Re: [Alsa-user] No sound on internal speaker
Card 1 is a Nvidia graphic card; it does not allow to play sound on the internal speaker. I found the solution. The Lenovo P520 is a dual codec workstation; ALC662 (hw:0,0) drives the rear audio plugs; ALC233 (hw:0,4) drives the front audio plugs and the internal speaker. So hw:0,4 should be used for playback. Hope it will help other people. NM On Thu, Jul 30, 2020 at 1:16 AM Alan Corey wrote: > Try using card 1. Look at amixer -c 1 > > On 7/29/20, Nicolas Martin wrote: > > Hi, > > > > I have a Lenovo P520 workstation with an internal speaker on which I'd > like > > to play sound. > > OS installed is RHEL 7U6 with alsa 1.1.6 > > > > the internal speaker is enabled in BIOS; however I don't have any sound > > output on it. > > > > The only way to have sound is to plug an external speaker. > > > > If I load the module snd-pcsp, there is an additional card that appears > > that I can use to play sound on the internal speaker, although the sound > > quality is very poor (crackling). > > > > I don't really know where to dig concerning this issue; whether it's a > bug > > with the kernel module or alsa; I'd appreciate some guidance. > > > > The audio chipset (not using the Nvidia output): > > # lspci -v | grep -i audio > > 00:1f.3 Audio device: Intel Corporation 200 Series PCH HD Audio > > 65:00.1 Audio device: NVIDIA Corporation GP107GL High Definition Audio > > Controller (rev a1) > > > > Below the output of aplay -l and aplay -L: > > # aplay -l > > List of PLAYBACK Hardware Devices > > card 0: PCH [HDA Intel PCH], device 0: ALC662 rev3 Analog [ALC662 rev3 > > Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 0: PCH [HDA Intel PCH], device 4: ALC233 Analog [ALC233 Analog] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] > > Subdevices: 1/1 > > Subdevice #0: subdevice #0 > > > > # aplay -L > > null > > Discard all samples (playback) or generate zero samples (capture) > > pulse > > PulseAudio Sound Server > > default:CARD=PCH > > HDA Intel PCH, ALC662 rev3 Analog > > Default Audio Device > > sysdefault:CARD=PCH > > HDA Intel PCH, ALC662 rev3 Analog > > Default Audio Device > > front:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > Front speakers > > surround21:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 2.1 Surround output to Front and Subwoofer speakers > > surround40:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 4.0 Surround output to Front and Rear speakers > > surround41:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 4.1 Surround output to Front, Rear and Subwoofer speakers > > surround50:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 5.0 Surround output to Front, Center and Rear speakers > > surround51:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 5.1 Surround output to Front, Center, Rear and Subwoofer speakers > > surround71:CARD=PCH,DEV=0 > > HDA Intel PCH, ALC662 rev3 Analog > > 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers > > hdmi:CARD=NVidia,DEV=0 > > HDA NVidia, HDMI 0 > > HDMI Audio Output > > hdmi:CARD=NVidia,DEV=1 > > HDA NVidia, HDMI 1 > > HDMI Audio Output > > hdmi:CARD=NVidia,DEV=2 > > HDA NVidia, HDMI 2 > > HDMI Audio Output > > hdmi:CARD=NVidia,DEV=3 > > HDA NVidia, HDMI 3 > > HDMI Audio Output > > > > > -- > - > No, I won't call it "climate change", do you have a "reality problem"? - > AB1JX > Cities are cages built to contain excess people and keep them from > cluttering up nature. > Impeach Impeach Impeach Impeach Impeach Impeach Impeach Impeach > ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] No sound on internal speaker
Hi, I have a Lenovo P520 workstation with an internal speaker on which I'd like to play sound. OS installed is RHEL 7U6 with alsa 1.1.6 the internal speaker is enabled in BIOS; however I don't have any sound output on it. The only way to have sound is to plug an external speaker. If I load the module snd-pcsp, there is an additional card that appears that I can use to play sound on the internal speaker, although the sound quality is very poor (crackling). I don't really know where to dig concerning this issue; whether it's a bug with the kernel module or alsa; I'd appreciate some guidance. The audio chipset (not using the Nvidia output): # lspci -v | grep -i audio 00:1f.3 Audio device: Intel Corporation 200 Series PCH HD Audio 65:00.1 Audio device: NVIDIA Corporation GP107GL High Definition Audio Controller (rev a1) Below the output of aplay -l and aplay -L: # aplay -l List of PLAYBACK Hardware Devices card 0: PCH [HDA Intel PCH], device 0: ALC662 rev3 Analog [ALC662 rev3 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 4: ALC233 Analog [ALC233 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default:CARD=PCH HDA Intel PCH, ALC662 rev3 Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, ALC662 rev3 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC662 rev3 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 3 HDMI Audio Output ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability
I also opened a ticket at Bugzilla but there is no follow-up on any place I try to get a fix on this. https://bugzilla.kernel.org/show_bug.cgi?id=203381 I am a total rookie so I don't know if the delays are normal and I don't want to be a total PITA so nothing get's solved. :-) Any suggestion? Nicolas Boisvert ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability
Here it is. This album is 24/96 BEGIN FILE ESI Audiotechnik GmbH GIGAPort HD+ at usb-:00:15.0-8.1, full speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 864 Momentary freq = 44100 Hz (0x2c.199a) Interface 1 Altset 1 Format: S16_LE Channels: 8 Endpoint: 1 OUT (ADAPTIVE) Rates: 44100 END FILE Nicolas -Message d'origine- De : Clemens Ladisch via Alsa-user Envoyé : 12 avril 2019 15:31 À : alsa-user@lists.sourceforge.net Objet : Re: [Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability Nicolas wrote: > https://www.esi-audio.com/products/gigaporthd+/ > > As I understand it, there is no specific Linux driver for this DAC as > there is one in Windows. The web site claims native CoreAudio supports, so there should not be any secret sauce. Please show the contents of /proc/asound/card?/stream0. Regards, Clemens ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Driver for ESI Gigaport HD+ for 24/96 capability
