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[alsa-devel] Event notification
by Miloslaw Smyk 20 Dec '07

20 Dec '07

Hi, My app needs to know about the events occurring to a sound card. Specifically, I am interested in start, stop, pause and volume change. While I could monitor volume by similar means as alsamixer does (is there any doc on this apart from the source code?), I am not sure if monitoring other events is at all possible. I would appreciate any suggestions as to how to do this. Kind Regards, Milek

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[alsa-devel] pcm file question
by Leigh Stewart 20 Dec '07

20 Dec '07

hey all, im writing a little app that needs to tie into the audio in/out-put from another alsa application. in other words i need to take all audio produced by the app and do some work with it, and supply data to its audio input. im not sure but i think the pcm_file plugin will allow me to do what i need to do. i.e. i as i understand it i could tie the audio output to a fifo file using the plugin, and then read from the file. anyway im not sure how write a program that uses the plugin code. i wrote something like ... > #include <alsa/asoundlib.h> > #include <iostream> > #include <alsa/pcm.h> > #include <alsa/pcm_plugin.h> > ... > err = snd_pcm_file_open (&pcm_handle, pcm_name, (char*)0, fdo, (char*)0, > fdi, "raw", 777, capture_handle, 0); > if(err < 0) { > fprintf (stderr, "cannot open pcm_file device %s (%s)\n", pcm_name, > snd_strerror (err)); > exit (1); > } but there are no method signatures in pcm_plugin.h. i've tried building alsa from source 1.0.15. i see method signatures for copy and dmix, and even jack, but not for file. how can i compile the plugin in? can i even use pcm_file programmatically? can anyone refer me to any examples? thanks, lstewart

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[alsa-devel] [PATCH 3/12] neo1973: ASoC include pathname fix
by Harald Welte 19 Dec '07

19 Dec '07

[PATCH] Fix s3c24xx include file path changes in asoc driver Signed-off-by: Harald Welte <laforge(a)openmoko.org> --- Index: linux-2.6.22/sound/soc/s3c24xx/neo1973_wm8753.c =================================================================== --- linux-2.6.22.orig/sound/soc/s3c24xx/neo1973_wm8753.c +++ linux-2.6.22/sound/soc/s3c24xx/neo1973_wm8753.c @@ -30,13 +30,15 @@ #include <asm/mach-types.h> #include <asm/hardware/scoop.h> -#include <asm/arch/regs-iis.h> #include <asm/arch/regs-clock.h> #include <asm/arch/regs-gpio.h> #include <asm/hardware.h> #include <asm/arch/audio.h> #include <asm/io.h> #include <asm/arch/spi-gpio.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + #include "../codecs/wm8753.h" #include "lm4857.h" #include "s3c24xx-pcm.h" Index: linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c =================================================================== --- linux-2.6.22.orig/sound/soc/s3c24xx/s3c24xx-i2s.c +++ linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -33,13 +33,14 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-iis.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> #include <asm/dma.h> #include <asm/arch/dma.h> +#include <asm/plat-s3c24xx/regs-iis.h> + #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -- - Harald Welte <laforge(a)openmoko.org> http://openmoko.org/ ============================================================================ Software for the world's first truly open Free Software mobile phone

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[alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by John Stile 19 Dec '07

19 Dec '07

I am trying to use a Via VT1708 audio card ( uses the snd-hda-intel.ko driver) to record from SPDIF-in. An off-list reply to my posting to alsa-users list suggested that snd-hda-intel.ko might not support SPDIF-in with the VT1708 card, which brought me here. Can someone who works on the Intel HDA driver tell me if it supports SPDIF-in on the VT1708 audio card? Should 'arecord -L' show "Digital Audio Input"? Environment: Audio Card: VT1708 ALSA version: 1.0.15 (as gentoo packages) Driver: snd-hda-intel.ko (In-kernel) Kernel: 2.6.23-gentoo-r3 Distribution: Gentoo Systemboard: PX10000G (VIA pico-itx) Case: Serener GS-L08 case

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[alsa-devel] interesting message, but not sure where its coming from
by Christopher Johnson 19 Dec '07

19 Dec '07

I'm working on a complicated program that uses the ALSA library in an embedded environment. I have things working pretty well, although occasionally I get the following message printed on the console: buffer empty, loop_cnt 4727, loop_cnt_of_last_copy 4682, last period elapsed 4700, elapsed flag 0, chip_data->prev_cnt F000 buffer empty, loop_cnt 4758, loop_cnt_of_last_copy 4727, last period elapsed 4700, elapsed flag 0, chip_data->prev_cnt F000 I know this isn't my code, and I can't find anything that looks like this in the alsa library or the kernel. Does anybody recognize this?

