Re: [Alsa-user] Analog line in output via IEC958?
Antony Gelberg wrote: I have an onboard NVidia sound card using snd_hda_intel. Full info about my setup is at http://www.alsa-project.org/db/?f=f407c7aceb5abe36b68731ff50fd1d21f272fb98 I can output everything to a HT amp via the IEC958 optical out by means of the .asoundrc that you see in the above link. I would like to plug something into the analogue line in and have it be output via the IEC958 as well. However, it is only output via the analogue line out, not the optical. How can I get this to work? The ALC883 cannot do this alone. You have to run a program to record from the line in and to play the data back to the SPDIF output. arecord -f dat | aplay -D spdif might work, but then you cannot use these devices with other programs. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Can't get amixer to access controls for a specific device on a card
Eliot Blennerhassett li...@audioscience.com wrote: Why do you think that it is not device #0 i.e. the default device? Best run http://www.alsa-project.org/alsa-info.sh to give us the full info. I suspect device #1 is the MIDI uart... Sorry about the delay... had to install lspci and fiddle with some other stuff to get the script to run and it still complained a bit, but it came up with the output below. Asking aplay to use device 1 results in no complaints, so surely it can't be a MIDI device? I've read somewhere that device 0 has four DXS devices, while device 1 has one six-channel playback device. All of this is academic anyway unless I can enable Line in as surround etc. I've run speaker-test in 6-channel mode and tweaked everything I could think of in amixer, but I'm getting nothing out of the other two jacks. Any help greatly appreciated! Thanks, -- Olly ---alsa-info.sh output--- !! !!ALSA Information Script v 0.4.56 !! !!Script ran on: Thu Mar 3 23:28:41 UTC 2005 !!Linux Distribution !!-- !!Kernel Information !!-- Kernel release:2.6.27.9 Operating System: Linux Architecture: i686 Processor: unknown SMP Enabled: No !!ALSA Version !! Driver version: 1.0.17 Library version: Utilities version: 1.0.19 !!Loaded ALSA modules !!--- !!Sound Servers on this system !! Pulseaudio: Installed - Yes () Running - No ESound Daemon: Installed - Yes () Running - No aRts: Installed - Yes () Running - No Jack: Installed - Yes () Running - No No sound servers found. !!Soundcards recognised by ALSA !!- 0 [V8235 ]: VIA8233 - VIA 8235 VIA 8235 with VT1616i at 0xe400, irq 10 !!PCI Soundcards installed in the system !!-- !!Advanced information - PCI Vendor/Device/Susbsystem ID's !! 00:11.5 0401: 1106:3059 (rev 50) Subsystem: 1106:aa01 !!Loaded sound module options !!-- !!AC97 Codec information !!--- --startcollapse-- 0-0/0: ICEnsemble VT1616i PCI Subsys Vendor: 0x1106 PCI Subsys Device: 0xaa01 Capabilities : -headphone out- DAC resolution : 18-bit ADC resolution : 18-bit 3D enhancement : IC Ensemble/KS Waves 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 LDAC SDAC CDAC DSA=0 SPDIF VRA Extended status : SPCV LDAC SDAC CDAC SPDIF=10/11 VRA PCM front DAC: 48000Hz PCM Surr DAC : 48000Hz PCM LFE DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control: Consumer PCM Copyright Category=0x22 Generation=1 Rate=48kHz 0:00 = 6d50 0:02 = 9f1f 0:04 = 9f1f 0:06 = 801f 0:08 = 0:0a = 801e 0:0c = 801f 0:0e = 801f 0:10 = 9f1f 0:12 = 9f1f 0:14 = 9f1f 0:16 = 9f1f 0:18 = 0808 0:1a = 0:1c = 0:1e = 0:20 = 0:22 = 0:24 = 0:26 = 000f 0:28 = 01c5 0:2a = 05f1 0:2c = bb80 0:2e = bb80 0:30 = bb80 0:32 = bb80 0:34 = 0:36 = 9f93 0:38 = 9f9f 0:3a = 2a20 0:3c = 0:3e = 0:40 = 0:42 = 0:44 = 0:46 = 0:48 = 0:4a = 0:4c = 0:4e = 0:50 = 0:52 = 0:54 = 0:56 = 0:58 = 0:5a = 8230 0:5c = 0:5e = 0:60 = 0:62 = 0:64 = 0:66 = 0:68 = 0:6a = 0:6c = 0:6e = 0:70 = 0:72 = 0:74 = 0:76 = 0:78 = 0:7a = 0:7c = 4943 0:7e = 4552 --endcollapse-- !!ALSA Device nodes !!- crw-rw1 root audio116, 7 Mar 3 23:27 /dev/snd/controlC0 crw-rw1 root audio116, 6 Mar 3 23:27 /dev/snd/pcmC0D0c crw-rw1 root audio116, 5 Mar 3 23:28 /dev/snd/pcmC0D0p crw-rw1 root audio116, 4 Mar 3 23:27 /dev/snd/pcmC0D1c crw-rw1 root audio116, 3 Mar 3 23:27 /dev/snd/pcmC0D1p crw-rw1 root audio116, 2 Mar 3 23:27 /dev/snd/timer !!Aplay/Arecord output !! APLAY List of PLAYBACK Hardware Devices card 0: V8235 [VIA 8235], device 0: VIA 8235 [VIA 8235] Subdevices: 2/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 0: V8235 [VIA 8235], device 1: VIA 8235 [VIA 8235] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD /bin/alsa-info.sh: line 632: arecord: not found !!Amixer output !!- !!---Mixer controls for card 0 [V8235] Card hw:0 'V8235'/'VIA 8235 with VT1616i at 0xe400, irq 10' Mixer name: 'ICEnsemble VT1616i' Components:
[Alsa-user] pulseaudio with alsa-plugin a52
Hello list
Since some weeks I test AC3 on top of pulseaudio-0.9.15 from git . This
works mostly well if only one stream is running at a time. After a
second stream is mixed in, the output begins stuttering. This is all
without the new glitch-free mode.