"ESI Gigaport HD+" This DAC has 8 channels and is 24/96 capable (if using 2 channels) but under 7.1, it is limited at 16/44. https://www.esi-audio.com/products/gigaporthd+/ As I understand it, there is no specific Linux driver for this DAC as there is one in Windows. Due to a lack of a linux driver, the DAC is always set to 8 channels so limited to 16/44. Hence, any FLAC above red book CD, like 24/96 for instance, are rendered at 16/44 which is frustrating for the HiFi community as folks that rely on this DAC do it for its exceptional acoustic quality and could not care less for the other 6 channels if it means downsampling. Nicolas Boisvert ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Stub for libpulse
Le nonidi 29 ventôse, an CCXXV, Ralf Mardorf a écrit : > Worked for me with Firefox 52.0 when at least making a test with > YouTube. Does it work for you with YouTube? > > $ pacman -Q apulse-git > apulse-git 0.1.7_13_gf445ae7-1 > > I can't test ogg at the moment, since I now have Firefox with alsa > enabled installed. Thanks for the info. I tested with ogg123 because I did not want to upgrade Firefox before being sure to have a solution. Strangely enough, it works with Firefox indeed (tested with "Für Elise" from Wikipedia rather than youtube, but that should not make any difference). Good news. Regards, -- Nicolas George -- Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Stub for libpulse
Hi. It seems that the Mozilla people have had the brilliant idea, starting with Firefox 52, to disable ALSA by default and only support PULSE. Apparently, they intend to remove the support for raw ALSA completely. Well, I do not want the PULSE server anywhere near my systems, and I believe many people here think the same. Does anyone know of a libpulse stub that provides the basic API but routes all the calls directly to libasound / ALSA? I have found this project: https://github.com/i-rinat/apulse but it seems unable to get something as simple as ogg123 working. Regards, -- Nicolas George signature.asc Description: Digital signature -- Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Enabling sound on Baytrail system
Hi. I am trying to get the audio controller working on a Lenovo Miix 3-1030 system, which is based on the Baytrail SST chipset. Some people have reported success on a similar (but not identical) setup. Currently, the further I could get is this: With the 4.1 kernel, when loading the snd_soc_sst_baytrail_pcm module, I get the following error messages: [6.208365] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.212216] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.244200] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.244691] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.268335] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.435501] (NULL device *): ipc: error DSP boot timeout and nothing more happens except the byt-rt5640 message gets printed a few more times later. The list in /proc/asound/cards is empty. The firmware file, intel/fw_sst_0f28.bin-48kHz_i2s_master, that I use comes from there: https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/ Its size is exactly 265684 and its SHA-1 is 2ddd16db3f587584d185ab7c0e2094dfb54b3828. Removing it causes a kernel message, so this file is used. Following hints on webforums, I also tried renaming the fw_sst_0f28.bin-i2s_master file from this repository: https://chromium.googlesource.com/chromiumos/third_party/linux-firmware/ (size 265892, SHA-1 46385c87d5e4d8267007d4353f7627db9715ca3a), with no changes. At some point, I managed to have cards detected (present in /proc) but I have not been able to reproduce the setting, and it did produce errors when trying to play something. Different kernel config and versions gave slightly different error messages, but this seems to be the most reliable config for the most recent kernel. Sound works with the OEM windows that I keep for just that kind of testing until mostly everything works. The entries in /sys/bus/acpi/devices/ (unfortunately, these are much less verbose than the corresponding lspci output) that seems relevant are: 10EC5640:00/adr:0x 10EC5640:00/hid:10EC5640 10EC5640:00/modalias:acpi:10EC5640:10EC5640: 10EC5640:00/path:\_SB_.I2C2.RTEK 10EC5640:00/power_state:D0 10EC5640:00/status:15 10EC5640:00/uevent:MODALIAS=acpi:10EC5640:10EC5640: 10EC5640:00/uid:1 - corresponding module snd-soc-rt5640 loaded 80860F28:00/adr:0x 80860F28:00/hid:80860F28 80860F28:00/modalias:acpi:80860F28:80860F28: 80860F28:00/path:\_SB_.LPEA 80860F28:00/power_state:D0 80860F28:00/real_power_state:D0 80860F28:00/status:15 80860F28:00/uevent:MODALIAS=acpi:80860F28:80860F28: 80860F28:00/uid:1 - corresponding modules snd-soc-sst-acpi and snd-intel-sst-acpi loaded ADMA0F28:00/adr:0x ADMA0F28:00/hid:ADMA0F28 ADMA0F28:00/modalias:acpi:ADMA0F28:ADMA0F28: ADMA0F28:00/path:\_SB_.LPEA.ADMA ADMA0F28:00/status:0 ADMA0F28:00/uevent:MODALIAS=acpi:ADMA0F28:ADMA0F28: ADMA0F28:00/uid:1 - Intel Audio DMA, no module found for this AMCR0F28:00/adr:0x AMCR0F28:00/hid:AMCR0F28 AMCR0F28:00/modalias:acpi:AMCR0F28:AMCR0F28: AMCR0F28:00/path:\_SB_.AMCR AMCR0F28:00/status:0 AMCR0F28:00/uevent:MODALIAS=acpi:AMCR0F28:AMCR0F28: AMCR0F28:00/uid:1 - Intel Audio Machine Driver, no module found for this SSPX:00/adr:0x SSPX:00/hid:SSPX SSPX:00/modalias:acpi:SSPX:SSPX: SSPX:00/path:\_SB_.LPEA.SSP1 SSPX:00/status:0 SSPX:00/uevent:MODALIAS=acpi:SSPX:SSPX: SSPX:00/uid:1 - Intel SSP Device, no module found for this Any help for that would be appreciated. For reference, my current efforts for configuring that hardware are summarized here: http://nsup.org/~george/comp/linux_lenovo_miix3/ Regards, -- Nicolas George signature.asc Description: Digital signature -- Monitor 25 network devices or servers for free with OpManager! OpManager is web-based network management software that monitors network devices and physical virtual servers, alerts via email sms for fault. Monitor 25 devices for free with no restriction. Download now http://ad.doubleclick.net/ddm/clk/292181274;119417398;o___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Enabling sound on Baytrail system
