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[alsa-devel] AIC23 SOC Driver on OSK5912
by Jesslyn Abdul Salam 07 Oct '08

07 Oct '08

Hi all, 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. 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 ? - Jesslyn Abdul Salam

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[alsa-devel] [PATCH 2.6.27?] ASoC: Correct inverted Mic PGA Switch control in wm8510 driver
by Mark Brown 07 Oct '08

07 Oct '08

Mic PGA Switch should be inverted in the WM8510 driver but isn't. Reported-by: ext-jukka.hynninen(a)vaisala.com Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8510.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 75ed041..9a37c8d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -199,7 +199,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; static const struct snd_kcontrol_new wm8510_boost_controls[] = { -SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Mic PGA Switch", WM8510_INPPGA, 6, 1, 1), SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0), SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0), }; -- 1.5.6.5

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[alsa-devel] [PATCH] snd-core: enlarge snd_card.components for up to 4 codecs
by Wu Fengguang 07 Oct '08

07 Oct '08

Enlarge snd_card.components[80] to 128 bytes, with space for 4 codecs. The previous size 80 cannot support HP 2230s which has 3 codecs. Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/include/sound/asound.h b/include/sound/asound.h index ca2f358..b3801b0 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -708,7 +708,7 @@ struct snd_ctl_card_info { unsigned char longname[80]; /* name + info text about soundcard */ unsigned char reserved_[16]; /* reserved for future (was ID of mixer) */ unsigned char mixername[80]; /* visual mixer identification */ - unsigned char components[80]; /* card components / fine identification, delimited with one space (AC97 etc..) */ + unsigned char components[128]; /* card components / fine identification, delimited with one space (AC97 etc..) */ unsigned char reserved[48]; /* reserved for future */ }; diff --git a/include/sound/core.h b/include/sound/core.h index f52ab6f..dcbc424 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -116,7 +116,7 @@ struct snd_card { char shortname[32]; /* short name of this soundcard */ char longname[80]; /* name of this soundcard */ char mixername[80]; /* mixer name */ - char components[80]; /* card components delimited with + char components[128]; /* card components delimited with space */ struct module *module; /* top-level module */

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[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for process_unsol_events(). Signed-off-by: Wu Fengguang<wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..7717a92 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -314,7 +314,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) } /* - * process queueud unsolicited events + * process queued unsolicited events */ static void process_unsol_events(struct work_struct *work) {

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[alsa-devel] [PATCH] hda: comment typo fix
by Wu Fengguang 07 Oct '08

07 Oct '08

fix a typo in comment for is_in_nid_list(). Signed-off-by: Wu Fengguang <wfg(a)linux.intel.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c742e10..6268667 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2807,7 +2807,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } /* - * Helper for automatic ping configuration + * Helper for automatic pin configuration */ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list)

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: add new param mux to dapm_mux_update_power
by Richard Zhao 07 Oct '08

07 Oct '08

Function dapm_mux_update_power needs enum index mux and register mask value val as parameters, but it only has a parameter val, and uses it as both val and mux. snd_soc_test_bits(widget->codec, e->reg, mask, val) val is register mask here, e->texts[val] but val should be enum index mux here. This patch adds a new param mux to fix it. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: Fix wrong param when call snd_soc_test_bits
by Richard Zhao 07 Oct '08

07 Oct '08

Fix wrong param when call snd_soc_test_bits in function dapm_mux_update_power. Signed-off-by: Richard Zhao <linuxzsc(a)gmail.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c016426..9ca9c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -693,7 +693,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, - int val, struct soc_enum* e) + int mux, int val, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -709,12 +709,12 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, if (path->kcontrol != kcontrol) continue; - if (!path->name || ! e->texts[val]) + if (!path->name || !e->texts[mux]) continue; found = 1; /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[val]))) + if (!(strcmp(path->name, e->texts[mux]))) path->connect = 1; /* new connection */ else path->connect = 0; /* old connection must be powered down */ @@ -1291,7 +1291,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, -- 1.5.4.3

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Re: [alsa-devel] [Alsa-user] [SOLVED] Re: snd_hda_intel on ALC662 (Classmate 2nd gen) does not mute and does not offer mic boost
by Vedran Miletić 07 Oct '08

