Re: [Alsa-user] sound card
sercan sen wrote: my OS is scientific linux cern 4 (SLC4) and my sound card is intel corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller, module is snd-intel8x0. the problem is that i cannot heard the sample sound when i detect the sound card. i cannot listen anything, too. Probably some mixer controls aren't configured correctly. Please make sure that Master and PCM controls are raised and unmuted. If that doesn't help, please show the output of amixer contents. HTH Clemens - 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] Surround sound via optical spdif out
Hi, I am running a mythtv box based on Mandriva 2007 on a AMD 3200xp (Dual core) on an AM2 chipset. I am attempting to get surround sound audio out to my stereo via an optical s/pidf cable. With some help from kirberich crimsum I adjusted the model= line in my modprobe.conf and applied the patch from http://hg-mirror.alsa-project.org/alsa-kernel?cmd=changeset;node=e51177a60747522a60bc1599aa5b2db22d4770a4;style=gitweb Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. So any more troubleshooting help would be greatly appreciated. Nasa Here comes the system info: uname -a Linux mythtv 2.6.17-5mdv #1 SMP Wed Sep 13 14:32:31 EDT 2006 i686 AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ GNU/Linux ALSA version 1.0.13 /sbin/lsmod|grep '^snd' snd_seq_dummy 3716 0 snd_seq_oss33152 0 snd_seq_midi_event 7072 1 snd_seq_oss snd_seq49488 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 7212 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss40576 0 snd_mixer_oss 16096 1 snd_pcm_oss snd_hda_intel 16440 0 snd_hda_codec 167072 1 snd_hda_intel snd_pcm70116 3 snd_pcm_oss,snd_hda_intel,snd_hda_codec snd_timer 19620 2 snd_seq,snd_pcm snd_page_alloc 8712 2 snd_hda_intel,snd_pcm snd47972 10 snd_seq_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_hda_intel aplay -l List of PLAYBACK Hardware Devices card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 scanpci: ... pci bus 0x cardnum 0x06 function 0x01: vendor 0x10de device 0x0371 nVidia Corporation MCP55 High Definition Audio Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [-9.00dB] [on] Front Right: Playback 25 [81%] [-9.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [3.00dB] [on] Front Right: Playback 25 [81%] [3.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [-9.00dB] [on] Front Right: Playback 25 [81%] [-9.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 25 [81%] [-9.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 25 [81%] [-9.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [3.00dB] [on] Front Right: Playback 25 [81%] [3.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [3.00dB] [on] Front Right: Playback 25 [81%] [3.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [on] Front Right: Capture 0 [0%] [-12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [on] Front Right: Capture 0 [0%] [-12.00dB] [on] Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0:
[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] password
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Re: [Alsa-user] Capturing audio with mencoder and alsa
Carl Weidling wrote: I have an abit av8 MB running slackware 10.1 (just build a new 2.6.18 kernel for it). I am using the built-in VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio controller. All I am trying to do is get stuff from my VCR onto the computer using mencoder (part of the MPlayer suite) and my TV capture card. Now I also built Linux From Scratch on a different partition, using the old OSS sound system, and that works just find. I use the ancient xmix program to select CD as my input and I record sound and video with mencoder. So I guess the TV card is connected to the CD input on your motherboard? I can even play back the files I capture under slackware using mplayer or xine. But, when I capture videos under slackware with alsa, I get no sound. I used: amixer set CD capture, and GUIs like kmix and alsamixer to try to select CD input for sound. When I run alsamixer it shows CD as capture (only after using the amixer command, I haven´t figured out how to actually switch on capture in alsamixer), Press Space when the input is selected. but it always shows volume as zero. When I use kmix it shows CD as input with a high volume, but when I go to switches it only gives me input1 and input2 as input source select. Why can it not be as simple as with xmix and oss? An AC'97 codec has more controls than OSS is aware of. Please show the output of amixer contents. Regards, Clemens - 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] routing audio to rear speakers
My problems is as follows:
I have a soundstorm chip (nforce2) using the intel8x0 driver with asla
1.0.11rc2. Both are build into the kernel. I want a seperate device to
play the audio on the rear speakers, without blocking any other audio
streams. I do not have hw mixing so I have to use dmix. The idea was to
define a new device:
pcm.skype {
typeplug
slave.pcm skypeduplex
}
pcm.skypeduplex {
type asym
playback.pcm nfrear
capture.pcm input
}
Which uses a playback device nfrear. And which routes the the audio to
the rear channels and gives it to a dmix plugin (output) which combines
all the streams.
# skype
pcm.nfrear {
type route
slave.pcm output # :0 here is 1st hardware card (hw:0,0)
slave.channels 6
ttable.0.2 1 #left stereo channel to left rear channel
ttable.1.3 1 #right stereo channel to right rear channel
}
Sadly, this does not work.
