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.
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
diff --git a/webrtc/modules/audio_coding/neteq/packet_buffer.c b/webrtc/modules/audio_coding/neteq/packet_buffer.c
index bb2d08e..c51805e 100644
--- a/webrtc/modules/audio_coding/neteq/packet_buffer.c
+++ b/webrtc/modules/audio_coding/neteq/packet_buffer.c
@@ -577,7 +577,9 @@
}
int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
- int noOfCodecs, int *maxBytes, int *maxSlots)
+ int noOfCodecs, int *maxBytes,
+ int *maxSlots,
+ int* per_slot_overhead_bytes)
{
int i;
int ok = 0;
@@ -794,5 +796,6 @@
/* Add the extra size per slot to the memory count */
*maxBytes += w16_tmp * (*maxSlots);
+ *per_slot_overhead_bytes = w16_tmp;
return ok;
}