You know what? I give up. I got the one second of audio lag again today. There was nothing in my dmesg output as to why. At this point, I think I've exhausted all the options.
No worries; my new setup will be video over HDMI, audio from a Behringer UCA 202 USB sound card directly to speakers. It just makes it a bit more complicated if I want to hook other devices up to the TV. EDIT: Hmm... the problem went away soon after I posted, suggesting that it's temporary. Still not something that I'm willing to deal with. |
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A year or two ago I spent a good amount of time trying to get HDMI audio to work after reading the documentation from NVIDIA and elsewhere and finally threw in the towel. :scratch: Kudos for all the effort you put into the project. :hattip: |
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https://projects.archlinux.org/svnto...ulseaudio-alsa |
Should be. To be honest, I can't remember to even have a .soundrc on Slackware with Pulseaudio and I definitely don't have one now on Gentoo.
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Got help from this thread. SOLVED] Proper pulseaudio setup for Slackware64-current They were:
Install alsa-plugins as well as pulseaudio Add start-pulseaudio-x11 to ~/.config/openbox/autostart (because I'm using openbox) Add to /etc/rc.d/rc.local: Code:
if [ -x /etc/rc.d/rc.pulseaudio ]; then Add the following ~/.asoundrc Code:
pcm.pulse { UPDATE: Hate to turn this into a Pulseaudio thread, but ugh... it turns out (ref) I need both a compat32 pulseaudio and a compat32 alsa-plugins to get audio in Wine now. I'll take care of that later. UPDATE 2: Runner2 crashed immediately for what I strongly suspect is the same reason: Code:
error 60: Error initializing output device. |
Tobi, were you able to get Bit Trip Runner2 working with Pulseaudio? I got this problem, and I can't fix it:
http://steamcommunity.com/app/218060...8810978055856/ Everything else I've tried (including other Steam games) works. |
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You can then use the following .asoundrc to set device 0:7 as the default: Code:
pcm.!default { The simplest .asoundrc file that will send audio to dmix and then to 0:7 is: Code:
pcm.ossmix { Code:
pcm.ossmix { |
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XRUN_DEBUG is turned on, of course. OOOOOh, I got the audio lag again (with the above setup, no Pulseaudio), and this time I had stuff in my DMESG! Code:
[ 2210.847854] ALSA: PCM: Lost interrupts? [Q] (stream=0, delta=422, new_hw_ptr=12276, old_hw_ptr=11854) If I started playing Google Play Music in Chromium, then my dmesg would be spammed with the above messages. I was able to get the messages to disappear by adding the following to /etc/modprobe.d/snd-hda-intel.conf: Code:
options snd-hda-intel enable_msi=1 bdl_pos_adj=64 We'll see if this solves the delayed audio problem. I'm hopeful, because it gets rid of the "lost interrupt" warnings, and lost interrupts are obviously consistent with the symptoms. |
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I'm wondering where the idea above came from, or if it was ever true, what and when has it changed? I ask this because I've never worried about this issue, because as long as I can remember, it has never been an issue for me with Alsa. I don't even have pulseaudio installed but I have always been able to, for example, speak and hear on TeamSpeak 3 while at the same time hearing for example, Professor Putricide. declare "Hmmmm where did those come from?" in WOTLK. Upon a few occasions I have added XMMS2 or Aqualung, or even the typical flash players in browsers so that the raid can hear music while we hurl insults at Valanar. I'm assuming this is what people mean by "multiple sound (sources) at the same time" and mine has always just worked. My system has 3 audio hardwares - 1] the HDMI built into my nVidia graphics card (though the audio portion of HDMI is blacklisted) 2] USB microphone section of Logitech webcam (though it still works just as well with a different Mic plugged into ======>>>> 3] Discrete semi-pro sound card ( ESI Juli@ I'm including the list in case alsa behaves differently with different hardware and notably mine has 4 discrete channels so has some onboard auto-mixing capability... however it is my understanding that essentially all sound cards have some level of internal mixing. I do use an .asoundrc but it is very basic and I can't see how it could be responsible for multi-source capability. Hopefully this may help even if only something to check off and discard. Any other data I can provide to help if you want it, just say so. Linux is so uneven when it comes to audio in that some stuff is so advanced and/or solid and others are Rube Goldberg at best, so it's difficult to diagnose something as complex as full second delays. This can be more complicated by other Rube like devices like Flash in Browsers which is giving me fits since a Firefox "upgrade". So again, anything you need..... |
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by direct i meant direct alsa by default, without an .asoundrc, resamples everything to 48kHz if you make it be direct it will fail when a request for a sample rate or bitrate is made that is different from the one currently being played at least on my 2 cards (i remember older ones too) example .asoundrc Code:
pcm.!default{ watch cat /proc/asound/card0/pcm0p/sub0/hw_params (for card0, subdevice 0) notice that when playing a, for example, 96kHz track without an asoundrc, it will say 48kHz |
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I can confirm that if I use an .asoundrc file to send audio to my NVidia card (which, and this is significant, is not card 0, device 0) and I do not specify that it goes through DMix, then running two MPlayer instances at the same time will result in the second one erroring out with "sound device busy." |
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problem is in the resampling part of mixing
mixing two inputs that have the same format is just add-ing them together with saturation funny, i tried now playing two songs with the same format and got a "device busy" i distinctly remember it working on my previous card (and that was a crap one) interesting anyway i agree, proper hw mixing is rarely found in non-professional hardware envy based cards being the only exception i found so far (ICE series has been continued as envy, they renamed after VIA bought them) |
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