Re: [alsa-devel] nvidia initial sound being skipped
Hi All,
I just want to say I upgraded to kernel 2.6.36 today and still miss the first 0.5seconds or so of the sound output.
Again I have ruled out player - mplayer, aplay and audacious all do it
I have ruled out source format - both flac and the wav decoded version have the same problem.
I have also ruled out physical connection to the AVR, both HDMI and SPDIF exhibit the same problem.
Any ideas?
Cheers Dave
On 16 November 2010 20:57, David Shirley tephra@gmail.com wrote:
Hi Developers,
I am using an nvidia MCP78 to connect to a Denon 1910 AVR via HDMI.
I was using Kernel 2.6.32.21 and noticed that the initial few seconds of my songs was being chopped off.
After seeing http://www.spinics.net/lists/alsa-devel/msg29408.html I upgraded to 2.6.35.8 (which has that fix in it) and the problem was been reduced considerably.
But it is still occurring to some extent.
So to see if the HDMI was the problem I changed it to coax spdif - no change/same problem.
Now I think it could be mplayer, so I try audacious - no change/same problem.
Now I think it could be the flac file/decoding, so I decode the flac to wav and use aplay - no change/same problem.
The ONLY way I can get the first 0.5seconds of the file to play is by using audacious and then MANUALLY rewinding (by dragging the scroll bar thingy) back to the start... then it will play (on either flac or wav)
Co-incidentally when I do this I get the following error message dumped to console: alsa: snd_pcm_prepare failed: Device or resource busy.
I hope someone can help :)
I am happy to get debug or whatever - just let me know what you need!
mythtv@crystal:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0
lspci -vvvv 00:07.0 Audio device: nVidia Corporation MCP72XE/MCP72P/MCP78U/MCP78S High Definition Audio (rev a1) Subsystem: ASUSTeK Computer Inc. M3N72-D Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 11 Region 0: Memory at fe020000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel
Cheers David
David Shirley wrote:
I am using an nvidia MCP78 to connect to a Denon 1910 AVR via HDMI.
I was using Kernel 2.6.32.21 and noticed that the initial few seconds of my songs was being chopped off.
I just want to say I upgraded to kernel 2.6.36 today and still miss the first 0.5seconds or so of the sound output. ... both HDMI and SPDIF exhibit the same problem.
Apparently, your receiver needs even more time to detect that some sound is playing; there's nothing that the computer could do about it.
Try playing some silence before starting playing music. (This requires a software mixer like dmix or PulseAudio.)
Regards, Clemens
Yeah it could be that. How would u setup a dmix to play silence?
Thanks D. On 19/11/2010 7:07 PM, "Clemens Ladisch" clemens@ladisch.de wrote:
David Shirley wrote:
I am using an nvidia MCP78 to connect to a Denon 1910 AVR via HDMI.
I was using Kernel 2.6.32.21 and noticed that the initial few seconds of my songs was being chopped off.
I just want to say I upgraded to kernel 2.6.36 today and still miss the first 0.5seconds or so of the sound output. ... both HDMI and SPDIF exhibit the same problem.
Apparently, your receiver needs even more time to detect that some sound is playing; there's nothing that the computer could do about it.
Try playing some silence before starting playing music. (This requires a software mixer like dmix or PulseAudio.)
Regards, Clemens
David Shirley wrote:
How would u setup a dmix to play silence?
If you haven't create or modified /etc/asound.conf or ~/.asoundrc, dmix should be enabled by default. (If PulseAudio is installed correctly, PA should be enabled by default.)
Try this to play ten seconds of silence:
aplay -d 10 -t raw -f dat /dev/zero
To play continuously, omit the "-d 10".
Regards, Clemens
OK that works!
But I am concerned about the loss of audio quality when using dmix..
Apart from the sample rate mismatch (and therefor resampling, which can be avoided by using the correct rate in asound.rc) is there anything else I should be worried about?
Cheers D.
On 19 November 2010 22:31, Clemens Ladisch clemens@ladisch.de wrote:
David Shirley wrote:
How would u setup a dmix to play silence?
If you haven't create or modified /etc/asound.conf or ~/.asoundrc, dmix should be enabled by default. (If PulseAudio is installed correctly, PA should be enabled by default.)
Try this to play ten seconds of silence:
aplay -d 10 -t raw -f dat /dev/zero
To play continuously, omit the "-d 10".
Regards, Clemens
David Shirley wrote:
But I am concerned about the loss of audio quality when using dmix..
Apart from the sample rate mismatch (and therefor resampling, which can be avoided by using the correct rate in asound.rc) is there anything else I should be worried about?
x + 0 = x
Regards, Clemens
Does that mean that dmix/alsa doesn't introduce or loose any bits when the input rate matches the output rate?
Cheers D.
On 22 November 2010 18:53, Clemens Ladisch clemens@ladisch.de wrote:
David Shirley wrote:
But I am concerned about the loss of audio quality when using dmix..
Apart from the sample rate mismatch (and therefor resampling, which can be avoided by using the correct rate in asound.rc) is there anything else I should be worried about?
x + 0 = x
Regards, Clemens
participants (2)
-
Clemens Ladisch -
David Shirley