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Bug 953265: Adjust Opus bitrate in WebRTC to pass >8KHz audio, and comment r=bwc

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This commit is contained in:
Randell Jesup 2015-09-24 09:23:37 -04:00
parent 46bfe823f7
commit 44905ab6e3
1 changed files with 15 additions and 1 deletions
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16
media/webrtc/signaling/src/jsep/JsepSessionImpl.cpp
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@ -2006,13 +2006,27 @@ void
JsepSessionImpl::SetupDefaultCodecs()
{
// Supported audio codecs.
// Per jmspeex on IRC:
// For 32KHz sampling, 28 is ok, 32 is good, 40 should be really good
// quality. Note that 1-2Kbps will be wasted on a stereo Opus channel
// with mono input compared to configuring it for mono.
// If we reduce bitrate enough Opus will low-pass us; 16000 will kill a
// 9KHz tone. This should be adaptive when we're at the low-end of video
// bandwidth (say <100Kbps), and if we're audio-only, down to 8 or
// 12Kbps.
mSupportedCodecs.values.push_back(new JsepAudioCodecDescription(
"109",
"opus",
48000,
2,
960,
16000));
#ifdef WEBRTC_GONK
// TODO Move this elsewhere to be adaptive to rate - Bug 1207925
16000 // B2G uses lower capture sampling rate
#else
40000
#endif
));
mSupportedCodecs.values.push_back(new JsepAudioCodecDescription(
"9",