commit 28d54ab18f18c7730105282d83df8a50a6b13e89
author turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Apr 22 18:53:35 2013 +0000
committer turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Apr 22 18:53:35 2013 +0000
tree 2804480b66fa8cfe0cdec2fa268fe1eab2b0760b
parent 1b427719dcfe21426aab89b1b85d90a39e26aea5

Improve AV-sync when initial delay is set and NetEq has long buffer.

Review URL: https://webrtc-codereview.appspot.com/1324006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/recin.c

diff --git a/webrtc/modules/audio_coding/neteq/recin.c b/webrtc/modules/audio_coding/neteq/recin.c
index f0dd210..15d618e 100644
--- a/webrtc/modules/audio_coding/neteq/recin.c
+++ b/webrtc/modules/audio_coding/neteq/recin.c
@@ -43,7 +43,8 @@
 #endif
 
     temp_bufsize = WebRtcNetEQ_PacketBufferGetSize(&MCU_inst->PacketBuffer_inst,
-                                                   &MCU_inst->codec_DB_inst);
+                                                   &MCU_inst->codec_DB_inst,
+                                                   MCU_inst->av_sync);
     /*
      * Copy from input RTP packet to local copy
      * (mainly to enable multiple payloads using RED)
@@ -223,7 +224,7 @@
                 MCU_inst->current_Codec = -1;
             }
             i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst,
-                &RTPpacket[i_k], &flushed);
+                &RTPpacket[i_k], &flushed, MCU_inst->av_sync);
             if (i_ok < 0)
             {
                 return RECIN_CNG_ERROR;
@@ -259,7 +260,8 @@
 
             /* Parse the payload and insert it into the buffer */
             i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k],
-                &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed);
+                &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst,
+                &flushed, MCU_inst->av_sync);
             if (i_ok < 0)
             {
                 return i_ok;
@@ -311,8 +313,8 @@
     {
         /* Calculate the total speech length carried in each packet */
         temp_bufsize = WebRtcNetEQ_PacketBufferGetSize(
-            &MCU_inst->PacketBuffer_inst, &MCU_inst->codec_DB_inst)
-            - temp_bufsize;
+            &MCU_inst->PacketBuffer_inst, &MCU_inst->codec_DB_inst,
+            MCU_inst->av_sync) - temp_bufsize;
 
         if ((temp_bufsize > 0) && (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF
             == 0) && (temp_bufsize