Hi. I am trying to get the audio controller working on a Lenovo Miix 3-1030 system, which is based on the Baytrail SST chipset. Some people have reported success on a similar (but not identical) setup. Currently, the further I could get is this: With the 4.1 kernel, when loading the snd_soc_sst_baytrail_pcm module, I get the following error messages: [6.208365] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.212216] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.244200] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.244691] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.268335] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [6.435501] (NULL device *): ipc: error DSP boot timeout and nothing more happens except the byt-rt5640 message gets printed a few more times later. The list in /proc/asound/cards is empty. The firmware file, intel/fw_sst_0f28.bin-48kHz_i2s_master, that I use comes from there: https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/ Its size is exactly 265684 and its SHA-1 is 2ddd16db3f587584d185ab7c0e2094dfb54b3828. Removing it causes a kernel message, so this file is used. Following hints on webforums, I also tried renaming the fw_sst_0f28.bin-i2s_master file from this repository: https://chromium.googlesource.com/chromiumos/third_party/linux-firmware/ (size 265892, SHA-1 46385c87d5e4d8267007d4353f7627db9715ca3a), with no changes. At some point, I managed to have cards detected (present in /proc) but I have not been able to reproduce the setting, and it did produce errors when trying to play something. Different kernel config and versions gave slightly different error messages, but this seems to be the most reliable config for the most recent kernel. Sound works with the OEM windows that I keep for just that kind of testing until mostly everything works. The entries in /sys/bus/acpi/devices/ (unfortunately, these are much less verbose than the corresponding lspci output) that seems relevant are: 10EC5640:00/adr:0x 10EC5640:00/hid:10EC5640 10EC5640:00/modalias:acpi:10EC5640:10EC5640: 10EC5640:00/path:\_SB_.I2C2.RTEK 10EC5640:00/power_state:D0 10EC5640:00/status:15 10EC5640:00/uevent:MODALIAS=acpi:10EC5640:10EC5640: 10EC5640:00/uid:1 - corresponding module snd-soc-rt5640 loaded 80860F28:00/adr:0x 80860F28:00/hid:80860F28 80860F28:00/modalias:acpi:80860F28:80860F28: 80860F28:00/path:\_SB_.LPEA 80860F28:00/power_state:D0 80860F28:00/real_power_state:D0 80860F28:00/status:15 80860F28:00/uevent:MODALIAS=acpi:80860F28:80860F28: 80860F28:00/uid:1 - corresponding modules snd-soc-sst-acpi and snd-intel-sst-acpi loaded ADMA0F28:00/adr:0x ADMA0F28:00/hid:ADMA0F28 ADMA0F28:00/modalias:acpi:ADMA0F28:ADMA0F28: ADMA0F28:00/path:\_SB_.LPEA.ADMA ADMA0F28:00/status:0 ADMA0F28:00/uevent:MODALIAS=acpi:ADMA0F28:ADMA0F28: ADMA0F28:00/uid:1 - Intel Audio DMA, no module found for this AMCR0F28:00/adr:0x AMCR0F28:00/hid:AMCR0F28 AMCR0F28:00/modalias:acpi:AMCR0F28:AMCR0F28: AMCR0F28:00/path:\_SB_.AMCR AMCR0F28:00/status:0 AMCR0F28:00/uevent:MODALIAS=acpi:AMCR0F28:AMCR0F28: AMCR0F28:00/uid:1 - Intel Audio Machine Driver, no module found for this SSPX:00/adr:0x SSPX:00/hid:SSPX SSPX:00/modalias:acpi:SSPX:SSPX: SSPX:00/path:\_SB_.LPEA.SSP1 SSPX:00/status:0 SSPX:00/uevent:MODALIAS=acpi:SSPX:SSPX: SSPX:00/uid:1 - Intel SSP Device, no module found for this Any help for that would be appreciated. For reference, my current efforts for configuring that hardware are summarized here: http://nsup.org/~george/comp/linux_lenovo_miix3/ Regards, -- Nicolas George -- Monitor 25 network devices or servers for free with OpManager! OpManager is web-based network management software that monitors network devices and physical virtual servers, alerts via email sms for fault. Monitor 25 devices for free with no restriction. Download now http://ad.doubleclick.net/ddm/clk/292181274;119417398;o ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Help configuring the E-MU 1820m
Le Lun 23 juillet 2007 23:25, James Courtier-Dutton a écrit : To capture sound on the EMU-1820m card, you need to run alsamixer, and adjust the EMU mixer (press F4 to get to it, it is over on the far right of the CAPTURE view) control to 0dB. This controls the digital capture gain from the EMU-1820m card. There is also a mixer control called DSP0 and DSP1 on the playback view. (should be in the capture view, but alsamixer has a bug) Adjust them to the Mic A input. then arecord -fdat -Ddefault test.wav should capture some sound from Mic A. Thank you very much for the instructions. I'll try them as soon as I have some time to play with my computer (which may wait till end of august) and report back if they don't work. -- Nicolas Mailhot - This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now http://get.splunk.com/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Help configuring the E-MU 1820m
Hi, I'm the owner of a second-hand EMU-1820m card, now supported in alsa thanks to James Courtier-Dutton efforts. I've mostly used it for analog digital multi channel output so far. Today I've decided to use it for microphone recording. Unfortunately while the hardware LED shows the micro is correctly plugged and the hardware is receiving stuff, I haven't managed to record anything in Linux (not even the usual microphone tap-tap-tap you do to check everything works) Can a good soul help me to configure alsa properly for a microphone? Just looking at the gnome alsa mixer makes me feel inadequate. (and trying to switch things there does not unmute the microphone input) I'd like to activate: Audiodock mono Mic A input Audiodock stereo Headphone output Audiodock stereo 1, 2, 3 outputs 1010 digital sp-dif (non optical) output Regards, -- Nicolas Mailhot signature.asc Description: Ceci est une partie de message numériquement signée - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Different PCM depending on the number of channels
Hi. Is it possible to make a PCM a that is a proxy (plug, whatever) for different PCM a1 and a2 depending on the number of channels is requested? In other words, I would like: aplay -c 6 -D pcm-a to be equivalent to: aplay -C 6 -D pcm-a1 while: aplay -c 2 -D pcm-a leads to: aplay -C 2 -D pcm-a2 Is there a way? signature.asc Description: Digital signature - This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] multi over dmix: no sound
Le septidi 27 frimaire, an CCXV, Ingo Müller a écrit : I'm very much interested in that issue, too. I had the same problem, having PCM not usable errors on some setups I tried. I finally gave up thinking that it just can't be done. Hoping that it will be possible one day, I filed a feature request. Maybe you could add a comment to show, that I'm not the only one desiring such a feature. If you should find a way how to do it with the current versions, please tell me how! Thanks in advance. Hi. I am glad I am not the only one with the problem. I did not manage to find your feature request. But I believe that if you did nothing wrong, nor I, the problem should be in a bug tracker and not a feature wish list. If it is not possible to use multi to bind two dmix, it should at least be written in the documentation. Regards, -- Nicolas George signature.asc Description: Digital signature - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.phpp=sourceforgeCID=DEVDEV___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] multi over dmix: no sound
Hi.