When I try to configure xmms to use the skype device I get the following
error message:
ALSA lib pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not
usable
** WARNING **: alsa_setup(): No configuration available for playback: No
such file or directory
But if I set
slave.pcm snd_card
I can play songs on the rear speakers with xmms.
What am I doing wrong? Any ideas?
Here is the rest of my .asoundrc
# Set default sound card
# Useful so that all settings can be changed to a different card here.
pcm.snd_card {
type hw
card 0
device 0
}
# Allow mixing of multiple output streams to this device
pcm.output {
type dmix
ipc_key 1024
ipc_perm 0660 # Sound for everybody in your group!
slave.pcm snd_card
slave {
# This stuff provides some fixes for latency issues.
# buffer_size should be set for your audio chipset.
channels 6
period_time 0
period_size 1024
buffer_size 8192
}
#bindings {
# 0 0
# 1 1
#}
}
# Allow reading from the default device.
# Also known as record or capture.
pcm.input {
type dsnoop
ipc_key 2048
slave.pcm snd_card
#Possible artsd full duplex fix:
slave {
period_time 0
period_size 1024
buffer_size 8192
}
bindings {
0 0
1 1
}
}
# This is what we want as our default device
# a fully duplex (read/write) audio device.
pcm.duplex {
type asym
playback.pcm output
capture.pcm input
}
###
# CONVERSION PLUG #
###
# Setting the default pcm device allows the conversion
# rate to be selected on the fly.
# duplex mode allows any alsa enabled app to read/write
# to the dmix plug (Fixes a problem with wine).
pcm.!default {
type plug
slave.pcm duplex
}
# Apparently this is wrong (breaks mplayer for me opening the device)
#ctl.!default {
# type plug
# slave.pcm snd_card
#}
# AOSS #
# OSS dsp0 device (OSS needs only output support, duplex will break some
stuff)
pcm.dsp0 {
type plug
slave.pcm output
}
# OSS control for dsp0 (needed?...this might not be useful)
ctl.dsp0 {
type plug
slave.pcm snd_card
}
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[Alsa-user] Fortissimo IV works great with 44.1khz material, not great at all in 48khz
Hi, I make my Fortissimo IV running by passing the model=ms300 option to the snd-ice1724 module ( https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1556 ). It works flawlessly on 44.1khz material, but horribly on 48khz, so *i can't watch movies for now* :(.. I don't exactly know how to describe it, it's too fast, and at the same time choppy on some players such as xine. I tryed everything in xine and alsa mixer options, nothing helps, and also GStreamer and Mplayer do the same. Something weird about MPlayer is that it resamples even 44.1khz sound to 48khz : == Opening audio decoder: [mp3lib] MPEG layer-2, layer-3 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3) == alsa-init: using device default alsa: 48000 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) ..and plays ok if I pass -af resample=44100:0:0 as option. What am I doing wrong ? Is there a workaround ? And.. will you ever support Fortissimo IV ? Cheers! - 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] mic boost (+20db) on snd_hda_intel?
Hi there, my sound card is a mobile's Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) device. I'd like to run skype, which is all fine, if not the mic input would be to low. Usually under Win I am using the mic boost which appears when I run other sound cards with alsa. But I cannot find a related switch for the snd_hda_intel driver module? Is it an error, or just not implemented yet? thanks for your support, Stefan K. 8^) - 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] MIDI (strange problem - Plays only one note)
I have some strange problem with midi. I read all gentoo.org documentation, and all wiki.gentoo documentation, and no success. I have Sound Blaster Live 5.1, emu10k1. A have kernel 2.6.17-r8, and I (think) checked all midi and sound , alsa options for compile. A installed asfxload , and when I load SF2 sound font file, and when I start some program for playing midi, and choose some midi file (kmid for example), only I hear is ONE NOTE (I think it is piano). It does not mather what mid file I play, any midi when I try to play, I hear only one note, and after that one note, player still playing that file. I load sonundfont with command asfxload something.sf2. So, here is some of my configuration and other information [EMAIL PROTECTED] /proc/asound $ cat /proc/asound/devices 0: [ 0] : control 1:: sequencer 4: [ 0- 0]: hardware dependent 6: [ 0- 2]: hardware dependent 8: [ 0- 0]: raw midi 9: [ 0- 1]: raw midi 10: [ 0- 2]: raw midi 16: [ 0- 0]: digital audio playback 18: [ 0- 2]: digital audio playback 19: [ 0- 3]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 26: [ 0- 2]: digital audio capture 33:: timer [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.11rc4 (Wed Mar 22 10:27:24 2006 UTC). [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ aplaymidi -l PortClient name Port name 14:0Midi Through Midi Through Port-0 16:0EMU10K1 MPU-401 (UART) EMU10K1 MPU-401 (UART) 17:0Emu10k1 WaveTableEmu10k1 Port 0 17:1Emu10k1 WaveTableEmu10k1 Port 1 17:2Emu10k1 WaveTableEmu10k1 Port 2 17:3Emu10k1 WaveTableEmu10k1 Port 3 [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.11rc4 emulation code) Kernel: Linux euforia 2.6.17-gentoo-r8 #6 SMP Thu Oct 5 10:10:49 CEST 2006 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: SB Live 5.1 [SB0220] (rev.10, serial:0x80651102) at 0xd400, irq 169 Audio devices: 0: ADC Capture/Standard PCM Playback (DUPLEX) Synth devices: 0: Emu10k1 Midi devices: 0: EMU10K1 MPU-401 (UART) Timers: 7: system timer Mixers: 0: eMicro EM28028 [EMAIL PROTECTED] /proc/asound $ here is output of cat /proc/asound/card0/wavetableD1 localhost bin # cat /proc/asound/card0/wavetableD1 Device: Emu10k1 Ports: 4 Addresses: 17:0 17:1 17:2 17:3 Use Counter: 0 Max Voices: 64 Allocated Voices: 0 Memory Size: 134217728 Memory Available: 130191678 Allocated Blocks: 489 SoundFonts: 1 Instruments: 3560 Samples: 488 Locked Instruments: 3560 Locked Samples: 488 localhost bin # And with aplaymidi, I still here just one note for any file. I dont know what is happening with that... - 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
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
Lee Revell wrote: arecord -f dat x.wav is silent arecord -r 441000 x.wav is not broken but distorted more then 8 bit quantization. I guess you mean 44100? Anyway that last command will record 8 bit audio at 44100Hz which still won't sound good. Try arecord -r 44100 -f S16_LE maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav Overe here mono works OK but stereo gives silence. What is wrong/missing/needs configuration to make recording (from S/PDIF in my case) work? (M-Audio DiO2496) - 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] symbol versioning issue in alsa-lib 1.0.13
Hi,I'm working on a set-top-box running mips, using linux 2.6.15, with alsa driver enabled v1.0.10rc3, and uclibc.I cross-compiled alsa-lib and alsa-utils, and got into issues with symbol versioning enabled : If I let symbol versioning enabled, aplay will fail to get hw/sw params correctly. But if I compile without symbol versioning, it works great. Does anyone know why such symbol versioning is used, and whether it is useful or not to keep it ? Maybe symbol versioning is not supported by uclibc, or is there a bug in the alsa-lib ?Thanks,--Yoann - 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
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
On Fri, 2006-10-13 at 18:15 +0200, Udo van den Heuvel wrote: Lee Revell wrote: arecord -f dat x.wav is silent arecord -r 441000 x.wav is not broken but distorted more then 8 bit quantization. I guess you mean 44100? Anyway that last command will record 8 bit audio at 44100Hz which still won't sound good. Try arecord -r 44100 -f S16_LE maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav Overe here mono works OK but stereo gives silence. What is wrong/missing/needs configuration to make recording (from S/PDIF in my case) work? (M-Audio DiO2496) No idea. Must be a driver bug. Please file a report in ALSA bug tracker. Lee - 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
Re: [Alsa-user] MIDI (strange problem - Plays only one note)
Evighetens Mørke wrote, On 2006-10-14 01:30: I have some strange problem with midi. I read all gentoo.org http://gentoo.org documentation, and all wiki.gentoo documentation, and no success. I have Sound Blaster Live 5.1, emu10k1. A have kernel 2.6.17-r8, and I (think) checked all midi and sound , alsa options for compile. A installed asfxload , and when I load SF2 sound font file, and when I start some program for playing midi, and choose some midi file (kmid for example), only I hear is ONE NOTE (I think it is piano). It does not mather what mid file I play, any midi when I try to play, I hear only one note, and after that one note, player still playing that file. I load sonundfont with command asfxload something.sf2. So, here is some of my configuration and other information [EMAIL PROTECTED] /proc/asound $ cat /proc/asound/devices 0: [ 0] : control 1: : sequencer 4: [ 0- 0]: hardware dependent 6: [ 0- 2]: hardware dependent 8: [ 0- 0]: raw midi 9: [ 0- 1]: raw midi 10: [ 0- 2]: raw midi 16: [ 0- 0]: digital audio playback 18: [ 0- 2]: digital audio playback 19: [ 0- 3]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 26: [ 0- 2]: digital audio capture 33: : timer [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.11rc4 (Wed Mar 22 10:27:24 2006 UTC). [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ aplaymidi -l Port Client name Port name 14:0 Midi Through Midi Through Port-0 16:0 EMU10K1 MPU-401 (UART) EMU10K1 MPU-401 (UART) 17:0 Emu10k1 WaveTable Emu10k1 Port 0 17:1 Emu10k1 WaveTable Emu10k1 Port 1 17:2 Emu10k1 