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[alsa-devel] Asus Eeepc missing in alsa models list?
by Herton Ronaldo Krzesinski 17 Oct '07

17 Oct '07

Looking at the change 291702f017efdfe556cb87b8530eb7d1ff08cbae, may be the addition of the Asus Eeepc model was missed? The following patch fixes it if it was not intended originally: diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4b48c2e..e2a1985 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53b0428..2ceaf4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_6ch] = "3stack-6ch", [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", + [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_AUTO] = "auto", }; -- []'s Herton

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 17 Oct '07

17 Oct '07

On Wednesday 17 October 2007 00:18:42 Thomas Meyer wrote: > $ dmesg > > [schnipp] > > ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 > PCI: Enabling bus mastering for device 0000:00:1b.0 > PCI: Setting latency timer of device 0000:00:1b.0 to 64 > hda_codec: STAC922x, Apple subsys_id=106b0200 > ACPI: PCI interrupt for device 0000:00:1b.0 disabled > HDA Intel: probe of 0000:00:1b.0 failed with error -16 > > [schnipp] > > > - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > Please bisect this (the alsa merge happened just short time ago) Which kernel did work? Best regards, Maxim Levitsky

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[alsa-devel] About the buffer_time and period_time
by Fengwei Yin 17 Oct '07

17 Oct '07

Hi all, I met a problem about period_time/buffer_time on PXA with ASOC enabled. The kernel version is 2.6.21. The Audio stream is 32k which is not supported by PXA ASOC driver. Please refer following RATES definition for PXA2xx. #define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) So I set plughw as default ALSA device enable auto samplerate conversion. The code to set the period_time/buffer_time is as following: err = snd_pcm_hw_params_set_rate_near(hPCM, params, &rate, 0); if (err < 0) { error("Rate not available: %d\n", rate); return err; } err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, 0); if (buffer_time > 500000) buffer_time = 500000; buffer_time = 400000; period_time = buffer_time/4; err = snd_pcm_hw_params_set_buffer_time_near(hPCM, params, &buffer_time, 0); if (err < 0) { error("Buffer time not available: %d, error number: %d\n", buffer_time, err); return err; } err = snd_pcm_hw_params_set_period_time_near(hPCM, params, &period_time, 0); if (err < 0) { error("Period time not available: %d\n", period_time); return err; } If I set buffer_time to 400000, I will get: " Buffer time not available: 400000, error number: -22 Cannot init audio driver. Floating point exception " If I set buffer time to 500000, buffer_time can be set correctly. But audio playback failed. If I set period_time before setting buffer_time, Everything is OK. Any thought about this problem. Thanks. Regards Yin, Fengwei

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[alsa-devel] Intel HDA digital out distortion
by pr0t0cod7 16 Oct '07

16 Oct '07

Hi, On Monday 23 July 2007 23:09:24 Ingmar Stieger wrote: > a couple of weeks ago, I bought an ASUS mainboard with integrated > Intel > HDA and an AD1988B codec. Unfortunately, the digital / SPDIF output > gets distorted with stock 1.0.14 drivers. It does not matter what the > volume levels of the various mixers are set to. I have the same issues. Any sound is distorted through spdif output (seems more pronounced with low frequencies.) Sound is not distorted: - using passthough (playing AC3) - when volume is set to the very minimal (no perceptible distortion) - with analog output My motherboard: ASUS P5B deluxe with integrated soundchip Kernel: 2.6.22 Alsa: Intel HD audio kernel built module asound.conf: http://pastebin.com/f2c54b47b > I think I might have found the source for this problem. Maybe it's > only a workaround, but it solved the issue and digital out now works > fine for me, so I thought I'd share. The patch worked for me, no more distortion :) Thanks for sharing, Alex.

