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[alsa-devel] [ALSA] fix private data pointer calculation in CS4270 driver
by Timur Tabi 14 Nov '07

14 Nov '07

Fix the calculation of the private_data pointer in the CS4270 driver. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5d601ad..abac628 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -725,7 +725,8 @@ static int cs4270_probe(struct platform_device *pdev) codec->owner = THIS_MODULE; codec->dai = &cs4270_dai; codec->num_dai = 1; - codec->private_data = codec + ALIGN(sizeof(struct snd_soc_codec), 4); + codec->private_data = (void *) codec + + ALIGN(sizeof(struct snd_soc_codec), 4); socdev->codec = codec; -- 1.5.2.4

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[alsa-devel] mixer API documentation
by Richard Kettlewell 13 Nov '07

13 Nov '07

Hello, Where is the ALSA mixer API documentation? I looked at http://www.alsa-project.org/alsa-doc/alsa-lib/mixer.html, but that only says "Mixer interface is designed to access the abstracted mixer controls. This is an abstraction layer over the hcontrol layer". ttfn/rjk

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[alsa-devel] [PATCH] ASoC TLV320AIC3X codec driver
by Vladimir Barinov 13 Nov '07

13 Nov '07

Hi Liam, This patch adds ALSA SoC support for TI TLV320AIC3X audio codecs. The features that are supported: o Capture/Playback/Bypass. o 16/20/24/32 bit audio. o 8k - 96k sample rates. o codec master only mode o DAPM. Vladimir

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[alsa-devel] Intel HD Audio
by Chris Pemberton 13 Nov '07

13 Nov '07

I've been trying to fix audio on my Lenovo 3000 N100 laptop for quite some time; on this list and via bug #0003468. I finally got tired of the xorg graphics woes and hacking for sound, so I am now "temporarily" running Vista. I'd like to get this audio fixed permanently, so if you need any details from the working Vista drivers, let me know before I get sick of Windows and switch back to Linux. Chris

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[alsa-devel] New developer question
by Pete Fawcett 13 Nov '07

13 Nov '07

Dear Alsa Developers I am trying to find out how to submit some fixes to ALSA drivers. All I could find on the web site was to subscribe to this mailing list so I hope this is the correct place to raise this. I have made some changes to the driver for AD1988 device. This is an HDA device and handled by the snd_hda_intel module. The specific hardware is handled by patch_analog.c I have made some changes so that 5.1 surround sound is now output on my motherboard. Previously stereo sound was output to all all three pairs of speakers. That's all fine for me, but in the spirit of open source I thought I should submit these changes for wider use. Please advise me how I proceed from here. Thanks Pete

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[alsa-devel] issues with pulse plugin
by Mike Gorse 13 Nov '07

13 Nov '07

I have run into what is basically another instance of bug 3470 (pulse_hw_params asserting because the stream is already initialized). The application that I am working with is calling snd_pcm_hw_params each time it plays a sample. The following patch makes things work for me (actually the app still doesn't work because it uses poll() in a way that is incompatible with the pulse plugin, but that is a separate issue, and I don't know if it should be considered a bug in the plugin.) --- pulse/pcm_pulse.c.orig 2007-11-08 15:13:12.000000000 -0500 +++ pulse/pcm_pulse.c 2007-11-08 15:13:12.000000000 -0500 @@ -497,7 +497,7 @@ assert(pcm->p); //Resolving bugtrack ID 0003470 - if(!(base && snd_pcm_state(base) == SND_PCM_STATE_PREPARED)) + if(!(base && (snd_pcm_state(base) == SND_PCM_STATE_PREPARED || snd_pcm_state(base) == SND_PCM_STATE_SETUP))) assert(!pcm->stream); pa_threaded_mainloop_lock(pcm->p->mainloop); -- I couldn't add this to the bug because it was marked as resolved. Any comments? Or is the app behaving in a way that is counter to the specification? Also, I'm still encountering bug 2601 (a separate assert failure that happens if period_size == 0) even with the patch to pcm_ioplug.c. I think that PERIOD_TIME and PERIODS need to be constrained; I added a couple of comments to the bug. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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[alsa-devel] HDA Intel SigmaTel STAC9228 in Dell Vostro 1400 defines different pin nids
by Ivan N. Zlatev 13 Nov '07