Consider the following .asoundrc:
pcm.mixer {
type dmix
ipc_key 1024
slave {
pcm {
type hw
card 0
}
period_time 0
period_size 1024
buffer_size 32768
rate 48000
}
bindings {
0 0
1 1
}
}
pcm.multi {
type multi
slaves {
a {
pcm mixer
channels 2
}
}
bindings {
0 { slave a channel 0 }
1 { slave a channel 1 }
}
}
Obviously the multi plugin is useless here, but it is a minimal test case
for a problem I had with a real situation.
If I play something through the mixer PCM, I get sound all right. But if I
send it through the multi PCM, I only get silence. Changing mixer to
hw:0 gets the sound back, but loses the mixer.
It is true with libasound 1.0.12 and .13 from Debian on two different
systems.
Does anyone see what I am doing wrong?
Regards,
--
Nicolas George
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[Alsa-user] RE : Re: Everything seems OK, but can't use amixer
Thank you Clemens, that was the problem.I didn't install alsa-lib with the Makefile, but copied directly the required libraries. The configuration files in /usr/share/alsa were missing.NicolasClemens Ladisch [EMAIL PROTECTED] a écrit: nicolas ricard wrote: I'm trying to install and configure alsa support for an embedded device, from scratch (no distro). My config is as follow : - ALSA 1.0.13 ... ~ # ls -al /dev/snd ... crw-rw-rw- 1 root root 116, 0 Jan 29 20:24 controlC0 ... ~ # amixer ALSA lib control.c:910:(snd_ctl_open_noupdate) Invalid CTL defaultWhen you installed alsa-lib, did it copy its configuration filesto /usr/share/alsa?Regards,Clemens Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire. - Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnkkid=120709bid=263057dat=121642___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Everything seems OK, but can't use amixer
HiI'm trying to install and configure alsa support for an embedded device, from scratch (no distro).My config is as follow :- kernel 2.6.13 , sound support selected as module, but no ALSA nor OSS support- ALSA 1.0.13- snd-intel8x0 compatible sound chipsetAfter a system boot, i get the following things :~ # lsmod | grep sndsnd_pcm_oss 47008 0 - Live 0xe011snd_mixer_oss 18560 1 snd_pcm_oss, Live 0xe00c8000snd_intel8x0 33564 0 - Live 0xe00a8000snd_ac97_codec 96292 1 snd_intel8x0, Live 0xe006d000snd_ac97_bus 2304 1 snd_ac97_codec, Live 0xe002d000snd_pcm 84360 3 snd_pcm_oss,snd_intel8x0,snd_ac97_codec, Live 0xe0086000snd_timer 25220 1 snd_pcm, Live 0xe0053000snd_page_alloc 10888 2 snd_intel8x0,snd_pcm, Live 0xe0032000snd 58212 7 snd_pcm_oss,snd_mixer_oss,snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer,snd_page_alloc, Live 0xe005d000soundcore 9952 1 snd, Live 0xe0013000~ # dmesg.ALSA /root/alsa-driver-1.0.13/pci/ac97/ac97_codec.c:2104: AC'97 0 analog subsections not readyintel8x0_measure_ac97_clock: measured 50054 usecsintel8x0: clocking to 48000.~ # cat /proc/asound/cards0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with AD1985 at 0xe0100c00, irq 177~ # cat /proc/asound/card0/oss_mixerVOLUME "Master" 0BASS "" 0TREBLE "" 0SYNTH "" 0PCM "PCM" 0SPEAKER "PC Speaker" 0LINE "Line" 0MIC "Mic" 0CD "CD" 0IMIX "" 0ALTPCM "" 0RECLEV "" 0IGAIN "Capture" 0OGAIN "" 0LINE1 "Aux" 0LINE2 "" 0LINE3 "" 0DIGITAL1 "IEC958" 0DIGITAL2 "" 0DIGITAL3 "" 0PHONEIN "Phone" 0PHONEOUT "Master Mono" 0VIDEO "Video" 0RADIO "" 0MONITOR "" 0~ # ls -al /dev | grep 14,crw-rw-rw- 1 root root 14, 14 Jan 29 20:24 admmidi0crw-rw-rw- 1 root root 14, 30 Jan 29 20:24 admmidi1crw-rw-rw- 1 root root 14, 46 Jan 29 20:24 admmidi2crw-rw-rw- 1 root root 14, 62 Jan 29 20:24 admmidi3crw-rw-rw- 1 root root 14, 12 Jan 29 20:24 adsp0crw-rw-rw- 1 