justinlin@chromium.org f81fad6267 Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher ...

than 2^16kbps.
Review URL: https://webrtc-codereview.appspot.com/1275004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3748 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-01 22:25:11 +00:00

..

codecs

WebRtc_Word -> stdint in audio_coding/g711/

2013-03-21 13:38:29 +00:00

main

Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher

2013-04-01 22:25:11 +00:00

neteq

Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.

2013-02-12 21:42:18 +00:00

neteq4

G722-stereo has been missing when creating AudioDecoder.

2013-03-27 20:42:48 +00:00