13 Nov '07

Hello, I've bought a new laptop Dell Vostro 1400 with the following audio card: 00:1b.0 Audio device [Class 0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02) and with the SigmaTel Codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 I was able to get sound to come out of the internal speakers by explicitly specifying either the 3stack or the dell-3stack models with alsa 1.0.14 . Unfortunately with neither of those does the internal microphone work. Now I am running ALSA HG and have decided to try and fix my problem by adding the proper pinconfig for my card (this is what I've done before for my Mac and it worked). I downloaded the win32 drivers which include all pinconfigs for all Dell machines with intel hda cards. The problem I am facing is that the win32 driver .ini provides a configuration for 13 pins which do not match the ones in patch_sigmatel.c . The ones in ALSA are: static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; Where as the ones defined by the driver are [1]: 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x21, 0x22 So I am wondering how should I approach this? I have attached my codec info and mixer output without the model to be specified. [1] win32 driver pinconfigs [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x02211210 AltCfg = hex: 20,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\0B] CfgDflt = dword: 0x400003FA [HKR\Settings\Pin\0C] CfgDflt = dword: 0x400003FB [HKR\Settings\Pin\0D] CfgDflt = dword: 0x90170310 ExtAmpCtrl = hex: 02,05,02 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x0281103E Bias = hex: FF,04,02 AltCfg = hex: 31,10,A1,02,22,10,11,02 Action = hex: 0A,0E [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0221122F AltCfg = hex: 21,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\10] CfgDflt = dword: 0x400003FC [HKR\Settings\Pin\11] CfgDflt = dword: 0x400003FD [HKR\Settings\Pin\12] CfgDflt = dword: 0x400003FE [HKR\Settings\Pin\13] CfgDflt = dword: 0x90A60330 [HKR\Settings\Pin\14] CfgDflt = dword: 0x400003FF [HKR\Settings\Pin\21] CfgDflt = dword: 0x04441340 [HKR\Settings\Pin\22] CfgDflt = dword: 0x40C003F1 [HKR\Settings\GPIO] 02 = hex: 0x82 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] aplay: xrun:1104: read/write error ???
by 13 Nov '07

13 Nov '07

Hi All: im using aplay from ALSA-1.0.15 to play music through bluetooth a2dp. im using montavista 2.6.18 on ARM9 platform. I got an "aplay: xrun:1104: read/write error, state = RUNNING" error after the music has been played for one second or two. Could someone tell me how this happened? Here's the command that i use : aplay -v -D bluetooth /1.wav And the message log is as follows: /$ aplay -v -D bluetooth /1.wav audio[842]: Accepted new client connection on unix socket (fd=8) audio[842]: Package PKT_TYPE_CFG_REQ:0 audio[842]: avdtp_ref(0x3c990): ref=2 audio[842]: a2dp_source_request_stream: selected SEP 0x3c218 audio[842]: avdtp_ref(0x3c990): ref=3 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[842]: session_cb audio[842]: DISCOVER request succeeded audio[842]: seid 1 type 1 media 0 in use 0 audio[842]: seid 2 type 1 media 0 in use 0 audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 1 type 1 media 0 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 2 type 1 media 0 audio[842]: Discovery complete audio[842]: avdtp_set_configuration(0x3c990): int_seid=1, acp_seid=1 audio[842]: session_cb audio[842]: SET_CONFIGURATION request succeeded audio[842]: SBC Source: Set_Configuration_Cfm audio[842]: avdtp_ref(0x3c990): ref=4 audio[842]: stream state changed: IDLE -> CONFIGURED audio[842]: session_cb audio[842]: OPEN request succeeded audio[842]: stream state changed: CONFIGURED -> OPEN audio[842]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[842]: SBC Source: Open_Cfm audio[842]: session_cb audio[842]: START request succeeded audio[842]: SBC Source: Start_Cfm audio[842]: SBC Source SEP 0x3c218 locked audio[842]: fd=10, fd_opt=1, pkt_len=672, sample_size=2, rate=44100 audio[842]: 34 bytes sent audio[842]: 1 bytes sent audio[842]: avdtp_unref(0x3c990): ref=3 audio[842]: stream state changed: OPEN -> STREAMING Playing WAVE '/1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Bluetooth Audio Device Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 6400 period_size : 128 period_time : 2902 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 128 xfer_align : 128 start_threshold : 6400 stop_threshold : 6400 silence_threshold: 0 silence_size : 0 boundary : 1677721600 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) Status(R/W): state : RUNNING trigger_time: 24.586883000 tstamp : 0.000000 delay : 0 avail : 6912 avail_max : 6912 aplay: xrun:1104: read/write error, state = RUNNING ---------------->>>>> this is where this error happens audio[842]: Unix client disconnected (fd=8) audio[842]: SBC Source SEP 0x3c218 unlocked audio[842]: avdtp_unref(0x3c990): ref=2 thanks semiyd _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] [PATCH] cmipci - allow capture of raw spdif subframes
by Timofei V. Bondarenko 13 Nov '07