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[alsa-devel] fsc esprimo v5505 (snd_intel_hda) - output in /var/log/messages
by Jan-Simon Möller 16 Oct '07

16 Oct '07

Hi! What do these messages mean ? Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Thanks ! Jan

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Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 16 Oct '07

16 Oct '07

On 10/07/2007 07:04 PM, Peteris Krisjanis wrote: >> "Current sample rate" only has meaning with the device open , and if I >> understand you right, a "cat >> /proc/asound/card0/pcm0{p,c}/sub0/hw_params will then get you what you >> want as to the "get" part (substitute for 0 in the above as required). > > I actually meant what kind of sample rate is set in sound card. For > example, my onboard sound card has fixed Hz. I would like to know such > information _before_ I touch sound card :) Okay, you want the information that the driver author supplied in the struct snd_pcm_hardware for the stream. I've in fact also wanted that information on a number of occasions and just looked at the driver source then. > So you suggest to open device via any means and then read that file? I > did this and it works. > I guess I can even try to implement it in app and check for sample > rate everytime before I play. > > But is this only solution? It's rather hack, and I would like to have > more reliable method. You might consider this less of of a hack -- it uses the ALSA API to query the information from the PCM handle directly. Please see the library docoumentation for other information you can extract from a hw_params struct. (I don't suppose you can ever get anything but dir == 0 for "hw" devices). Introducing a /proc/asound/card0/pcm0{p,c}/hw_limits or similar might not be a bad idea though? Rene /* gcc -W -Wall -o snd_rate snd_rate.c -l asound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(void) { snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; int err; snd_pcm_hw_params_alloca(&params); err = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_get_rate_min(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return -1; } printf("min: (%c) %u\n", "<=>"[dir + 1], val); err = snd_pcm_hw_params_get_rate_max(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return -1; } printf("max: (%c) %u\n", "<=>"[dir + 1], val); snd_pcm_close(handle); return 0; }

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[alsa-devel] intel-hda patch for Dell Latitude D531
by glogow＠fbihome.de 16 Oct '07

16 Oct '07

Hi I have attached three files to this mail: 1. patch_sigmatel.c.diff I have a Latitude D531 which has a ATI SB600 with a STAC 9205. The chip is detected, but no config was read with the latest drivers. The problems are the dmic_nids arrays, which are ignored for auto config lookup. This calles is_in_nid_list, which expects a 0 terminated list, but these arrays are not. Additionally I updated the dell_9205_m43_pin_configs with values from the latest win drivers. 2. create_dell_tables.pl When looking into the ini files and comparing the nids and config values of the various tables, I got really confused, so I wrote a litte script to extract the values and build the table. This is the extended version, after I got my sound working. It extracts all pin infos from all ini files, generates the apropriate tables and prepends the Version.txt from the driver, resultiung in a kind of C header. Maybe it's usefull for you. 3. patch_sigmatel_dell.h Output from running create_dell_tables.pl on the latest drivers for my D531 from Dell. HTH Jan-Marek Glogowski ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

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[alsa-devel] ALSA 1.0.15 final
by Jaroslav Kysela 16 Oct '07

16 Oct '07

Hello all, delayed but ... all ALSA packages 1.0.15 final are available for download on http://www.alsa-project.org . The changelog is available on web for 1.0.14 -> 1.0.15. I will make also available 1.0.15rc3 -> 1.0.15 changelog in next few hours. Have fun, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

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[alsa-devel] How to detect 'dead air' while recording?
by william estrada 16 Oct '07

16 Oct '07

Hi group, I want to filter out 'dead air' while recording. What is the best way to do this? I don't want to use JACK, just ALSA. Any URLs? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] LGPL Licensing
by WOODEN John 15 Oct '07

15 Oct '07

I wish to use either the ALSA or SALSA API within my application. I also intend to commercially distribute my application and sell it. What affect does ALSA's and/or SALSA's Lesser GPL license have upon applications that utilize their API's? Do I have to make the source code of my application freely available if I choose to utilize the ALSA or SALSA API? Thanks, John. IMPORTANT NOTICE: This e-mail and any attachment to it are intended only to be read or used by the named addressee. It is confidential and may contain legally privileged information. No confidentiality or privilege is waived or lost by any mistaken transmission to you. The RTA is not responsible for any unauthorised alterations to this e-mail or attachment to it. Views expressed in this message are those of the individual sender, and are not necessarily the views of the RTA. If you receive this e-mail in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or use any part of this e-mail if you are not the intended recipient.

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