December 2007 - Alsa-devel - mailman.alsa-project.org

[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

2 3

[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; }

2 2

[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

3 11

[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

2 5

[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 */

2 1

[alsa-devel] Firmware problem
by Adrian McMenamin 19 Dec '07

19 Dec '07

My user - yes there is one - of the aica sound driver on the Dreamcast is having a problem with formware loading. This is what he has sent me - anyone know what is wrong? When loading snd-aica it gives: [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Inter then after like 15 or 20 seconds it will give: FATAL: Error inserting snd_aica (/lib/modules/2.6.24-rc4/kernel/sound/sh/snd-aica.ko): Unknown symbol in module, or unknown parameter (see dmesg) and dmesg gives this: [ 54.632531] fill_kobj_path: path = '/module/snd_page_alloc' [ 54.643434] kobject notes: registering. parent: snd_page_alloc, set: <NULL> [ 54.645602] kobject_uevent_env [ 54.646181] kobject filter function caused the event to drop! [ 54.669703] kobject soundcore: registering. parent: <NULL>, set: module [ 54.671737] kobject holders: registering. parent: soundcore, set: <NULL> [ 54.673616] kobject_uevent_env [ 54.675375] kobject filter function caused the event to drop! [ 54.677096] kobject_uevent_env [ 54.678780] fill_kobj_path: path = '/module/soundcore' [ 54.690116] kobject notes: registering. parent: soundcore, set: <NULL> [ 54.690904] kobject_uevent_env [ 54.693891] kobject filter function caused the event to drop! [ 54.695845] device class 'sound': registering [ 54.697748] kobject sound: registering. parent: <NULL>, set: class [ 54.699657] kobject_uevent_env [ 54.701350] fill_kobj_path: path = '/class/sound' [ 54.787597] kobject snd: registering. parent: <NULL>, set: module [ 54.789664] kobject holders: registering. parent: snd, set: <NULL> [ 54.791557] kobject_uevent_env [ 54.793340] kobject filter function caused the event to drop! [ 54.795228] kobject_uevent_env [ 54.797010] fill_kobj_path: path = '/module/snd' [ 54.808092] kobject notes: registering. parent: snd, set: <NULL> [ 54.808886] kobject_uevent_env [ 54.811852] kobject filter function caused the event to drop! [ 54.849584] kobject snd_timer: registering. parent: <NULL>, set: module [ 54.850378] kobject holders: registering. parent: snd_timer, set: <NULL> [ 54.853372] kobject_uevent_env [ 54.855118] kobject filter function caused the event to drop! [ 54.856902] kobject_uevent_env [ 54.858680] fill_kobj_path: path = '/module/snd_timer' [ 54.869736] kobject notes: registering. parent: snd_timer, set: <NULL> [ 54.871802] kobject_uevent_env [ 54.872452] kobject filter function caused the event to drop! [ 54.875569] DEV: registering device: ID = 'timer' [ 54.877400] kobject sound: registering. parent: virtual, set: <NULL> [ 54.881274] kobject_uevent_env [ 54.883351] unset subsystem caused the event to drop! [ 54.886599] kobject timer: registering. parent: sound, set: devices [ 54.888593] kobject_uevent_env [ 54.890420] fill_kobj_path: path = '/devices/virtual/sound/timer' [ 55.000017] kobject snd_pcm: registering. parent: <NULL>, set: module [ 55.002208] kobject holders: registering. parent: snd_pcm, set: <NULL> [ 55.004074] kobject_uevent_env [ 55.004636] kobject filter function caused the event to drop! [ 55.007502] kobject_uevent_env [ 55.009274] fill_kobj_path: path = '/module/snd_pcm' [ 55.020295] kobject notes: registering. parent: snd_pcm, set: <NULL> [ 55.021108] kobject_uevent_env [ 55.024045] kobject filter function caused the event to drop! [ 55.053022] kobject snd_aica: registering. parent: <NULL>, set: module [ 55.055131] kobject holders: registering. parent: snd_aica, set: <NULL> [ 55.057074] kobject_uevent_env [ 55.057643] kobject filter function caused the event to drop! [ 55.060587] kobject_uevent_env [ 55.061260] fill_kobj_path: path = '/module/snd_aica' [ 55.073491] kobject notes: registering. parent: snd_aica, set: <NULL> [ 55.074288] kobject_uevent_env [ 55.077295] kobject filter function caused the event to drop! [ 55.079239] bus platform: add driver AICA [ 55.079929] kobject AICA: registering. parent: <NULL>, set: drivers [ 55.082948] kobject_uevent_env [ 55.084712] fill_kobj_path: path = '/bus/platform/drivers/AICA' [ 55.089946] Registering platform device 'AICA'. Parent at platform [ 55.091828] DEV: registering device: ID = 'AICA' [ 55.093558] kobject AICA: registering. parent: platform, set: devices [ 55.093744] bus platform: add device AICA [ 55.093849] kobject_uevent_env [ 55.093918] fill_kobj_path: path = '/devices/platform/AICA' [ 55.094045] platform: Matched Device AICA with Driver AICA [ 55.094096] platform: Probing driver AICA with device AICA [ 55.094492] DEV: registering device: ID = 'card0' [ 55.094564] kobject sound: registering. parent: AICA, set: <NULL> [ 55.094646] kobject_uevent_env [ 55.094682] unset subsystem caused the event to drop! [ 55.094739] kobject card0: registering. parent: sound, set: devices [ 55.094888] kobject_uevent_env [ 55.094959] fill_kobj_path: path = '/devices/platform/AICA/sound/card0' [ 55.095133] DEV: registering device: ID = 'pcmC0D0p' [ 55.095195] kobject pcmC0D0p: registering. parent: card0, set: devices [ 55.095454] kobject_uevent_env [ 55.095528] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/pcmC0D0p' [ 55.095745] DEV: registering device: ID = 'controlC0' [ 55.095811] kobject controlC0: registering. parent: card0, set: devices [ 55.095980] kobject_uevent_env [ 55.096051] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/controlC0' [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Intelligent Sound Processor [ 55.128339] bound device 'AICA' to driver 'AICA' [ 55.128394] platform: Bound Device AICA to Driver AICA [ 57.018452] DEV: registering device: ID = 'firmware-AICA' [ 57.019121] kobject firmware: registering. parent: AICA, set: <NULL> [ 57.022447] kobject_uevent_env [ 57.023025] unset subsystem caused the event to drop! [ 57.023099] kobject firmware-AICA: registering. parent: firmware, set: devices [ 57.027620] kobject_uevent_env [ 57.029196] kobject filter function caused the event to drop! [ 57.029842] kobject_uevent_env [ 57.032535] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.955604] DEV: Unregistering device. ID = 'firmware-AICA' [ 115.955788] kobject_uevent_env [ 115.955898] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.956090] kobject firmware-AICA: cleaning up [ 115.956144] kobject firmware: cleaning up [ 115.960813] kobject holders: unregistering [ 115.960863] kobject_uevent_env [ 115.960900] kobject filter function caused the event to drop! [ 115.960972] kobject holders: cleaning up [ 115.961018] kobject snd_aica: unregistering [ 115.961059] kobject_uevent_env [ 115.961128] fill_kobj_path: path = '/module/snd_aica'

