poor audio quality in chrome from freeswitch

[Jozsef Vass]

We are experiencing audio quality issues in Chrome (tested OSX, version 31.0.1650.63). We are using freeswitch for bridging calls. The original audio is PCMU at 8 kHz.  freeswitch is configured to convert use Opus at 48 kHz Opus at 32 kbps, mono, no dtx and no fec. After encryption, the audio is sent to Chrome. At Chrome, audio sounds muffled, talking through a pipe. Wireshark at Chrome shows no packet loss, packet timestamps are incremented by 960.

The same audio played back in Firefox sounds perfect. Extracting packets from pcap, decrypting and decoding also sounds fine.

What could be the reason audio distortion in Chrome? 

Jozsef

[Brave Yao]

Could you please try Canary? From other reports, https://code.google.com/p/webrtc/issues/detail?id=2633, it seems a Chrome 31 only problem, which is weird since it's same Opus running in all Chrome 30+ versions. 

[Jozsef Vass]

Hello Brave,

I have tried osx Chrome Canari 34.0.1771.0 and audio is still distorted. I have taken freeswitch out of the picture. I have recorded audio using QuickTime (aac, 44.1 kHz, stereo). Then, I converted this file to 48 kHz, mono wav format using http://audio.online-convert.com/. Finally, I compressed to Opus and looped  to Chrome (replacing audio from freeswitch). 

One more interesting thing. Distortion will not occur when Chrome generates the offer. Only when Chrome generates the answer we get the distortion.

This issue did not exist in Chrome 30. I have tested Chromium 31.0.1608.0 (218892) and sounded good. I will narrow down tomorrow the exact build and look into the corresponding webrtc code.

Jozsef

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[Jozsef Vass]

I check Chromium continuous builds and 219161 is OK and 219234 is BAD.

Jozsef

[Brave Yao]

Thanks for the bisecting. Between r219161 and 219234, I only see a webrtc roll from 4533 to 4595 in r219171 which might be relevant. With a quick review, no suspicious revision could be found [4533, 4595]. According to your description, "Only when Chrome generates the answer we get the distortion", it doesn't look like a Codec issue too. Would there be same problem with other codec?

Anyway, we can follow it in issue tracker. Please attach the full SDP from chrome side when it's either caller and callee. Also the decrypted Opus RTP stream. Chrome log with --vmodule=*libjingle/source/talk/*=4 might be helpful too.

Thanks!

[Jozsef Vass]

I have created issue https://code.google.com/p/webrtc/issues/detail?id=2768. 

We have tested PCMU and iSAC and they are fine, this is only an Opus issue.

I will add more logs to the bug later. I am pretty sure that the issue came in with the webrtc update. It is pretty extensive. 

Jozsef