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[alsa-devel] [PATCH] hda: 92hd71bxxx DMIC nid
by Matthew Ranostay 20 Mar '08

20 Mar '08

Added missing DMIC verb to dell_4_1_pin_configs[]. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6ffc943c3b39 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Mar 18 17:18:18 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Mar 18 14:09:17 2008 -0400 @@ -1317,7 +1317,7 @@ static unsigned int ref92hd71bxx_pin_con static unsigned int dell_m4_1_pin_configs[13] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, 0x40f000f0, 0x4f0000f0, };

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[alsa-devel] [PATCH]model for alc262 to support Lenovo 3000
by Jiang zhe 20 Mar '08

20 Mar '08

Hi, Please refer to [0003856] on the alsa mantis. This model is to support the Lenovo 3000 y410. Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com>

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Re: [alsa-devel] Random inverted audio channels
by Alexandre BOUIN 20 Mar '08

20 Mar '08

> I'm currently working on an internal project. > Here is some details : > This embedded project includes a AT91SAM9260 CPU and a WM8900 codec. > Audio driver is using ASoC API. > Audio data could be either switch from I2S or DSP A, but I prefer the > second one. > Audio Codec is master. > > We have an issue : when we play a right channel stereo track, sound is > played on right channel, but sometimes on left. > This issue could happen right after starting kernel, or later after 10- > 20 sounds played. Hi list, I'm back on this issue. I have few questions about it and I'd hope someone could answer : - does someone have already used AT91 SSC controller in slave mode ? Can he send me his register configuration and tell if he already heard and L/R channels swaps ? - is there a way to play a sound at the lowest software level, maybe without using alsa or dsp devices ? - where is the software audio buffer ? As you can see I really want to trace any difficulties to find where comes this problem. As always, if you have any other ideas to test, I'll be glad to hear them. Thanks and regards, Alex.

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[alsa-devel] ALSA and SALSA install
by Alan Horstmann 19 Mar '08

19 Mar '08

Following on from thread 'Problems with ALSA in embedded device', running code such as aplay with both ALSA and SALSA libs. Is there a way to install both at the same time, so on a PC code can be compiled and experimented with first -lasound and then -lsalsa without alternately re-installing the libraries with make install? Thanks Alan

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[alsa-devel] Same pci id for different ASUS machine
by Jiang zhe 19 Mar '08

19 Mar '08

Hi, According to the [0003839] https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3839 The PCI ID of ASUS F9S is 1043:1339 and "model=lenovo" works for it. But I found that ASUS G1 have the same id and model is 3stack in the alsa driver. Which one is true? --- alsa-driver-hg20080313/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 09:00:08.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-13 10:38:57.000000000 +0800 @@ -12591,7 +12591,7 @@ static struct snd_pci_quirk alc861vd_cfg SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x1339, "Asus F9S", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),

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[alsa-devel] HW decoding abstracted in ALSA
by Harsha priya gupta 19 Mar '08

19 Mar '08

Hi, I am new to ALSA and audio stack in Linux. I have a sound card that can do HW decoding as well. I am looking at writing an ALSA driver such that whatever may be the player or MMF above, it should use the HW decoder and not the inherit SW decoder that is available with the player. Can someone help me with some pointers or existing like-drivers? Thanks in advance, Harsha

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[alsa-devel] About question of "ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default"
by wangxf 19 Mar '08

19 Mar '08

alsa-devel，你好： I am new to ALSA, now I try to add one sound soc chip in linux 2.6.21 kernel on samsung arm designed board. I have do some underside work: 1: I have porting linux 2.6.21 kernel on samsung arm designed board.(log.txt add accessories) 2. kernel have recognise sound chip form log information. 3. I have a test app working in linux 2.6.18.(this test app is tested ok!) 4. I have download "alsa-lib-1.0.15.tar.bz2" and buid in linux host(Linux 2.6.18-5-686 #1 SMP Fri Jun 1 00:47:00 UTC 2007),create libasound.so.2.0.0. build order: 1)./configure --host=arm-linux CC=arm-linux-gcc ( i do not affirm this config) 2) make 3) and then change to su to use to make install. 5. download libasound.so.2.0.0 lib to samsung designed board and download test app to samsung designed board. 6. but i run test ,system prompt err: ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: No such file or directory Segmentation fault now,I doubt libasound.so.2.0.0 lib is conflict with kernel 2.6.21. hope help . 夏新移动通讯有限公司软件设计工程师王小锋电话：86－592－6516777EXT6643 E-MAIL:wangxf@amoi.com.cn wangxf 2008-03-19

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[alsa-devel] Having a fight with snd_sctl_install:src/control/setup.c in alsa-lib-1.0.14a
by Bouterse, Bob 19 Mar '08

