webrtc /
src /
522227012d36f629a2e4628bdce942597f648283 Reset audio bufer if codec changes, b/10835525.
If there is audio in a codec's audio buffer and sample-rate or number of channels change the audio buffer has to reset. Otherwise, the amount of audio in the buffer is misinterpreted any syncronization between 10ms audio blocks and their associated timestamps is lost.
For instance, assume changing from stereo to mono when there is 10ms stereo in the buffer. The "new" codec will interpret this as 20 ms audio, therefore, 2 blocks of 10 ms, but there is only one timestamp. This will results in ACMGenericCodec::in_timestamp_ix_write_ updated to a negative number after an encode is performed.
The drawback with this solution is that if packet-size of the codec is changed then audio buffer is reset wich is not necessary. We accept this as it is a rare case in practice that clients of ACM re-register send codecs to change packet-size.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2151006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4887 4adac7df-926f-26a2-2b94-8c16560cd09d
2 files changed