WaveTable Emu10k1 Port 2 17:3 Emu10k1 WaveTable Emu10k1 Port 3 [EMAIL PROTECTED] /proc/asound $ [EMAIL PROTECTED] /proc/asound $ cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.11rc4 emulation code) Kernel: Linux euforia 2.6.17-gentoo-r8 #6 SMP Thu Oct 5 10:10:49 CEST 2006 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: SB Live 5.1 [SB0220] (rev.10, serial:0x80651102) at 0xd400, irq 169 ^^^ Audio devices: 0: ADC Capture/Standard PCM Playback (DUPLEX) Synth devices: 0: Emu10k1 Midi devices: 0: EMU10K1 MPU-401 (UART) Timers: 7: system timer Mixers: 0: eMicro EM28028 [EMAIL PROTECTED] /proc/asound $ I don't like the look of your IRQ setting. Could this be a hardware problem? My output looks like: # cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.13 emulation code) Kernel: Linux victoria 2.6.18 #3 SMP PREEMPT Sun Oct 1 05:32:21 CST 2006 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: SB Live 5.1 [SB0220] (rev.10, serial:0x80651102) at 0xe400, irq 12 Audio devices: 0: ADC Capture/Standard PCM Playback (DUPLEX) Synth devices: 0: Emu10k1 Midi devices: 0: EMU10K1 MPU-401 (UART) Timers: 7: system timer Mixers: 0: eMicro EM28028 here is output of cat /proc/asound/card0/wavetableD1 localhost bin # cat /proc/asound/card0/wavetableD1 Device: Emu10k1 Ports: 4 Addresses: 17:0 17:1 17:2 17:3 Use Counter: 0 Max Voices: 64 Allocated Voices: 0 Memory Size: 134217728 Memory Available: 130191678 Allocated Blocks: 489 SoundFonts: 1 Instruments: 3560 Samples: 488 Locked Instruments: 3560 Locked Samples: 488 localhost bin # # cat /proc/asound/card0/wavetableD1 Device: Emu10k1 Ports: 4 Addresses: 17:0 17:1 17:2 17:3 Use Counter: 0 Max Voices: 64 Allocated Voices: 0 Memory Size: 134217728 Memory Available: 126801544 Allocated Blocks: 527 SoundFonts: 1 Instruments: 1849 Samples: 526 Locked Instruments: 1849 Locked Samples: 526 I have the same model sound card (SB0220), and have it working fine. Could it be that you have a corrupt soundfont? # md5sum /usr/share/sounds/sf2/*.sf2 568ddfaa56db2bb45fc96e28dcc711ad /usr/share/sounds/sf2/8mbgmsfx.sf2 # ls -al /usr/share/sounds/sf2/*.sf2 -rwxr-xr-x 1 root root 7557598 2005-12-26 13:03 /usr/share/sounds/sf2/8mbgmsfx.sf2 For completeness, these are the snd* modules loaded: # lsmod|grep snd snd_rtctimer3500 0 snd_emu10k1_synth 7904 0 snd_emux_synth 33664 1 snd_emu10k1_synth snd_seq_virmidi 7264 1 snd_emux_synth snd_seq_midi_emul 6080 1 snd_emux_synth snd_seq_dummy 3876 0 snd_seq_oss30688 0 snd_seq_midi9184 0 snd_seq_midi_event 6976 3 snd_seq_virmidi,snd_seq_oss,snd_seq_midi snd_seq50416 9 snd_emux_synth,snd_seq_virmidi,snd_seq_midi_emul,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_emu10k1 108576 2 snd_emu10k1_synth snd_rawmidi23712 3 snd_seq_virmidi,snd_seq_midi,snd_emu10k1 snd_ac97_codec 88804 1 snd_emu10k1 snd_ac97_bus2432 1 snd_ac97_codec snd_pcm_oss
Re: [Alsa-user] Problem with an ice1712 soundcard on recording
Lee Revell wrote: On Fri, 2006-10-13 at 18:15 +0200, Udo van den Heuvel wrote: Lee Revell wrote: arecord -f dat x.wav is silent [] Try arecord -r 44100 -f S16_LE maybe add -c2 to record stereo: arecord -r 44100 -f S16_LE -c2 blah.wav Overe here mono works OK but stereo gives silence. What is wrong/missing/needs configuration to make recording (from S/PDIF in my case) work? (M-Audio DiO2496) No idea. Must be a driver bug. Please file a report in ALSA bug tracker. Thank, done that, I hope the fix is easy. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2526 Udo - 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] hda intal mic
hi, i googled but i can't find any useful information about mic and hda-intel drivers.I wuold like to know if it's possible to get it work or if this feature is not yet implemented. I can hear sound and alsamixer tell me that i have a capture interface working, but i still can't use a mic. thanks cirpo - 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] Re : Fortissimo IV works great wi th 44.1khz material, not great at all in 48khz
Le 13.10.2006 18:56:53, Sergei Steshenko a écrit : On Fri, 13 Oct 2006 09:59:32 -0400 Lee Revell [EMAIL PROTECTED] wrote: Maybe you need to use alsamixer to set the clock rate to 48000? Lee FWIW - I do need to set clock in 'alsamixer' to 32000Hz when I use 'openwengo' VOIP application - with 44100Hz the symptoms are similar to the ones described by Elie Morisse. So, changing sample rate is definitely worth trying. --Sergei. -- Visit my http://appsfromscratch.berlios.de/ open source project. What are your exact settings ? I played with Multi Track switches and until now none of combinations gives me accurate sound.. What is Analog Bypass btw ? Another weirdness : I found out that setting Internal Clock to 44100 and Rate Locking to true makes 48khz material sound better ( though it's not the cure-all, the sound pops nastily ) - 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