19 Mar '08

I am using alsa-lib-1.0.14a on a VIA EPIA 15000g board. I am hoping that someone can help me. I have been in an entrenched fight with the sctl functions setting my ctl setting back to the config file values. I am working on some extensions to the alsa audio output plugins in xine to provide simultaneous analog and iec958 on a Via EPIA 15000g, board. The issue is that I need to set the iec958 status bits at runtime, but everytime I call: /*---------------------------------------------------------------------- -----*/ ... snd_ctl_elem_value_get_iec958(cval, &iec958); llprintf(LOG_MIXER,"Read AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2 */ iec958.status[0] = IEC958_AES0_NONAUDIO | IEC958_AES0_CON_NOT_COPYRIGHT; iec958.status[1] = IEC958_AES1_CON_PCM_CODER | IEC958_AES1_CON_ORIGINAL; iec958.status[2] = 0x00; iec958.status[3] = IEC958_AES3_CON_FS_48000; llprintf(LOG_MIXER,"Writing AES0=0x%02x,AES1=0x%02x,AES2=0x%02x,AES3=0x%02x\n", iec958.status[0], iec958.status[1], iec958.status[2], iec958.status[3]); /* store the values */ snd_ctl_elem_value_set_iec958(cval, &iec958); if ((err = snd_ctl_elem_write(ctl_handle, cval)) < 0) { xprintf (this->class->xine, XINE_VERBOSITY_DEBUG, "snd_ctl_elem_write(): %s\n", snd_strerror(err)); } ... /*---------------------------------------------------------------------- -----*/ snd_sctl_install resets the IEC958 status bits back to the original values! I am pulling my hair out trying to understand how to stop this. Unfortunately I don't have much time to fix this, please help. ---------------------------------------------------- Bob Bouterse Sr. Software Engineer Office of the CTO D&M Holdings Inc. 1-317-616-6541 (fax) 1-317-616-6541

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[alsa-devel] [PATCH] Use base ALC883 mixer for 6stack-dell model
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the same as alc883_base_mixer. Avoid duplicated code and use alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer definition. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 17:54:42.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 18:00:08.000000000 -0300 @@ -6992,46 +6992,6 @@ static struct snd_kcontrol_new alc888_3s { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - 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", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7941,7 +7901,7 @@ static struct alc_config_preset alc883_p .input_mux = &alc883_capture_source, }, [ALC888_6ST_DELL] = { - .mixers = { alc888_6st_dell_mixer, alc883_chmode_mixer }, + .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_6st_dell_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, -- []'s Herton

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[alsa-devel] [PATCH] Remove now uneeded 6stack-hp model from ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

After DAC assignment fix in ALC883, the 6stack-hp model is now the same as 6stack-dig. So just remove 6stack-hp model and replace its use with 6stack-dig. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/Documentation/ALSA-Configuration.txt.orig alsa-kernel/Documentation/ALSA-Configuration.txt --- alsa-kernel/Documentation/ALSA-Configuration.txt.orig 2008-03-18 17:41:33.000000000 -0300 +++ alsa-kernel/Documentation/ALSA-Configuration.txt 2008-03-18 17:43:06.000000000 -0300 @@ -879,7 +879,6 @@ Prior to version 0.9.0rc4 options had a lenovo-nb0763 Lenovo NB0763 lenovo-ms7195-dig Lenovo MS7195 haier-w66 Haier W66 - 6stack-hp HP machines with 6stack (Nettle boards) 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 17:30:24.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 17:41:13.000000000 -0300 @@ -197,7 +197,6 @@ enum { ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, ALC883_HAIER_W66, - ALC888_6ST_HP, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, @@ -6955,46 +6954,6 @@ static struct snd_kcontrol_new alc883_me { } /* end */ }; -static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - 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", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc883_mux_enum_info, - .get = alc883_mux_enum_get, - .put = alc883_mux_enum_put, - }, - { } /* end */ -}; - static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7720,7 +7679,6 @@ static const char *alc883_models[ALC883_ [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", [ALC883_HAIER_W66] = "haier-w66", - [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", @@ -7739,7 +7697,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), - SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), @@ -7972,17 +7930,6 @@ static struct alc_config_preset alc883_p .unsol_event = alc883_haier_w66_unsol_event, .init_hook = alc883_haier_w66_automute, }, - [ALC888_6ST_HP] = { - .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs }, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .dac_nids = alc883_dac_nids, - .dig_out_nid = ALC883_DIGOUT_NID, - .dig_in_nid = ALC883_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), - .channel_mode = alc883_sixstack_modes, - .input_mux = &alc883_capture_source, - }, [ALC888_3ST_HP] = { .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, -- []'s Herton

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