07 Oct '08

http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commit;h=ac… See diff of that commit for some hints. Or don't. I got it wrong. :-( Basically, you need to find out subsystem ID for your soundcard. "lspci -vnn" will be your friend here. You should find out something like this for your soundcard: 00:07.0 Audio device [0403]: nVidia Corporation Device [10de:0774] (rev a1) Subsystem: ASRock Incorporation Device [1849:3662] <------ THIS IS THE IMPORTANT ROW Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 21 Memory at f9e78000 (32-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel So I should have put 0x1849, 0x3662 for my souncard instead of 0x1849, 0x0774. I probably wouldn't notice, but now I will patch it. Thanks for asking! 2008/10/4 Pedro Pessoa <pessoa(a)angulosolido.pt>: > On Saturday 04 October 2008 17:54, Vedran Miletić wrote: >> Can you create a patch to add this device to quirks list? > > Sure! Let me know what I have to do. > > The quirks list is something that will make 'auto' load ECS? > > -- > Angulo Sólido - Tecnologias de Informação > http://angulosolido.pt > -- Vedran Miletić

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[alsa-devel] [PATCH] ALSA: snd-usb-audio: support for Edirol UA-4FX device
by pedro.lopez.cabanillas＠gmail.com 06 Oct '08

06 Oct '08

From: Pedro Lopez-Cabanillas <pedro.lopez.cabanillas(a)gmail.com> Renamed the old quirk function for ua-700/ua-25 to become more generic, moving the MIDI interfaces to the quirk data header. Added a new quirk for the Edirol UA-4FX. 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 c5cf682..05c7849 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2996,12 +2996,12 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25 interface. The only way - * to detect the sample rate is by looking at wMaxPacketSize. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. */ -static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, @@ -3016,37 +3016,12 @@ static int create_ua700_ua25_quirk(struct snd_usb_audio *chip, struct audioformat *fp; int stream, err; - /* both PCM and MIDI interfaces have 2 altsettings */ - if (iface->num_altsetting != 2) + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) return -ENXIO; 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 ua25_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk ua25_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua25_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, - &ua25_quirk); - } - if (altsd->bNumEndpoints != 1) return -ENXIO; @@ -3376,9 +3351,9 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index f3ca77f..36e4f7a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,9 +159,9 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA700_UA25, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UA101, + QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 3f68359..ad7a318 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -855,15 +855,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } }, { .ifnum = -1 @@ -1197,15 +1201,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_AUDIO_EDIROL_UAXX }, { .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA700_UA25 + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } }, { .ifnum = -1 @@ -1338,6 +1346,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * This quirk is for the "Advanced Driver" mode. If off, the UA-4FX + * is standard compliant, but has only 16-bit PCM and no MIDI. + */ + USB_DEVICE(0x0582, 0x00a3), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-4FX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, /* TODO: add Edirol MD-P1 support */ { USB_DEVICE(0x582, 0x00a6), -- 1.5.2.4

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Re: [alsa-devel] ICH9 Sound
by Ronald Horn 06 Oct '08

06 Oct '08

Hello, all mixer levels are ok. I have several times checked all settings. I deinstalled and reinstalled the drivers in yast2, too. The phenomenon is that then a sound is playing, once an incredible noice is playing over the original sound. And this sound is stopping first about ten seconds after I closed the player. It's like an feedback loop. Greetings Ronald > ICH9 sound should be supported. And, as you see, your card is > detected. Perhaps the codec or that specific notebook isn't supported. > Are you sure mixer levels are adjusted properly? > > I'm using a HP Pavilion dv7-1045eg. The Problem is I have no sound, > > because there is no driver witch works. > > Here are some reports from my system: > > / > > lspci -nv/ > > 00:1b.0 0403: 8086:293e (rev 03) > > Subsystem: 103c:30f4 > > Flags: bus master, fast devsel, latency 0, IRQ 22 > > Memory at df300000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: [50] Power Management version 2 > > Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ > > Queue=0/0 Enable+ > > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > > Capabilities: [100] Virtual Channel <?> > > Capabilities: [130] Root Complex Link <?> > > Kernel driver in use: HDA Intel > > Kernel modules: snd-hda-intel > > > > / > > cat /proc/asound/cards/ > > 0 [Intel ]: HDA-Intel - HDA Intel > > HDA Intel at 0xdf300000 irq 22 > > > > > > /head -n 1 /proc/asound/card0/codec*/ > > ==> /proc/asound/card0/codec#0 <== > > Codec: IDT 92HD71B7X > > ==> /proc/asound/card0/codec#2 <== > > Codec: Generic 10de ID 6 > > > > > > I'm using openSuse 11.0 with Kernel 2.6.25.16-0.1-pae. > > Maybe your project could develop some drivers.

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