Re: [Alsa-user] Capturing audio with mencoder and alsa
Clemens Ladisch responded to my request for help in gettingmencoder to record sound:Me: I have an abit av8 MB running slackware 10.1 (just build a new 2.6.18 kernel for it). I am using the built-in VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio controller.I thank him for his interest. He askedfor more clarificationLadisch:So I guess the TV card is connected to the CD input on your motherboard?Yes. I have a choice of CD input or AUX2. I used CD for what you might call historical reasons. I could also use an audio cablewrapped around from the external audio out of the TV card to the external input of sound, but I use LINE-IN on the audio forother things.Me: I haven?t figured out how to actually switch on capture in alsamixer),Ladisch:Press Space when the input is selected. I have tried that. Thinking about it overnight, I suspect thatsomehow the driver in alsa cannot detect the CD as input.It´s as though it knows it´s there, but doesn´t believe it canbe used as input, so when one issues the command, it refuses it. When I bring up alsamixer, and use F4 to look at CAPTURE,it shows CD as capture but with zero volume, and it will notincrease the volume. There is a whole separate column,labeled capture, and there I can increase the volume. Please show the output of amixer contents.Regards,ClemensSee the attached, and thanks again -Carl amixer.contents.output Description: Binary data - 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
Re: [Alsa-user] Re : Fortissimo IV works great wit h 44.1khz material, not great at all in 48khz
On Fri, 13 Oct 2006 21:04:13 +0400 Elie Morisse [EMAIL PROTECTED] wrote: Le 13.10.2006 18:56:53, Sergei Steshenko a écrit : On Fri, 13 Oct 2006 09:59:32 -0400 Lee Revell [EMAIL PROTECTED] wrote: Maybe you need to use alsamixer to set the clock rate to 48000? Lee FWIW - I do need to set clock in 'alsamixer' to 32000Hz when I use 'openwengo' VOIP application - with 44100Hz the symptoms are similar to the ones described by Elie Morisse. So, changing sample rate is definitely worth trying. --Sergei. -- Visit my http://appsfromscratch.berlios.de/ open source project. What are your exact settings ? I played with Multi Track switches and until now none of combinations gives me accurate sound.. What is Analog Bypass btw ? Another weirdness : I found out that setting Internal Clock to 44100 and Rate Locking to true makes 48khz material sound better ( though it's not the cure-all, the sound pops nastily ) - 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 My reply was meant as a confirmation to try another sample rate. I have a different than you soundcard (M-Audio Revolution 7.1) and I haven't yet used it for real sound recording - just verified once that capture worked in mono mode at 8kHz sample rate. So, my settings are most likely irrelevant - I do not use digital output of any kind, my output device is stereo headphones. The 32000Hz sample rate for 'openwengo' makes sense because the codec uses that sample rate and apparently the application lacks resampler. --Sergei. -- Visit my http://appsfromscratch.berlios.de/ open source project. - 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
Re: [Alsa-user] symbol versioning issue in alsa-lib 1.0.13
On Fri, 13 Oct 2006, Yoann WALTHER wrote: Hi, I'm working on a set-top-box running mips, using linux 2.6.15, with alsa driver enabled v1.0.10rc3, and uclibc. I cross-compiled alsa-lib and alsa-utils, and got into issues with symbol versioning enabled : If I let symbol versioning enabled, aplay will fail to get hw/sw params correctly. But if I compile without symbol versioning, it works great. Does anyone know why such symbol versioning is used, and whether it is useful or not to keep it ? Maybe symbol versioning is not supported by uclibc, or is there a bug in the alsa-lib ? If you compile all applications against one alsa-lib version I suggest to disable versioning. It's only useful to run old (ALSA 0.9) binaries with newer versions of alsa-lib. I guess, that your runtime linker (ld.so) does not care about symbol versions. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SUSE Labs - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote: Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. SPDIF only supports 2 channels unless the signal is AC3-encoded. Thanks for the reply Lee, Does that mean I should do something like this: speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav where the wav file is a ac3 encoded file? Nasa Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote: Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. SPDIF only supports 2 channels unless the signal is AC3-encoded. Thanks for the reply Lee, Does that mean I should do something like this: speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav where the wav file is a ac3 encoded file? I don't know if speaker-test will play AC3 encoded files. Try xine or mplayer. Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 16:36 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote: Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. SPDIF only supports 2 channels unless the signal is AC3-encoded. Thanks for the reply Lee, Does that mean I should do something like this: speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav where the wav file is a ac3 encoded file? I don't know if speaker-test will play AC3 encoded files. Try xine or mplayer. Damn, No love at all Sound from Left and right speakers, but nothing else. How do I troubleshoot this? Nasa Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 16:52 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 16:36 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote: Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. SPDIF only supports 2 channels unless the signal is AC3-encoded. Thanks for the reply Lee, Does that mean I should do something like this: speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav where the wav file is a ac3 encoded file? I don't know if speaker-test will play AC3 encoded files. Try xine or mplayer. Damn, No love at all Sound from Left and right speakers, but nothing else. How do I troubleshoot this? Does the receiver indicate an AC3 signal rather than standard PCM? IIRC AC3 passthrough might not work on nvidia chipsets due to a proprietary hardware implementation, but it might have changed. Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 17:05 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 16:52 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 16:36 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 16:34 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 09:56 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 04:10 -0400, Nathan A. Smith wrote: Unfortunately, I still am unable to get surround sound to work. The way we were testing this is with speaker-test -Dplug:iec958 -c6 (note: iec958 and spdif are the same for this motherboard). I am getting sound from my left and right speakers, but nothing from anything else. SPDIF only supports 2 channels unless the signal is AC3-encoded. Thanks for the reply Lee, Does that mean I should do something like this: speaker-test -Dplug:iec958 -c6 -w SURROUNDTEST_011212.wav where the wav file is a ac3 encoded file? I don't know if speaker-test will play AC3 encoded files. Try xine or mplayer. Damn, No love at all Sound from Left and right speakers, but nothing else. How do I troubleshoot this? Does the receiver indicate an AC3 signal rather than standard PCM? No, nothing visibile on my receiver. IIRC AC3 passthrough might not work on nvidia chipsets due to a proprietary hardware implementation, but it might have changed. Anywhere I could look to see if that's true or not? (hopefully it's not but I would like to check). Nasa Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote: Does the receiver indicate an AC3 signal rather than standard PCM? No, nothing visibile on my receiver. Do you mean the receiver has no way to display whether it's receiving PCM or AC3, or that the indicator is not active? Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 17:57 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote: Does the receiver indicate an AC3 signal rather than standard PCM? No, nothing visibile on my receiver. Do you mean the receiver has no way to display whether it's receiving PCM or AC3, or that the indicator is not active? The receiver I have is: http://reviews.cnet.com/Pioneer_VSX_816_K/4505-6466_7-31848960.html I don't have the instructions right now, as I am still moving into my new place. It's surely in one of these boxes Nasa Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
just to chime in ... I have a similar pioneer look on your display, where the little digital / analog display box is. when receiving a encoded pcm bitstream, the display will show: DOLBY DIGITAL or DIGITAL DTS depending on the encoding used... when a normal non-encoded pcm is received the indicator will just read DIGITAL gordon On 10/13/06, Nathan A. Smith [EMAIL PROTECTED] wrote: On Fri, 2006-10-13 at 17:57 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote: Does the receiver indicate an AC3 signal rather than standard PCM? No, nothing visibile on my receiver. Do you mean the receiver has no way to display whether it's receiving PCM or AC3, or that the indicator is not active? - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 18:50 -0400, Gordon McLellan wrote:
just to chime in ... I have a similar pioneer
look on your display, where the little digital / analog display box
is. when receiving a encoded pcm bitstream, the display will show:
DOLBY
DIGITAL
or
DIGITAL
DTS
depending on the encoding used... when a normal non-encoded pcm is
received the indicator will just read DIGITAL
Thanks Gordon...
