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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.

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TEST=ACM unit test is added, also a manual integration test is writen. 
Review URL: https://webrtc-codereview.appspot.com/1097009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2013-02-12 21:42:18 +00:00
parent e6e344a7dc
commit 6388c3e2fd
24 changed files with 839 additions and 85 deletions
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6
webrtc/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
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@ -127,7 +127,7 @@ int NETEQTEST_NetEQClass::init(WebRtc_UWord16 fs)
int NETEQTEST_NetEQClass::assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType)
{
int numPackets, memSize, ret;
int numPackets, memSize, ret, overhead_bytes;
if (!_inst)
{
@ -149,7 +149,9 @@ int NETEQTEST_NetEQClass::assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int n
}
}
ret = WebRtcNetEQ_GetRecommendedBufferSize(_inst, usedCodec, noOfCodecs, nwType, &numPackets, &memSize);
ret = WebRtcNetEQ_GetRecommendedBufferSize(_inst, usedCodec, noOfCodecs,
nwType, &numPackets, &memSize,
&overhead_bytes);
if (ret != 0)
{