root root 14, 28 Jan 29 20:24 adsp1crw-rw-rw- 1 root root 14, 44 Jan 29 20:24 adsp2crw-rw-rw- 1 root root 14, 60 Jan 29 20:24 adsp3crw-rw-rw- 1 root root 14, 13 Jan 29 20:24 amidi0crw-rw-rw- 1 root root 14, 29 Jan 29 20:24 amidi1crw-rw-rw- 1 root root 14, 45 Jan 29 20:24 amidi2crw-rw-rw- 1 root root 14, 61 Jan 29 20:24 amidi3crw-rw-rw- 1 root root 14, 11 Jan 29 20:24 amixer0crw-rw-rw- 1 root root 14, 27 Jan 29 20:24 amixer1crw-rw-rw- 1 root root 14, 43 Jan 29 20:24 amixer2crw-rw-rw- 1 root root 14, 59 Jan 29 20:24 amixer3crw-rw-rw- 1 root root 14, 4 Jan 29 20:24 audio0crw-rw-rw- 1 root root 14, 20 Jan 29 20:24 audio1crw-rw-rw- 1 root root 14, 36 Jan 29 20:24 audio2crw-rw-rw- 1 root root 14, 52 Jan 29 20:24 audio3crw-rw-rw- 1 root root 14, 10 Jan 29 20:24 dmfm0crw-rw-rw- 1 root root 14, 26 Jan 29 20:24 dmfm1crw-rw-rw- 1 root root 14, 42 Jan 29 20:24 dmfm2crw-rw-rw- 1 root root 14, 58 Jan 29 20:24 dmfm3crw-rw-rw- 1 root root 14, 9 Jan 29 20:24 dmmidi0crw-rw-rw- 1 root root 14, 25 Jan 29 20:24 dmmidi1crw-rw-rw- 1 root root 14, 41 Jan 29 20:24 dmmidi2crw-rw-rw- 1 root root 14, 57 Jan 29 20:24 dmmidi3crw-rw-rw- 1 root root 14, 3 Jan 29 20:24 dsp0crw-rw-rw- 1 root root 14, 19 Jan 29 20:24 dsp1crw-rw-rw- 1 root root 14, 35 Jan 29 20:24 dsp2crw-rw-rw- 1 root root 14, 51 Jan 29 20:24 dsp3crw-rw-rw- 1 root root 14, 2 Jan 29 20:24 midi00crw-rw-rw- 1 root root 14, 18 Jan 29 20:24 midi01crw-rw-rw- 1 root root 14, 34 Jan 29 20:24 midi02crw-rw-rw- 1 root root 14, 50 Jan 29 20:24 midi03crw-rw-rw- 1 root root 14, 0 Jan 29 20:24 mixer0crw-rw-rw- 1 root root 14, 16 Jan 29 20:24 mixer1crw-rw-rw- 1 root root 14, 32 Jan 29 20:24 mixer2crw-rw-rw- 1 root root 14, 48 Jan 29 20:24 mixer3crw-rw-rw- 1 root root 14, 8 Jan 29 20:24 musiccrw-rw-rw- 1 root root 14, 1 Jan 29 20:24 sequencercrw-rw-rw- 1 root root 14, 6 Jan 29 20:24 sndstat~ # ll /dev/mixerlrwxrwxrwx 1 root root 6 Jan 29 20:24 /dev/mixer - mixer0~ # ls -al /dev/snddrwxr-xr-x 2 root root 2048 Jan 29 20:24 .drwxr-xr-x 3 essais users 16384 Jan 30 21:16 ..crw-rw-rw- 1 root root 116, 0 Jan 29 20:24 controlC0crw-rw-rw- 1 root root 116, 32 Jan 29 20:24 controlC1crw-rw-rw- 1 root root 116, 64 Jan 29 20:24 controlC2crw-rw-rw- 1 root root 116, 96 Jan 29 20:24 controlC3crw-rw-rw- 1 root root 116, 4 Jan 29 20:24 hwC0D0crw-rw-rw- 1 root root 116, 5 Jan 29 20:24 hwC0D1crw-rw-rw- 1 root root 116, 6 Jan 29 20:24 hwC0D2crw-rw-rw- 1 root root 116, 7 Jan 29 20:24 hwC0D3crw-rw-rw- 1 root root 116, 36 Jan 29 20:24 hwC1D0crw-rw-rw- 1 root root 116, 37 Jan 29 20:24 hwC1D1crw-rw-rw- 1 root root 116, 38 Jan 29 20:24 hwC1D2crw-rw-rw- 1 root root 116, 39 Jan 29 20:24 hwC1D3crw-rw-rw- 1 root root 116, 68 Jan 29 20:24 hwC2D0crw-rw-rw- 1 root root 116, 69 Jan 29 20:24 hwC2D1crw-rw-rw- 1 root root 116, 70 Jan 29 20:24 hwC2D2crw-rw-rw- 1 root root 116, 71 Jan 29 20:24 hwC2D3crw-rw-rw- 1 root root 116, 100 Jan 29 20:24 hwC3D0crw-rw-rw- 1 root root 116, 101 Jan 29 20:24 hwC3D1crw-rw-rw- 1 root root 116, 102 Jan 29 20:24 hwC3D2crw-rw-rw- 1 root root 116, 103 Jan 29 20:24 hwC3D3crw-rw-rw- 1 root root 116, 8 Jan 29 20:24 midiC0D0crw-rw-rw- 1 root root 116, 9 Jan 29 20:24 midiC0D1crw-rw-rw-
Re: [Alsa-user] 1.0.0rc2 and snd-intel8x0
At 18:36 08/12/03 +0100, Clemens Ladisch wrote:
Nicolas Croiset (VDL) wrote:
We always have no sound... Even if I change the volume with xmixer.
Please try alsamixer, and make sure to unmute ('M') and raise the
volume of all channels.
Hello,
I used it in oss mode exclusively so alsamixer is not working.
With the ens1371, xmixer is working correctly with the same configuration.
I have also this problem with the emu10k1.
Bye.