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[alsa-devel] How do I use SNDRV_PCM_INFO_JOINT_DUPLEX?
by Timur Tabi 19 Dec '07

19 Dec '07

I'm working on an ASoC driver. I have a limitation in my hardware that if I have an active substream, I cannot re-program the hardware registers that control the sample size. This means, for example, if I'm playing 16-bit audio, and I want to start capture, I cannot program the sample size into the hardware. My only choice is to have already programmed the capture sample size *before* I started playback. The easiest way to implement this is to make sure that the second stream has the same sample size as the first. That way, I won't need to reprogram the registers, because they'll already be programmed correctly. I discovered the SNDRV_PCM_INFO_JOINT_DUPLEX option, but I can't figure out how to use it. I know it needs to be set in my snd_pcm_hardware.info, but then what? How do I tell ALSA that the "joint" part is just that the sample sizes must be the same? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] uniwill (0x1558, 0x9075) dup
by Andy Shevchenko 18 Dec '07

18 Dec '07

Hi! I investigated the duplicate problem and try to compare two cases. I don't deep to search of the specifications, but the current variant (with asus-laptop) should be more correct. In last HG snapshot second case already comented. I just noticed on my reseaches. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

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[alsa-devel] [PATCH] hda: STAC927x DMIC Cleanup
by Matthew Ranostay 18 Dec '07

18 Dec '07

Cleaned up STAC927x and added several subsystem id's for more laptops. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 4a204306b295 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Dec 14 14:42:41 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Dec 14 12:37:16 2007 -0700 @@ -107,6 +107,7 @@ enum { STAC_D965_3ST, STAC_D965_5ST, STAC_DELL_3ST, + STAC_DELL_BIOS, STAC_927X_MODELS }; @@ -1408,22 +1409,24 @@ static unsigned int dell_3st_pin_configs static unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, - 0x403003fa, 0x40000100, 0x40000100, 0x404003fb, + 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb, 0x40c003fc, 0x40000100 }; static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { - [STAC_D965_REF] = ref927x_pin_configs, - [STAC_D965_3ST] = d965_3st_pin_configs, - [STAC_D965_5ST] = d965_5st_pin_configs, - [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_D965_REF] = ref927x_pin_configs, + [STAC_D965_3ST] = d965_3st_pin_configs, + [STAC_D965_5ST] = d965_5st_pin_configs, + [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_DELL_BIOS] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { - [STAC_D965_REF] = "ref", - [STAC_D965_3ST] = "3stack", - [STAC_D965_5ST] = "5stack", - [STAC_DELL_3ST] = "dell-3stack", + [STAC_D965_REF] = "ref", + [STAC_D965_3ST] = "3stack", + [STAC_D965_5ST] = "5stack", + [STAC_DELL_3ST] = "dell-3stack", + [STAC_DELL_BIOS] = "dell-bios", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -1450,13 +1453,21 @@ static struct snd_pci_quirk stac927x_cfg SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), + /* Dell 3 stack systems with verb table in BIOS */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), @@ -1992,6 +2003,7 @@ static int stac92xx_add_dyn_out_pins(str for (i = 0; i < codec->num_nodes; i++) { wcaps = codec->wcaps[i]; wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) num_dacs++; } @@ -2079,7 +2091,6 @@ static int stac92xx_auto_fill_dac_nids(s wcaps = snd_hda_param_read(codec, conn[j], AC_PAR_AUDIO_WIDGET_CAP); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) continue; @@ -3293,8 +3304,11 @@ static int patch_stac927x(struct hda_cod stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0 || + stac927x_brd_tbl[spec->board_config] != NULL) { + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); if (err < 0) { stac92xx_free(codec); @@ -3306,59 +3320,46 @@ static int patch_stac927x(struct hda_cod stac92xx_set_config_regs(codec); } + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + spec->multiout.dac_nids = spec->dac_nids; + switch (spec->board_config) { case STAC_D965_3ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_D965_5ST: + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; - case STAC_D965_5ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_DELL_BIOS: + case STAC_DELL_3ST: + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = STAC927X_NUM_DMICS; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; + spec->dmux_nids = stac927x_dmux_nids; break; default: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; } - switch (codec->subsystem_id) { - case 0x10280242: /* STAC 9228 */ - case 0x102801f3: - case 0x1028020A: - case 0x10280209: - spec->dmic_nids = stac927x_dmic_nids; - spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nids = stac927x_dmux_nids; - - /* Enable DMIC0 */ - stac92xx_set_config_reg(codec, 0x13, 0x90a60040); - - /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000004; - break; - default: - spec->num_dmics = 0; - - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; - } - - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + stac92xx_enable_gpio_mask(codec); - err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) {

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[alsa-devel] snd-hda-intel: HG regression - NULL pointer dereference
by Ivan N. Zlatev 18 Dec '07

