Home › Forums › Breakaway Professional Products – [discontinued] › Audio stuttering using Breakaway Live with Station Playlist
- This topic has 14 replies, 5 voices, and was last updated 14 years, 4 months ago by Don935.
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July 13, 2010 at 12:42 am #896Don935Member
Hi! I’ve been using Breakaway Live since early this year,and I love it, but I’ve developed a problem.
I’ve recently switched my playout software for my internet station from SAM Broadcaster to Station Playlist (SPL). Now, every 65-70 hours of operation, the audio output starts to stutter. The audio stops every 2-3 seconds. When this occurs, I have to restart SPL, or sometimes reboot the PC to get rid of the problem, which returns after another 65-70 hours of play.
Here’s my setup:
Windows 7 Home Premium
CPU: Pentium 4, 3GHz
3 GBs RAMThe main player and voicetrack players of SPL output to Line 1 of the Pipeline, where it’s processed and sent to Edcast. There is no soundcard involved (I had one initially, but took it out of the system to see if it was the problem. It wasn’t.).
Settings for Live:
Input: KS, Pipeline 1, 44100; Buffer size – 2048, 16 buffers
Output: KS, Pipeline 2, 48000; Buffer size – 2048, 16 buffersI’d appreciate any ideas or tips you could give. If I can’t solve this problem, I’ll have to start using Sound Solution…and to say I don’t like that possibility would be putting it mildly. 🙁
Thanks,
Don C.
Project42Radio.comJuly 13, 2010 at 2:49 am #11060timmywaParticipantFirst thing that pops out its the speed. Keep everything at 44100 khz. Check the main config file in edcast to make sure its set to 44100 as well.
July 13, 2010 at 6:51 am #11061Dr.JMemberIf the only thing you’ve changed is the playout software, then that’s where the problem must lie. If you’ve changed other parts of your system, then they could be possible culprits also. Make sure you’re barking up the right tree before cutting it down.
July 13, 2010 at 10:04 am #11062JesseGMemberWhere does SPL get the clock it uses for its audio outputs? That would be the first thing I would try to figure out.
July 13, 2010 at 5:31 pm #11063Don935MemberJesseG: Good idea; I’ll ask Ross on the SPL forums and update here when I get an answer.
Timmywa: I switched pipeline 2 to 44100; the edcast encoders are already set there. Thanks.
July 13, 2010 at 6:20 pm #11064timmywaParticipantDon, use the built-in jitter measurement tool to get your buffer #’s and size down to the lowest size while maintaining under 20% jitter. There’s some posts about this in another thread, I think up in a sticky post. Not sure at the moment.
You should be able to get it down to 2 buffers each, then just tweak the size until your jitter stabilizes around 10-20% or better.
July 13, 2010 at 7:05 pm #11065JesseGMember[quote author=”timmywa”]Don, use the built-in jitter measurement tool to get your buffer #’s and size down to the lowest size while maintaining under 20% jitter. There’s some posts about this in another thread, I think up in a sticky post. Not sure at the moment.
You should be able to get it down to 2 buffers each, then just tweak the size until your jitter stabilizes around 10-20% or better.[/quote]
I don’t recommend doing this if you’re trying to get stable (at all) sound right now. Find out where the problem is, before tossing more wood on the fire. 😉
July 13, 2010 at 7:18 pm #11066timmywaParticipant[quote author=”JesseG”][quote author=”timmywa”]Don, use the built-in jitter measurement tool to get your buffer #’s and size down to the lowest size while maintaining under 20% jitter. There’s some posts about this in another thread, I think up in a sticky post. Not sure at the moment.
You should be able to get it down to 2 buffers each, then just tweak the size until your jitter stabilizes around 10-20% or better.[/quote]
I don’t recommend doing this if you’re trying to get stable (at all) sound right now. Find out where the problem is, before tossing more wood on the fire. 😉[/quote]
Yes, listen to Jesse! He is wise.
July 13, 2010 at 10:40 pm #11067Don935MemberI shall do that. 😀
By the way: if I am not using a sound card at all, should SRC be checked?
July 14, 2010 at 12:02 am #11068michi95Member[quote author=”Don935″]if I am not using a sound card at all, should SRC be checked?[/quote]When you move the mouse pointer over SRC button in the I/0 configuration, the popup tooltip says:
"Adaptive Sample Rate Conversion
If both the input and output are on the
same sound card, disable this option."July 14, 2010 at 6:00 am #11069Don935MemberWell, then if I’m using Live as the virtual soundcard, it should be unchecked, then?
Ross Levis of SPL says the output sound device controls the speed, which in my case is Live. SPL Studio (the playout software) uses Direct Sound for all audio output. Ross has suggested increasing the output buffer in SPL (it’s currently set at 500 ms), so I’ll at least double that, and take it from there.
July 14, 2010 at 10:16 am #11070Dr.JMemberI would enable SRC to see if it helps. I highly doubt it will hurt.
July 14, 2010 at 11:25 am #11071JesseGMemberSRC should absolutely be checked.
July 14, 2010 at 6:57 pm #11072Don935Member[quote author=”JesseG”]SRC should absolutely be checked.[/quote]
Done. It was anyway; just wanted to be sure. You’ve all been great so far. Thanks for the comments!
July 16, 2010 at 2:35 pm #11073Don935MemberThe problem seems to be solved. The critical time period passed with no breakup! I believe it might have been a combination of properly setting the sample rate in Live and enlarging the buffer size in SPL. In any case, everything is smooooth now. Thanks, folks.
One more thing…if you folks have a moment, could you check the stream and see how it’s sounding to you? I’m using a tweaked Plutonium and it sounds good to me, but another set of ears is always helpful.
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