I tried to use glitch-free mode with linux-2.6.28 and realtime
scheduling enabled on alsa-lib 1.0.19 + alsa-plugins-1.0.19 .
But that doesn't work with a52. Pulseaudio gives this:
D: alsa-sink.c: snd_pcm_avail: Broken pipe
D: alsa-sink.c: snd_pcm_avail: Buffer underrun!
I: alsa-sink.c: Starting playback.
I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed
D: alsa-sink.c: Cutting sleep time for the initial iterations by half.
D: alsa-sink.c: snd_pcm_avail: Broken pipe
D: alsa-sink.c: snd_pcm_avail: Buffer underrun!
I: alsa-sink.c: Starting playback.
I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed
and so on forever.
The pcm is setup like this:
pcm.a52 {
type a52
@args.0 {
type integer
default 0
}
}
So the question is, is it planned to get the a52 plugin working with
pulseaudio and glitch-free? Or can I do something other to get it at
least working without glitch-free? Any ideas how this can be improved?
Thanks in advance
Carsten Luedtke
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Re: [Alsa-user] Analog line in output via IEC958?
Clemens Ladisch wrote: Antony Gelberg wrote: I have an onboard NVidia sound card using snd_hda_intel. Full info about my setup is at http://www.alsa-project.org/db/?f=f407c7aceb5abe36b68731ff50fd1d21f272fb98 I can output everything to a HT amp via the IEC958 optical out by means of the .asoundrc that you see in the above link. I would like to plug something into the analogue line in and have it be output via the IEC958 as well. However, it is only output via the analogue line out, not the optical. How can I get this to work? The ALC883 cannot do this alone. You have to run a program to record from the line in and to play the data back to the SPDIF output. arecord -f dat | aplay -D spdif might work, but then you cannot use these devices with other programs. HTH Clemens -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user I am playing a tv program using mplayer in which the sound input is analog and the output is on the hdmi (I have a asus m3n78 pro motherboard where the nvidia hd sound is on the motherboard). You could use audacity to monitor your audio input and send it to the desired output. If you go to edit preferences in audacity you should see a listing of inputs and outputs that mirror the system audio mixer. For my case the input was hw:0,0 (analog line in) and the output was hw:0,3 (hdmi out). I remember seeing spdif as one of the outputs. You just need to make sure the iec958 output switches on the system mixer are checked and the associated levels set correctly (I have to check iec958 2 to get the hdmi output working). -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Analog line in output via IEC958?
On Tue, Mar 17, 2009 at 10:10 AM, Antony Gelberg antony.gelb...@wayforth.com wrote: .asoundrc that you see in the above link. I would like to plug something into the analogue line in and have it be output via the IEC958 as well. However, it is only output via the analogue line out, not the optical. How can I get this to work? Antony, I don't know of a solution for you under linux, but perhaps this will jog the memory of someone who is more knowledgeable about such things... Under Windows, with my nvidia sound, the nvmixer app used to have an option digitize input. I see this option is gone now, so I wonder if nvidia had problems with it and nuked it, or if they just hide it somehow. My desktop setup used to consist of two computers, one monitor and a pair of digital speakers. One of the computers would output analog audio to the second, which would digitize it, mix it with its own audio and output to my digital speakers. I don't recall any delay, but never really paid attention to those details. If there's a switch for it under windows, and it is a function of the chipset rather than the software, is it possible to turn it on under linux? -Gordon -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