Nicolas Croiset [EMAIL PROTECTED]
Tel : +33 4 72 84 06 04 Fax : +33 4 72 84 06 02
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[Alsa-user] EWX 24/96 Configuration
Hi! I am currently trying to configure my Terratec EWX 24/96 sound card but I am having some troubles having a 100% working config and I need help. I just run alsaconf in order to get everything up. 1) I can't use the alsa-out plugin of xmms (only OSS works). 2) I can't hear multiple sounds at once. I am using alsa 0.9.8. Thanks, Nicolas. --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] 1.0.0rc2 and snd-intel8x0
Hello, I use the card intel 8x0 : cat /proc/asound/card0/codec97#0/ac97#0-0 0-0/0: Analog Devices AD1885 Capabilities : -headphone out- DAC resolution : 16-bit ADC resolution : 16-bit 3D enhancement : Analog Devices Phat Stereo Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=0 DSA=0 VRA Extended status : VRA PCM front DAC: 44100Hz PCM ADC : 48000Hz AD18XX configuration Unchained: 0x1000,0x,0x Chained : 0x,0x,0x -- We always have no sound... Even if I change the volume with xmixer. cat /proc/asound/card0/oss_mixer VOLUME Master 0 BASS 0 TREBLE 0 SYNTH 0 PCM PCM 0 SPEAKER PC Speaker 0 LINE Line 0 MIC Mic 0 CD CD 0 IMIX 0 ALTPCM 0 RECLEV 0 IGAIN Capture 0 OGAIN 0 LINE1 Aux 0 LINE2 0 LINE3 0 DIGITAL1 0 DIGITAL2 0 DIGITAL3 0 PHONEIN Phone 0 PHONEOUT 0 VIDEO Video 0 RADIO 0 MONITOR 0 - My modules.conf : alias char-major-14 soundcore alias sound-slot-0 snd-card-0 alias sound sound-slot-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-4 snd-pcm-oss alias sound-service-0-5 snd-pcm-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-0 snd-ens1371 post-install snd-ens1371 modprobe snd-pcm-oss alias sound-slot-1 snd-card-1 alias sound-service-1-0 snd-mixer-oss alias sound-service-1-3 snd-pcm-oss alias sound-service-1-4 snd-pcm-oss alias sound-service-1-5 snd-pcm-oss alias sound-service-1-12 snd-pcm-oss alias snd-card-1 snd-intel8x0 post-install snd-intel8x0 modprobe snd-pcm-oss The sound card #0 is working correctly. How is it possible to play something with this card ? Bye. --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Installation pb, card echoaudio mia, alsa 1.0.0rc1
Hi folks, It's the first time I post here so please bear with me I've read many documents but I'm stuck and I don't know where to investigate further. I try to setup my soundcard echoaudio Mia with alsa. My environement: - Linux Mandrake 9.2 - Kernel 2.4.22 - Alsa 1.0.0-rc1 - Driver EchoAudio 0.8.0b I compiled alsa-driver, alsa-lib, alsa-oss, alsa-utils without any problems. Everything installed fine. I was able to modprobe snd-mia fine as well. The card seemed recognised, cat /proc/asound/cards says: 0 [Mia]: Echoaudio - Mia Mia rev.0 (DSP56361) at 0xec00 irq 20 I used the Emixer tool provided in the EchoAudio package to un-mute the output channels of the card (I think). I try to play a wav file with aplay, and I hear nothing, and I see no signal in the little VU-meter provided in the Emixer tool. When I play the file, I see the following logs in /var/log/messages: Dec 4 22:28:39 localhost kernel: pcmaout_open Dec 4 22:28:39 localhost kernel: AvailCh=2 Dec 4 22:28:39 localhost kernel: CEchoGals::OpenAudio: ch=-121781490 int=-121762112 Dec 4 22:28:39 localhost kernel: SGlist: memory allocated phys=f8bdcb18 Dec 4 22:28:39 localhost kernel: ResetTransport f8bdc205 Dec 4 22:28:39 localhost kernel: ECHOSTATUS_OK Dec 4 22:28:39 localhost kernel: OpenAudio()=0 Dec 4 22:28:39 localhost kernel: pcm_hw_params (bufsize=88200B periods=3 persize=22052B) Dec 4 22:28:39 localhost kernel: pcm_hw_params table size=131072 pages=32 Dec 4 22:28:39 localhost kernel: pcm_hw_params ok Dec 4 22:28:39 localhost kernel: Prepare rate=44100 format=2 channels=2 Dec 4 22:28:39 localhost kernel: CEchoGals::SetAudioSampleRate: to -121780859 Hz Dec 4 22:28:39 localhost kernel: CGina24::QueryAudioSampleRate() Dec 4 22:28:39 localhost kernel: CEchoGals::SetAudioFormat: for pipe -121780398 Dec 4 22:28:39 localhost kernel: CEchoGals::QueryAudioFormat Dec 4 22:28:39 localhost kernel: CDspCommObject::SetAudioFormat [-121782475] = f8be0dc0 Dec 4 22:28:39 localhost kernel: Prepare ok Dec 4 22:28:39 localhost kernel: pcm_trigger start Dec 4 22:28:39 localhost kernel: StartTransport f8bdc065 My modules.conf is as follow: # # -- SOUND # # ALSA alias char-major-116 snd alias snd-card-0 snd-mia # OSS/Free portion alias char-major-14 soundcore alias sound-slot-0 snd-card-0 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-12 snd-pcm-oss My lsmod says: snd-seq-oss30464 0 (unused) snd-seq-midi-event 6016 0 [snd-seq-oss] snd-seq42576 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 5788 0 [snd-seq-oss snd-seq] snd-pcm-oss42756 0 snd-mixer-oss 14288 0 [snd-pcm-oss] snd-mia66436 1 snd-pcm78980 1 [snd-pcm-oss snd-mia] snd-timer 18500 0 [snd-seq snd-pcm] snd-page-alloc 9300 0 [snd-mia snd-pcm] snd44612 0 [snd-seq-oss snd-seq-midi-event snd-seq snd-seq-device snd-pcm-oss snd-mixer-oss snd-mia snd-pcm snd-timer] soundcore 6340 0 [snd] What can I do more to understand what's going wrong ?? Thanks, Eric. --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ALSA 1.0.0rc1 released
Hello, I have a problem when I want to compil alsa-driver 1.0.0rc1 on Linux Redhat 7.0 kernel 2.2.16-22. I obtain this error : mv /usr/src/redhat/BUILD/alsa-driver-1.0.0rc1/include/modules/acore__memalloc.ver.tmp /usr/src/redhat/BUILD/alsa-driver-1.0.0rc1/include/modules/acore__memalloc.ver i386-redhat-linux-gcc -M -D__KERNEL__ -D__isapnp_now__ -DMODULE=1 -I/usr/src/redhat/BUILD/alsa-driver-1.0.0rc1/include -I/lib/modules/2.2.16-22/build/include -O2 -DLINUX -Wall -Wstrict-prototypes -fomit-frame-pointer -Wno-trigraphs -O2 -fno-strict-aliasing -fno-common -pipe -DALSA_BUILD hwdep.c memalloc.c sgbuf.c memory_wrapper.c pcm.c pcm_native.c pcm_lib.c pcm_timer.c pcm_misc.c pcm_memory.c rawmidi.c timer.c sound.c init.c memory.c info.c control.c misc.c device.c wrappers.c sound_oss.c info_oss.c .depend sound.c:352:27: macro devfs_mk_dir requires 4 arguments, but only 1 given make[2]: *** [fastdep] Erreur 1 make[2]: Quitte le répertoire `/usr/src/redhat/BUILD/alsa-driver-1.0.0rc1/acore'make[1]: *** [dep] Erreur 1 make[1]: Quitte le répertoire `/usr/src/redhat/BUILD/alsa-driver-1.0.0rc1' make: *** [include/sndversions.h] Erreur 2 How is it possible to modify this ? Bye. Nicolas Croiset [EMAIL PROTECTED] Tel : +33 4 72 84 06 04 Fax : +33 4 72 84 06 02 --- This SF.net email is sponsored by: SF.net Giveback Program. Does SourceForge.net help you be more productive? Does it help you create better code? SHARE THE LOVE, and help us help YOU! Click Here: http://sourceforge.net/donate/ ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] asoundrc, spdif and nforce on 1.0pre1
Hi,
The question is : How config alsa to have spdif used by xine and mplayer.
My suspicion: I have no controls IEC958 threw amixer controls ???
I just installed last dev release 1.0-pre1 (with alsa-driver-1.0-pre2).
Technical data of my hardware and system configurations are at the end of the
mail.
Compilation and installation worked perfectly.
The only problem I have (and it might not be linked to this release version),
is in configuring my .asoundrc to have spdif work with xine, xmms, or
mplayer.
I expected to run xine and have 5.1 outputs on spdif with divx and the same
with mplayer.
But it is not the case
I tried:
mplayer -ao alsa9:nforce divx.avi
mplayer logs:
---
alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit
(Little-Endian)
alsa-init: soundcard set to nforce
alsa9: 44100 Hz/2 channels/4 bpf/16384 bytes buffer/Signed 16 bit Little
Endian
AO: [alsa9] 44100Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
But it goes to my analog speakers.
The same with xine:
xine divx.avi
logs:
--
main : détection du plugin de sortie audio alsa
audio_alsa_out : supported modes are 8bit mono stereo 4-channel 5-channel
5.1-channel a/52 and DTS pass-through
[...]
audio_alsa_out: Audio Device name = default
audio_alsa_out: Number of channels = 2
I test with aplay:
aplay -D plughw:0,0 test.wav: OK. Goes to my analog speakers
aplay -D plughw:0,2 test.wav: OK. Goes to my 5.1 speakers
I have read maybe every post, and docs available, but I didn't reach a config
that enables spdif with 5.1.
My .asoundrc:
--
pcm.nforce-hw {
type hw
card 0
}
pcm.!default {
type plug
slave.pcm nforce
}
pcm.nforce {
type dmix
ipc_key 1234
slave {
pcm hw:0,0
period_time 0
period_size 1024
buffer_size 4096
rate 44100
}
}
ctl.nforce-hw {
type hw
card 0
}
In fact, I can have results out threw spdif (so with 5.1 speakers) by setting
in .asoundrc the pcm hw:0,2 instead of hw:0.0, what means, I think, the
spdif CAN work.
But even though the sound goes to my 5.1 speakers, the system still is not
spdif apparently:
I say that because of this mplayer's logs, after the command:
mplayer -channels 6 -ao alsa9:nforce divx.avi
alsa-init: requested format: 44100 Hz, 6 channels, Signed 16-bit
(Little-Endian)
alsa-init: soundcard set to nforce
alsa-init: unable to set channels: Invalid argument
Could not open/initialize audio device - no sound.
If I don't specified 6 channels, it works but with 2 channels (that outputs
threw my 5.1 speakers because of my trick on .asoundrc):
alsa-init: requested format: 44100 Hz, 2 channels, Signed 16-bit
(Little-Endian)
alsa-init: soundcard set to nforce
alsa9: 44100 Hz/2 channels/4 bpf/16384 bytes buffer/Signed 16 bit Little
Endian
AO: [alsa9] 44100Hz 2ch Signed 16-bit (Little-Endian) (2 bps)
Building audio filter chain for 44100Hz/2ch/16bit - 44100Hz/2ch/16bit...
xine neither works with 6 channels (but the sound still outputs to 5.1
speakers) :
it detects alsa:
main : détection du plugin de sortie audio alsa
audio_alsa_out : supported modes are 8bit mono stereo 4-channel 5-channel
5.1-channel a/52 and DTS pass-through
[..]
audio_alsa_out: Audio Device name = default
audio_alsa_out: Number of channels = 2
but thinks it is 2 channels.
Moreover, this tricks disabled the volumes controls from xine or xmms, or
alsamixer.
I thought if I didn't have .asoundrc, alsa would work with default
configuration, but I don't know how files from
/usr/share/alsa/cards/NFORCE.conf are involved in setting the configuration.
Yes, you guess I have nforce.
I then terminate with my hardware and system configuration:
*
*
Mandrake 9.1 - Linux 2.4.21-0.13mdk - i686
Athlon XP
MSI K7N420D Pro with GPU and APU (Nforce 1)
lspci -v:
00:05.0 Multimedia audio controller: nVidia Corporation: Unknown device 01b0
(rev c2)
Subsystem: Micro-Star International Co., Ltd.: Unknown device 3730
Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 11
Memory at e300 (32-bit, non-prefetchable) [size=512K]
Capabilities: available only to root
00:06.0 Multimedia audio controller: nVidia Corporation nForce Audio (rev c2)
Subsystem: Micro-Star International Co., Ltd.: Unknown device 3730
Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 12
I/O ports at e000 [size=256]
I/O ports at e400 [size=128]
Memory at e308 (32-bit, non-prefetchable) [size=4K]
Capabilities: available only to root
cat /proc/asound/version:
--
Advanced Linux Sound Architecture Driver Version 1.0.0pre2.
Compiled on Nov 21 2003 for kernel 2.4.21-0.13mdk with versioned symbols.
cat /proc/asound/devices:
--
0: [0- 0]: ctl
18: [0- 2]: digital audio playback
25: [0- 1]:
[Alsa-user] No way to record. Here is my /etc/asound.state
Hello, Config: via82xx (vt8235), microphone plugged on mic in, playing is working, using alsa 0.9.7 with 2.4.20 kernel. Recording is perfectely working on W2K. Playing with alsamixer for 1 year to find the right solution. Help would me much appreciated. Here are my /etc/asound.state file and amixer output.. Thx a lot, Nicolas amixer: Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 22 [71%] [on] Front Right: Playback 22 [71%] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 22 [71%] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined Playback channels: Mono Limits: 0 - 15 Mono: 10 [67%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined Playback channels: Mono Limits: 0 - 15 Mono: 11 [73%] Simple mixer control '3D Control - Switch',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 22 [71%] [on] Front Right: Playback 22 [71%] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 20 [65%] [on] Front Right: Playback 20 [65%] [on] Simple mixer control 'Surround Down Mix',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 27 [87%] [on] Simple mixer control 'Center/LFE Down Mix',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 26 [84%] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 22 [71%] [on] Capture [off] Front Right: Playback 22 [71%] [on] Capture [off] Simple mixer control 'Line-In As Surround',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 22 [71%] [on] Capture [off] Front Right: Playback 22 [71%] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 29 [94%] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic As Center/LFE',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: Mono: Simple mixer control 'Video',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 22 [71%] [on] Capture [off] Front Right: Playback 22 [71%] [on] Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [on] Capture [off] Simple mixer control 'IEC958 Input Monitor',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Output',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined Playback channels: Mono Limits: 0 - 3 Mono: 3 [100%] Simple mixer control 'PC Speaker',0
[Alsa-user] No way to record. Here is my /etc/asound.state
Config: via82xx (vt8235), microphone plugged on mic in, sound playing
is working, using alsa 0.9.1 with 2.4.20 kernel.
/etc/asound.state:
state.8235 {
control.1 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Master Playback Switch'
value true
}
control.2 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Master Playback Volume'
value.0 22
value.1 22
}
control.3 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Center Playback Switch'
value true
}
control.4 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Center Playback Volume'
value 24
}
control.5 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'LFE Playback Switch'
value true
}
control.6 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'LFE Playback Volume'
value 25
}
control.7 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Surround Playback Switch'
value.0 true
value.1 true
}
control.8 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Surround Playback Volume'
value.0 23
value.1 23
}
control.9 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Master Mono Playback Switch'
value true
}
control.10 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Master Mono Playback Volume'
value 24
}
control.11 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'PC Speaker Playback Switch'
value true
}
control.12 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 15'
iface MIXER
name 'PC Speaker Playback Volume'
value 15
}
control.13 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Phone Playback Switch'
value true
}
control.14 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Phone Playback Volume'
value 22
}
control.15 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Mic Playback Switch'
value true
}
control.16 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Mic Playback Volume'
value 22
}
control.17 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Mic Boost (+20dB)'
value true
}
control.18 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'Line Playback Switch'
value true
}
control.19 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'Line Playback Volume'
value.0 25
value.1 25
}
control.20 {
comment.access 'read write'
comment.type BOOLEAN
iface MIXER
name 'CD Playback Switch'
value true
}
control.21 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 31'
iface MIXER
name 'CD Playback Volume'
value.0 22
value.1 22
}
control.22 {
[alsa-user] chips ac97
- Original Message - From: Nicolas Forget To: [EMAIL PROTECTED] Sent: Sunday, April 21, 2002 12:43 PM Subject: [alsa-use] chips ac97 hi everyone, i'm using a msi k7t 266 proII. on this one i have a sound chipset ac97. but i don't know how to use it with alsa... i've found ni th kernel ac97 support but not in alsa. an idea??? thank you for response on my mail because i'm not member of the list.
Re: [Alsa-user] How to upgrade ALSA driver from 0.5.10 to 0.5.12 ?!
Maybe you have not install the kernel sources ? Le Mercredi 28 Novembre 2001 08:11, Joseph Chan a ?crit : Hi, I got an ALSA driver problem in SuSE 7.3. (the default driver version is ALSA 0.5.10) After uncompressing the alsa-driver-0.5.12.tar.bz2 , I run ./configure , and the error occured. The system told me, missfile /usr/src/linux/include/linux/version.h And I can't find the file version.h at that location. How/What can I do ?! Any help is appreciated.. Thanks! Joseph ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] use of aplay
Thank you, it works, now I can play a file on specified lines.
But now, the problem is that I can not play files simultaneously. If I start
two aplay process on different lines the second process wait for the end of
the first one to begin, maybe it is because the soundcard is used by the
first process.
Does it exist an alsa sound wrapper (aserver maybe ?) in order to play
several files in the same time ?
Thanks,
Nicolas
Le Mardi 20 Novembre 2001 17:36, Takashi Iwai a écrit :
At Tue, 20 Nov 2001 16:34:05 +0100,
Nicolas DEVERGE wrote:
Hi,
I'm a newbie in sound and I finally installed my Terratec EWS88MT using
the latest drivers (0.9.0beta9).
It seems to work, when I use aplay with default parameters, I can hear
sound on the lines 1 ans 2 of my external box.
My question is how can I play in same time another sound file on the
other lines (I have 8 output lines on my external box). It seems that
aplay provides a device parameter but I don't understand how it works.
You'd better to define a new pcm device in your ~/.asoundrc file.
For example, to output via channel 6 and 7 (count from channel 0),
define as follows:
pcm.my_output {
type plug
ttable.0.6 1
ttable.1.7 1
slave.pcm {
type hw
card 0
}
}
where my_output can be named as you like.
Now you can play two-channel data through channel 6 and 7 via aplay:
% aplay -D my_output foo.wav
Similarly, when you'd like to output SPDIF on EWS88MT, use channel 8
and 9 (they are hardcoded).
For more precise info, see alsa-lib/doc/asoundrc.txt.
These pcm definitions are necessary also when you use alsa-oss
emulation lib (which is different from oss emulation module).
You can/need to define the pcm device pcm.dsp0 as well as above, so
that output from an OSS application will be routed to channel 6 and
7.
ciao,
Takashi
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