In PeerConnectionTestWrapper, put audio input on a separate thread.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 8992425b537b0e32d329a470d58fe50d2619171d..164b99fd60a63b3733f9687af9c4d23d2ea011d8 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -717,6 +717,7 @@ Channel::Channel(int32_t channelId,
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
+    video_sync_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _channelId(channelId),
     rtp_header_parser_(RtpHeaderParser::Create()),
@@ -899,6 +900,7 @@ Channel::~Channel()
     delete &_callbackCritSect;
     delete &_fileCritSect;
     delete &volume_settings_critsect_;
+    delete &video_sync_critsect_;
 }
 
 int32_t
@@ -3561,6 +3563,7 @@ void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
 
 bool Channel::GetDelayEstimate(int* jitter_buffer_delay_ms,
                                int* playout_buffer_delay_ms) const {
+  CriticalSectionScoped cs(&video_sync_critsect_);
   if (_average_jitter_buffer_delay_us == 0) {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::GetDelayEstimate() no valid estimate.");
@@ -3640,26 +3643,30 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
     return;
   }
 
-  jitter_buffer_playout_timestamp_ = playout_timestamp;
+  {
+    CriticalSectionScoped cs(&video_sync_critsect_);
+    jitter_buffer_playout_timestamp_ = playout_timestamp;
 
-  // Remove the playout delay.
-  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+    // Remove the playout delay.
+    playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
 
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
-               "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
-               playout_timestamp);
+    WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
+                 "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
+                 playout_timestamp);
 
-  if (rtcp) {
-    playout_timestamp_rtcp_ = playout_timestamp;
-  } else {
-    playout_timestamp_rtp_ = playout_timestamp;
+    if (rtcp) {
+      playout_timestamp_rtcp_ = playout_timestamp;
+    } else {
+      playout_timestamp_rtp_ = playout_timestamp;
+    }
+    playout_delay_ms_ = delay_ms;
   }
-  playout_delay_ms_ = delay_ms;
 }
 
 int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::GetPlayoutTimestamp()");
+  CriticalSectionScoped cs(&video_sync_critsect_);
   if (playout_timestamp_rtp_ == 0)  {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_RETRIEVE_VALUE, kTraceError,
@@ -3929,6 +3936,8 @@ void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
                "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
                rtp_timestamp, sequence_number);
 
+  CriticalSectionScoped cs(&video_sync_critsect_);
+
   // Get frequency of last received payload
   int rtp_receive_frequency = GetPlayoutFrequency();