 116, 6 okt 6 12:40 /dev/snd/hwC0D0 > crw-rw---- 1 root audio 116, 5 okt 6 12:40 /dev/snd/pcmC0D0c > crw-rw---- 1 root audio 116, 4 okt 6 12:41 /dev/snd/pcmC0D0p > crw-rw---- 1 root audio 116, 3 okt 6 11:41 /dev/snd/seq > crw-rw---- 1 root audio 116, 2 okt 6 11:41 /dev/snd/timer > > > !!Aplay/Arecord output > !!------------ > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 0/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] > Subdevices: 2/2 > Subdevice #0: subdevice #0 > Subdevice #1: subdevice #1 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xf6fdc000 irq 21' > Mixer name : 'IDT 92HD71B7X' > Components : 'HDA:111d76b2' > Controls : 19 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 127 > Mono: Playback 94 [74%] [-24.75dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Front',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 127 > Mono: > Front Left: Playback 127 [100%] [0.00dB] [on] > Front Right: Playback 127 [100%] [0.00dB] [on] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [on] > Front Right: Capture 0 [0%] [0.00dB] [on] > Simple mixer control 'Capture Mux',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 1 [33%] [10.00dB] > Front Right: 1 [33%] [10.00dB] > Simple mixer control 'Capture Mux',1 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Capture',1 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 0 [0%] [0.00dB] [off] > Front Right: Capture 0 [0%] [0.00dB] [off] > Simple mixer control 'Analog Loopback',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Analog Loopback',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'Digital Input Source',0 > Capabilities: enum > Items: 'Analog Inputs' 'Digital Mic 1' > Item0: 'Analog Inputs' > Simple mixer control 'Digital Mic 1',0 > Capabilities: cvolume > Capture channels: Front Left - Front Right > Limits: Capture 0 - 3 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > Simple mixer control 'Input Source',1 > Capabilities: cenum > Items: 'Mic' 'Front Mic' > Item0: 'Mic' > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.2 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.3 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > value false > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Analog Loopback' > index 1 > value false > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > value.0 0 > value.1 0 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 15' > comment.dbmin 0 > comment.dbmax 2250 > iface MIXER > name 'Capture Volume' > index 1 > value.0 0 > value.1 0 > } > control.9 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.10 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Capture Mux Volume' > index 1 > value.0 0 > value.1 0 > } > control.11 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Front Playback Volume' > value.0 127 > value.1 127 > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.13 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Headphone Playback Volume' > value.0 127 > value.1 127 > } > control.14 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.15 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 3' > comment.dbmin 0 > comment.dbmax 3000 > iface MIXER > name 'Digital Mic 1 Capture Volume' > value.0 0 > value.1 0 > } > control.16 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 'Analog Inputs' > comment.item.1 'Digital Mic 1' > iface MIXER > name 'Digital Input Source' > value 'Analog Inputs' > } > control.17 { > comment.access 'read write' > comment.type INTEGER > comment.count 1 > comment.range '0 - 127' > comment.dbmin -9525 > comment.dbmax 0 > iface MIXER > name 'Master Playback Volume' > value 94 > } > control.18 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Master Playback Switch' > value true > } > control.19 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > comment.dbmin -5100 > comment.dbmax 0 > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > snd_hda_intel > snd_hwdep > nls_cp437 > vfat > fat > loop > isofs > udf > nvidia > ppdev > lp > parport > acpi_cpufreq > cpufreq_performance > cpufreq_stats > cpufreq_ondemand > cpufreq_powersave > cpufreq_conservative > freq_table > ipv6 > af_packet > hfsplus > nls_utf8 > ntfs > nls_base > fuse > snd_pcm_oss > snd_pcm > snd_page_alloc > snd_mixer_oss > pcmcia > firmware_class > snd_seq_dummy > rtc_cmos > snd_seq_oss > rtc_core > rtc_lib > dcdbas > snd_seq_midi_event > snd_seq > snd_timer > snd_seq_device > serio_raw > psmouse > i2c_i801 > snd > pcspkr > evdev > i2c_core > yenta_socket > rsrc_nonstatic > pcmcia_core > soundcore > video > output > wmi > ac > button > battery > intel_agp > ext3 > jbd > mbcache > dm_mirror > dm_log > dm_snapshot > dm_mod > firewire_sbp2 > sg > sr_mod > cdrom > sd_mod > usbhid > hid > ff_memless > ahci > libata > scsi_mod > dock > ricoh_mmc > sdhci > mmc_core > firewire_ohci > firewire_core > crc_itu_t > ehci_hcd > uhci_hcd > usbcore > e1000e > thermal > processor > fan > > > On Monday 06 October 2008 11:44:33 you wrote: > > At Sun, 5 Oct 2008 22:10:46 +0200, > > > > Klaas De Craemer wrote: > > > Hello, > > > > > > A few days ago, I posted a patch to the linux kernel list, but afterwards > > > I realised that it is more appropriate for the alsa list. So I repost it > > > here. > > > > > > Below is my patch for an issue on Dell E6500 notebooks with the Sigmatel > > > stac92hd71b7 codec. In the default snd-hda-intel configuration, the sound > > > going to the headphone jack goes comes directly from the DAC, resulting > > > in very low volume output. The patch changes the codec's widget > > > connections so that the internal mixer is put in between and sets its > > > input amplifiers to the maximum value. The output to the internal speaker > > > port has been left untouched. > > > I have tested this and it gives a much better output. > > > > Thanks for the patch. > > > > I wonder whether this really improves the quality. > > Basically what you does is to set the volume of the input mixer over > > 0dB, and this is usually bad when really high signals come in. > > > > Just to be sure, is the output pin set to the headphone pin control? > > You can give alsa-info.sh output (run with --no-upload option) for > > further analysis. The script is found in > > http://www.alsa-project.org/alsa-info.sh > > > > > > thanks, > > > > Takashi > >

2 6