18 Dec '07

When loading snd-hda-intel from Alsa HG I get a NULL pointer dereference and the trace below. The card is with the following codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 The trace: ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2792: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2796: speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2800: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2808: inputs: mic=0x13, fmic=0x0, line=0xe, fline=0x0, cd=0x0, aux=0x0 ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2088: dac_nids=2 (0x5/0x4/0x0/0x0/0x0) BUG: unable to handle kernel NULL pointer dereference at virtual address 00000000 printing eip: f91cfcb0 *pde = 00000000 Oops: 0000 [#1] SMP last sysfs file: /devices/pci0000:00/0000:00:1c.1/0000:0c:00.0/cmd Modules linked in: snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep snd soundcore nls_iso8859_1 nls_cp437 vfat fat usb_storage ide_core rndis_host cdc_ether usbnet mii aes_i586 ieee80211_crypt_ccmp af_packet xt_tcpudp xt_pkttype ipt_LOG xt_limit vboxdrv ipt_REJECT xt_state iptable_mangle iptable_nat nf_nat iptable_filter nf_conntrack_ipv4 nf_conntrack nfnetlink ip_tables ip6_tables x_tables cpufreq_conservative cpufreq_userspace cpufreq_powersave acpi_cpufreq speedstep_lib microcode apparmor loop dm_mod rfcomm l2cap nvidia(P) ipw3945 sdhci ohci1394 ieee1394 mmc_core ieee80211 hci_usb ac ieee80211_crypt battery i2c_i801 button tg3 rtc_cmos bluetooth rtc_core sr_mod rtc_lib cdrom intel_agp firmware_class i2c_core agpgart sg usbhid hid ff_memless sd_mod ehci_hcd uhci_hcd usbcore edd ext3 mbcache jbd fan ata_piix ahci libata scsi_mod thermal processor CPU: 1 EIP: 0060:[<f91cfcb0>] Tainted: P N VLI EFLAGS: 00010246 (2.6.22.13-0.3-default #1) EIP is at stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] eax: 00000000 ebx: 00000000 ecx: f54b7014 edx: d7e241ec esi: d7e24014 edi: f71e6014 ebp: 00000000 esp: d9ea1ea0 ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process alsactl (pid: 25097, ti=d9ea0000 task=c7457030 task.ti=d9ea0000) Stack: 00000000 d7e240a4 f54b7014 cbe43694 fffffff3 f911c2b3 cbe43694 c2175154 df8a4c14 df8a4d60 df8a4c14 00000000 f54b7014 df8a4dc4 f911cd85 cb940b7c c03c5aec f52693c0 f52693c0 bf8ef020 c2175154 c0172331 c016e87a dfbef2c0 Call Trace: [<f911c2b3>] snd_ctl_elem_write+0xa6/0xe6 [snd] [<f911cd85>] snd_ctl_ioctl+0x494/0x7af [snd] [<c0172331>] chrdev_open+0x0/0x133 [<c016e87a>] __dentry_open+0xe4/0x178 [<c016e988>] nameidata_to_filp+0x24/0x33 [<c016e9ce>] do_filp_open+0x37/0x3e [<f911c8f1>] snd_ctl_ioctl+0x0/0x7af [snd] [<c017a13d>] do_ioctl+0x21/0xa0 [<c017a3f3>] vfs_ioctl+0x237/0x249 [<c017a451>] sys_ioctl+0x4c/0x67 [<c0104e22>] sysenter_past_esp+0x6b/0xa9 [<c02c0000>] unix_bind+0x5e/0x28e ======================= Code: 31 c0 c3 57 89 d1 56 53 83 ec 08 8b 78 60 8b 5a 3c 2b 58 44 8b 77 4c 8d 84 9e 90 00 00 00 8b 96 8c 00 00 00 89 44 24 04 8b 46 6c <0f> b7 04 58 89 04 24 89 f8 e8 99 24 ff ff 5f 5a 5b 5e 5f c3 53 EIP: [<f91cfcb0>] stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] SS:ESP 0068:d9ea1ea0 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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