In PeerConnectionTestWrapper, put audio input on a separate thread.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 256 matching lines @@
     int SetChannelOutputVolumeScaling(float scaling);
     int GetChannelOutputVolumeScaling(float& scaling) const;
 
     // VoENetEqStats
     int GetNetworkStatistics(NetworkStatistics& stats);
     void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
     // VoEVideoSync
     bool GetDelayEstimate(int* jitter_buffer_delay_ms,
                           int* playout_buffer_delay_ms) const;
-    int least_required_delay_ms() const { return least_required_delay_ms_; }
+    int least_required_delay_ms() const {
+      CriticalSectionScoped cs(&video_sync_critsect_);

[pthatcher1, 2015/07/24 09:04:09]

Can you put these in a separate CL and have Tina's team review it?

+      return least_required_delay_ms_;
+    }
     int SetInitialPlayoutDelay(int delay_ms);
     int SetMinimumPlayoutDelay(int delayMs);
     int GetPlayoutTimestamp(unsigned int& timestamp);
     void UpdatePlayoutTimestamp(bool rtcp);
     int SetInitTimestamp(unsigned int timestamp);
     int SetInitSequenceNumber(short sequenceNumber);
 
     // VoEVideoSyncExtended
     int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
@@ skipping 183 matching lines @@
     int SetRedPayloadType(int red_payload_type);
     int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
                                   unsigned char id);
 
     int32_t GetPlayoutFrequency();
     int64_t GetRTT(bool allow_associate_channel) const;
 
     CriticalSectionWrapper& _fileCritSect;
     CriticalSectionWrapper& _callbackCritSect;
     CriticalSectionWrapper& volume_settings_critsect_;
+    CriticalSectionWrapper& video_sync_critsect_;
     uint32_t _instanceId;
     int32_t _channelId;
 
     ChannelState channel_state_;
 
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
     rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
     rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
@@ skipping 87 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_