I am not getting AC3 info to my receiver... I have never seen either of
those on my display :{
Nasa
gordon
On 10/13/06, Nathan A. Smith [EMAIL PROTECTED] wrote:
On Fri, 2006-10-13 at 17:57 -0400, Lee Revell wrote:
On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote:
Does the receiver indicate an AC3 signal rather than standard PCM?
No, nothing visibile on my receiver.
Do you mean the receiver has no way to display whether it's receiving
PCM or AC3, or that the indicator is not active?
-
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
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-
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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
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Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 19:05 -0400, Nathan A. Smith wrote:
On Fri, 2006-10-13 at 18:50 -0400, Gordon McLellan wrote:
just to chime in ... I have a similar pioneer
look on your display, where the little digital / analog display box
is. when receiving a encoded pcm bitstream, the display will show:
DOLBY
DIGITAL
or
DIGITAL
DTS
depending on the encoding used... when a normal non-encoded pcm is
received the indicator will just read DIGITAL
Thanks Gordon...
I am not getting AC3 info to my receiver... I have never seen either of
those on my display :{
Nasa
BTW: using files from
http://www.sr.se/cgi-bin/mall/index.asp?programid=2445
I ran aplay -v ... and got the following,
aplay -v SURROUNDTEST_DD_640.wav
Playing WAVE 'SURROUNDTEST_DD_640.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)
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 : 15052
period_size : 940
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 940
xfer_align : 940
start_threshold : 15040
stop_threshold : 15052
silence_threshold: 0
silence_size : 0
boundary : 986447872
Slave: Soft volume PCM
Control: PCM Playback Volume
min_dB: -51
resolution: 256
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Direct Stream Mixing PCM
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 1
stop_threshold : 1073741824
silence_threshold: 0
silence_size : 1073741824
boundary : 1073741824
From looking at this aplay played this files as stereo (2 channel) --
which doesn't seem right. Why would it do that?
It has a slightly different output when I do the following:
aplay -v -Dplug:iec958 SURROUNDTEST_011212.wav
Playing WAVE 'SURROUNDTEST_011212.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Stereo
Plug PCM: Hooks PCM
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 : 16384
period_size : 4096
period_time : 92879
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Hardware PCM card 0 'HDA NVidia' device 1 subdevice 0
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 : 16384
period_size : 4096
period_time : 92879
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 107374182
notice the different rates
Any ideas?
Nasa
gordon
On 10/13/06, Nathan A. Smith [EMAIL PROTECTED] wrote:
On Fri, 2006-10-13 at 17:57 -0400, Lee Revell wrote:
On Fri, 2006-10-13 at 17:39 -0400, Nathan A. Smith wrote:
Does the receiver indicate an AC3 signal rather than standard PCM?
No, nothing visibile
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 21:48 -0400, Nathan A. Smith wrote:
On Fri, 2006-10-13 at 19:05 -0400, Nathan A. Smith wrote:
On Fri, 2006-10-13 at 18:50 -0400, Gordon McLellan wrote:
just to chime in ... I have a similar pioneer
look on your display, where the little digital / analog display box
is. when receiving a encoded pcm bitstream, the display will show:
DOLBY
DIGITAL
or
DIGITAL
DTS
depending on the encoding used... when a normal non-encoded pcm is
received the indicator will just read DIGITAL
Thanks Gordon...
I am not getting AC3 info to my receiver... I have never seen either of
those on my display :{
Nasa
BTW: using files from
http://www.sr.se/cgi-bin/mall/index.asp?programid=2445
I ran aplay -v ... and got the following,
aplay -v SURROUNDTEST_DD_640.wav
Playing WAVE 'SURROUNDTEST_DD_640.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Stereo
Plug PCM: Rate conversion PCM (48000, sformat=S16_LE)
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 : 15052
period_size : 940
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 940
xfer_align : 940
start_threshold : 15040
stop_threshold : 15052
silence_threshold: 0
silence_size : 0
boundary : 986447872
Slave: Soft volume PCM
Control: PCM Playback Volume
min_dB: -51
resolution: 256
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Direct Stream Mixing PCM
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 0
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : MMAP_INTERLEAVED
format : S16_LE
subformat: STD
channels : 2
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 16384
period_size : 1024
period_time : 21333
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 1024
xfer_align : 1024
start_threshold : 1
stop_threshold : 1073741824
silence_threshold: 0
silence_size : 1073741824
boundary : 1073741824
From looking at this aplay played this files as stereo (2 channel) --
which doesn't seem right. Why would it do that?
It has a slightly different output when I do the following:
aplay -v -Dplug:iec958 SURROUNDTEST_011212.wav
Playing WAVE 'SURROUNDTEST_011212.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Stereo
Plug PCM: Hooks PCM
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 : 16384
period_size : 4096
period_time : 92879
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Hardware PCM card 0 'HDA NVidia' device 1 subdevice 0
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 : 16384
period_size : 4096
period_time : 92879
tick_time: 4000
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 4096
xfer_align : 4096
start_threshold : 16384
stop_threshold : 16384
silence_threshold: 0
silence_size : 0
boundary : 107374182
notice the different rates
Any ideas?
Nasa
Forgot to note: each time I
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 21:48 -0400, Nathan A. Smith wrote: BTW: using files from http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 I ran aplay -v ... and got the following, aplay -v SURROUNDTEST_DD_640.wav Are you still testing AC3/DTS/etc passthrough? aplay won't play those files. What happens if you test AC3 passthrough using xine or mplayer as described in this article: http://linuxgazette.net/118/knaggs.html For example: mplayer -ao alsa -ac hwac3 filename.ac3 Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 22:06 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 21:48 -0400, Nathan A. Smith wrote: BTW: using files from http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 I ran aplay -v ... and got the following, aplay -v SURROUNDTEST_DD_640.wav Are you still testing AC3/DTS/etc passthrough? aplay won't play those files. What happens if you test AC3 passthrough using xine or mplayer as described in this article: http://linuxgazette.net/118/knaggs.html For example: mplayer -ao alsa -ac hwac3 filename.ac3 Lee This is what happens... mplayer -ao alsa -ac hwac3 SURROUNDTEST_011212.wav MPlayer 1.0pre8-4.1.1 (C) 2000-2006 MPlayer Team CPU: AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ (Family: 15, Model: 75, Stepping: 2) CPUflags: MMX: 1 MMX2: 1 3DNow: 1 3DNow2: 1 SSE: 1 SSE2: 1 Compiled with runtime CPU detection. 93 audio 211 video codecs Opening joystick device /dev/input/js0 Setting up LIRC support... mplayer: could not connect to socket mplayer: Connection refused Failed to open LIRC support. You will not be able to use your remote control. Playing SURROUNDTEST_011212.wav. Audio file file format detected. == Forced audio codec: hwac3 Cannot find codec for audio format 0x2001. Read DOCS/HTML/en/codecs.html! == Segmentation fault remember: the test file is from http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 which I found via a link out of an ALSA HOWTO Nasa - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 22:16 -0400, Nathan A. Smith wrote: Playing SURROUNDTEST_011212.wav. Audio file file format detected. == Forced audio codec: hwac3 Cannot find codec for audio format 0x2001. Read DOCS/HTML/en/codecs.html! == Segmentation fault remember: the test file is from http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 which I found via a link out of an ALSA HOWTO I think that's a DTS/DD file. Try this one: http://downloads.lightspeed.cx/lynne/bjorn_lynne-secret_world_(surround_version).ac3 Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 22:39 -0400, Lee Revell wrote: On Fri, 2006-10-13 at 22:16 -0400, Nathan A. Smith wrote: Playing SURROUNDTEST_011212.wav. Audio file file format detected. == Forced audio codec: hwac3 Cannot find codec for audio format 0x2001. Read DOCS/HTML/en/codecs.html! == Segmentation fault remember: the test file is from http://www.sr.se/cgi-bin/mall/index.asp?programid=2445 which I found via a link out of an ALSA HOWTO I think that's a DTS/DD file. Try this one: http://downloads.lightspeed.cx/lynne/bjorn_lynne-secret_world_(surround_version).ac3 First off Nice song -- I might have to look for more of him/her. Which points out that the file actually worked (and it does sound really nice). So I must ask -- DD/DTS/AC3 are all forms of encoding surround sound. Is it typical that only AC3 works on our systems? Is AC3 the DVD standard (I didn't think so...)? Nasa BTW: THANKS Lee, I appreciate your efforts and patience. Lee - 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
Re: [Alsa-user] Surround sound via optical spdif out
On Fri, 2006-10-13 at 22:52 -0400, Nathan A. Smith wrote: On Fri, 2006-10-13 at 22:39 -0400, Lee Revell wrote: I think that's a DTS/DD file. Try this one: http://downloads.lightspeed.cx/lynne/bjorn_lynne-secret_world_(surround_version).ac3 First off Nice song -- I might have to look for more of him/her. Which points out that the file actually worked (and it does sound really nice). So I must ask -- DD/DTS/AC3 are all forms of encoding surround sound. Is it typical that only AC3 works on our systems? Is AC3 the DVD standard (I didn't think so...)? I'm not really a home theater expert, but the DVD standard is AC3 which is also known as DD. DTS is an older, less common format. So you should be able to play DVDs in surround. I'm not sure why the other file didn't work. It looks like mplayer tried and failed to guess the file type? I *think* that if AC3 passthrough works then DTS must work - it's just a matter of setting a non-audio bit on the stream. Lee - 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