13 Nov '07

Hi. Enable capturing of raw 32bit IEC958_SUBFRAME. The 24-bits PCM data can be obtained using iec958 plugin. Known problem: captured stream may begin with either left or right subframe. Since the iec958 plugin doesn't decode preamble it may swap the channels sometime. Regards, Tim.

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[alsa-devel] patch for Dell 1210 which using stac9221 codec
by zhejiang 13 Nov '07

13 Nov '07

Hi, Dell 1210 have three jacks on the front. Two of them serve as output jack and one serve as microphone. The microphone can be switched to output model too. There are some problems with current dell 1210 driver: 1.Unable to use the front microphone 2.Unable to switch the microphone 3.If plug the left hp,another hp will be muted,so it's impossible to work as 6-channel According to the issue 0003427 on alsa-bugzilla, I wrote a patch and the reporter helped me to test it on his machine. It can support the front microphone. When switch the microphone to output model, it can mute/unmute the internal speaker with jack-sensing. diff -Nur a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 08:00:08.000000000 +0800 +++ b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 17:07:27.000000000 +0800 @@ -171,6 +171,10 @@ 0x02, }; +static hda_nid_t dell_m82_1210_dac_nids[3] = { + 0x02, 0x05, 0x03, +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -874,10 +878,10 @@ 102801D7 (Dell XPS M1210) */ static unsigned int dell_922x_m82_pin_configs[10] = { - 0x0221121f, 0x408103ff, 0x02111212, 0x90100310, - 0x408003f1, 0x02111211, 0x03451340, 0x40c003f2, + 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 0x508003f3, 0x405003f4, -}; +}; static unsigned int d945gtp3_pin_configs[10] = { 0x0221401f, 0x01a19022, 0x01813021, 0x01014010, @@ -1512,6 +1516,15 @@ AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | event)); +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1542,6 +1555,18 @@ pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } + + if (spec->board_config == STAC_922X_DELL_M82) { + if (val) { + /*If front mic serves as output,let it mute the internal speaker*/ + spec->autocfg.hp_outs = 3; + spec->autocfg.hp_pins[2] = 0x0c; + enable_pin_detect(codec, 0x0c, STAC_HP_EVENT); + } else { + spec->autocfg.hp_outs = 2; + } + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + } return 1; } @@ -1811,9 +1836,15 @@ struct sigmatel_spec *spec = codec->spec; unsigned int wid_caps; + int max_controls; + if (spec->board_config == STAC_922X_DELL_M82) { + max_controls = spec->multiout.num_dacs; + } else { + max_controls = cfg->line_outs; + } - for (i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < max_controls; i++) { if (!spec->multiout.dac_nids[i]) continue; @@ -2268,15 +2299,6 @@ AC_VERB_SET_GPIO_DATA, gpiostate); } -static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); -} - static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2677,6 +2699,11 @@ spec->multiout.dac_nids = spec->dac_nids; + if (spec->board_config == STAC_922X_DELL_M82) { + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = dell_m82_1210_dac_nids; + } + err = stac92xx_parse_auto_config(codec, 0x08, 0x09); if (!err) { if (spec->board_config < 0) { @@ -2687,6 +2714,14 @@ } err = -EINVAL; } + + if ((err >= 0) && (spec->board_config == STAC_922X_DELL_M82)) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Front Mic as Output Switch", (spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC] << 8) | 1); + /*In case that add_controls realloc the spec->kctl_alloc*/ + if ((err >= 0) && (spec->kctl_alloc != spec->mixers[spec->num_mixers - 1])) + spec->mixers[spec->num_mixers - 1] = spec->kctl_alloc; + } + if (err < 0) { stac92xx_free(codec); return err;

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