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[alsa-devel] Fix capture mixers of ALC662 models
by Herton Ronaldo Krzesinski 19 Dec '07

19 Dec '07

The below patches fixes capture mixers of ALC662 models. The first one applies to current hg tree, the second one to Linus tree (2.6.24-rcX). The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-18 21:57:02.000000000 -0200 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-18 22:01:24.000000000 -0200 @@ -12364,17 +12364,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12398,17 +12387,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12422,17 +12400,6 @@ static struct snd_kcontrol_new alc662_le HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12463,6 +12430,25 @@ static struct snd_kcontrol_new alc662_ch { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -12585,26 +12571,6 @@ static struct hda_verb alc662_auto_init_ { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12746,8 +12712,7 @@ static struct alc_config_preset alc662_p .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..85e54d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11728,17 +11728,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11762,17 +11751,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11786,17 +11764,6 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11827,6 +11794,25 @@ static struct snd_kcontrol_new alc662_chmode_mixer[] = { { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -11949,26 +11935,6 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12109,8 +12075,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, -- []'s Herton

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[alsa-devel] [PATCH] Dmux mixer conflict
by Matthew Ranostay 19 Dec '07

19 Dec '07

DMUX mixers need to be created before input source mixers, otherwise some STAC9228 codecs fail to record correctly or at all. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r fbe3540c8df4 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 18 15:27:34 2007 -0700 @@ -834,15 +834,16 @@ static int stac92xx_build_controls(struc if (err < 0) return err; - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } if (spec->num_dmuxes > 0) { stac_dmux_mixer.count = spec->num_dmuxes; err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&stac_dmux_mixer, codec)); + if (err < 0) + return err; + } + + for (i = 0; i < spec->num_mixers; i++) { + err = snd_hda_add_new_ctls(codec, spec->mixers[i]); if (err < 0) return err; }

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[alsa-devel] rate plugin issue
by Takashi Iwai 19 Dec '07

19 Dec '07

Hi, I committed your patch now on HG tree, as it's likely OK from the possible breakage on other apps, too. Anyway, I swear that I'll never read any mails during my next vacation. That was so exhausting to be dragged to a lengthy discussion for band-aiding a badly designed compoment... :) Takashi

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[alsa-devel] speaker-test stalls
by Dave Dillow 19 Dec '07

19 Dec '07

While working with Fernando to get the sis7019 driver working in his buildroot setup, we kept running into a lockup of the PCM stream when using speaker-test. aplay was fine, as was other I finally tracked it down to a combination of uclibc not doing block reads, which led to XRUNs, and what I thought was issue in snd_pcm_lib_write1() when trying to fill up the tail of the PCM buffer with a large write, when there was an intervening small write. As it turns out, the lockup problem may simply be in speaker-write's sw_params settings -- it calls snd_pcm_sw_params_set_avail_min() to set the minium write to the period size, and snd_pcm_sw_params_set_start_threshold() to start once the buffer has been filled. This doesn't account for odd wave file lengths and XRUNs. The problem arises when the wave data is not a multiple of the period size, and you get an XRUN when there is less than buffer_size samples remaining in the current loop. So for example, period size is 8192, buffer is 32768, and the wave file has 71042 samples (Front_Left.wav for speaker-test). speaker-test will break that into 8 writes of 8192 samples, and one of 5506. If you get a XRUN in the last 32k of the file, the PCM will be stopped, and prepared, and we'll start loading data into again. Because we have a tail of 5506 samples, we get to the point where the available space is less than our minimum, and then snd_pcm_lib_write1() will wait forever for the available space to be >= runtime->avail_min, even though the channel is stopped. [snip from a trace, patch to pcm_lib.c attached] writei size 8192 avail 32768 min 8192 xfer 1 state 2 writei size 8192 avail 24576 min 8192 xfer 1 state 2 writei size 5506 avail 16384 min 8192 xfer 1 state 2 writei size 8192 avail 10878 min 8192 xfer 1 state 2 writei size 8192 avail 2686 min 8192 xfer 1 state 2 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 Assuming that this is desired behavior from the kernel side, then I see three options to fix this in speaker-test: 1) snd_pcm_sw_params_set_avail_min() to 1 sample 2) snd_pcm_sw_params_set_start_threshold() to start when (nperiods - 1) are ready (or be even more aggressive) 3) call snd_pcm_drain() at the end of each loop to clear out the buffer. This won't help if the wave file has fewer samples than the buffer size. Options 1 or 2 seem to be best. Thoughts? Dave

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[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h
by Andrew Paprocki 19 Dec '07

19 Dec '07

Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.c Tue Dec 18 21:48:58 2007 -0500 @@ -1435,7 +1435,8 @@ int snd_hda_create_spdif_out_ctls(struct return err; } codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.h Tue Dec 18 21:48:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,10 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_realtek.c Tue Dec 18 21:48:58 2007 -0500 @@ -572,7 +572,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -586,7 +586,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

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