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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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11
webrtc/modules/audio_coding/neteq/automode.c
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@ -216,7 +216,7 @@ int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
streamingMode);
if (tempvar > 0)
{
inst->optBufLevel = (WebRtc_UWord16) tempvar;
inst->optBufLevel = tempvar;
if (streamingMode != 0)
{
@ -238,7 +238,7 @@ int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */
/* Enforce upper limit; 75% of maxBufLen */
inst->optBufLevel = (WebRtc_UWord16) WEBRTC_SPL_MIN( inst->optBufLevel,
inst->optBufLevel = WEBRTC_SPL_MIN( inst->optBufLevel,
(maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */
}
else
@ -575,9 +575,8 @@ int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst
*
* levelFiltFact is in Q8
*/
inst->buffLevelFilt = (WebRtc_UWord16) (WEBRTC_SPL_RSHIFT_W32(
WEBRTC_SPL_MUL_16_U16(inst->levelFiltFact, inst->buffLevelFilt), 8)
+ WEBRTC_SPL_MUL_16_16(256 - inst->levelFiltFact, curSizeFrames));
inst->buffLevelFilt = ((inst->levelFiltFact * inst->buffLevelFilt) >> 8) +
(256 - inst->levelFiltFact) * curSizeFrames;
}
/* Account for time-scale operations (accelerate and pre-emptive expand) */
@ -589,7 +588,7 @@ int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst
* from samples to packets in Q8. Make sure that the filtered value is
* non-negative.
*/
inst->buffLevelFilt = (WebRtc_UWord16) WEBRTC_SPL_MAX( inst->buffLevelFilt -
inst->buffLevelFilt = WEBRTC_SPL_MAX( inst->buffLevelFilt -
WebRtcSpl_DivW32W16(
WEBRTC_SPL_LSHIFT_W32(inst->sampleMemory, 8), /* sampleMemory in Q8 */
inst->packetSpeechLenSamp ), /* divide by packetSpeechLenSamp */