--- a/media/webrtc/trunk/webrtc/audio/audio_receive_stream.cc
+++ b/media/webrtc/trunk/webrtc/audio/audio_receive_stream.cc
@@ -198,17 +198,17 @@ webrtc::AudioReceiveStream::Stats AudioR
   stats.jitter_buffer_ms = ns.currentBufferSize;
   stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
   stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
   stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
   stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
   stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
   stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
 
-  auto ds = channel_proxy_->GetDecodingCallStatistics();
+  auto ds(channel_proxy_->GetDecodingCallStatistics());
   stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
   stats.decoding_calls_to_neteq = ds.calls_to_neteq;
   stats.decoding_normal = ds.decoded_normal;
   stats.decoding_plc = ds.decoded_plc;
   stats.decoding_cng = ds.decoded_cng;
   stats.decoding_plc_cng = ds.decoded_plc_cng;
   stats.decoding_muted_output = ds.decoded_muted_output;
 

--- a/media/webrtc/trunk/webrtc/common_types.h
+++ b/media/webrtc/trunk/webrtc/common_types.h
@@ -9,16 +9,17 @@
  */
 
 #ifndef WEBRTC_COMMON_TYPES_H_
 #define WEBRTC_COMMON_TYPES_H_
 
 #include <stddef.h>
 #include <string.h>
 
+#include <atomic>
 #include <ostream>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
@@ -403,24 +404,47 @@ struct AudioDecodingCallStats {
       : calls_to_silence_generator(0),
         calls_to_neteq(0),
         decoded_normal(0),
         decoded_plc(0),
         decoded_cng(0),
         decoded_plc_cng(0),
         decoded_muted_output(0) {}
 
-  int calls_to_silence_generator;  // Number of calls where silence generated,
+  AudioDecodingCallStats(const AudioDecodingCallStats& other)
+  {
+    calls_to_silence_generator = other.calls_to_silence_generator.load();
+    calls_to_neteq = other.calls_to_neteq.load();
+    decoded_normal = other.decoded_normal.load();
+    decoded_plc = other.decoded_plc.load();
+    decoded_cng = other.decoded_cng.load();
+    decoded_plc_cng = other.decoded_plc_cng.load();
+    decoded_muted_output = other.decoded_muted_output.load();
+  }
+
+  AudioDecodingCallStats& operator=(const AudioDecodingCallStats& other)
+  {
+    calls_to_silence_generator = other.calls_to_silence_generator.load();
+    calls_to_neteq = other.calls_to_neteq.load();
+    decoded_normal = other.decoded_normal.load();
+    decoded_plc = other.decoded_plc.load();
+    decoded_cng = other.decoded_cng.load();
+    decoded_plc_cng = other.decoded_plc_cng.load();
+    decoded_muted_output = other.decoded_muted_output.load();
+    return *this;
+  }
+
+  std::atomic<int> calls_to_silence_generator;  // Number of calls where silence generated,
                                    // and NetEq was disengaged from decoding.
-  int calls_to_neteq;              // Number of calls to NetEq.
-  int decoded_normal;  // Number of calls where audio RTP packet decoded.
-  int decoded_plc;     // Number of calls resulted in PLC.
-  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
-  int decoded_plc_cng;  // Number of calls resulted where PLC faded to CNG.
-  int decoded_muted_output;  // Number of calls returning a muted state output.
+  std::atomic<int> calls_to_neteq;              // Number of calls to NetEq.
+  std::atomic<int> decoded_normal;  // Number of calls where audio RTP packet decoded.
+  std::atomic<int> decoded_plc;     // Number of calls resulted in PLC.
+  std::atomic<int> decoded_cng;  // Number of calls where comfort noise generated due to DTX.
+  std::atomic<int> decoded_plc_cng;  // Number of calls resulted where PLC faded to CNG.
+  std::atomic<int> decoded_muted_output;  // Number of calls returning a muted state output.
 };
 
 // Type of Noise Suppression.
 enum NsModes {
   kNsUnchanged = 0,   // previously set mode
   kNsDefault,         // platform default
   kNsConference,      // conferencing default
   kNsLowSuppression,  // lowest suppression

--- a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -114,19 +114,16 @@ int AcmReceiver::InsertPacket(const WebR
   }
   return 0;
 }
 
 int AcmReceiver::GetAudio(int desired_freq_hz,
                           AudioFrame* audio_frame,
                           bool* muted) {
   RTC_DCHECK(muted);
-  // Accessing members, take the lock.
-  rtc::CritScope lock(&crit_sect_);
-
   if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) {
     LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed.";
     return -1;
   }
 
   const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz();
 
   // Update if resampling is required.

--- a/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -273,21 +273,23 @@ class AcmReceiver {
       uint8_t first_payload_byte) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::CriticalSection crit_sect_;
   rtc::Optional<CodecInst> last_audio_decoder_ GUARDED_BY(crit_sect_);
   rtc::Optional<SdpAudioFormat> last_audio_format_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
-  std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
+  // After construction, this is only ever touched on the thread that calls
+  // AcmReceiver::GetAudio, and only modified in this method.
+  std::unique_ptr<int16_t[]> last_audio_buffer_;
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
-  NetEq* neteq_;
+  NetEq* const neteq_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
-  bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
+  std::atomic<bool> resampled_last_output_frame_;
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
   std::atomic<int> last_audio_format_clockrate_hz_;
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc