From bb8ada686e9becb9cea6db79fa691ee5aacfde73 Mon Sep 17 00:00:00 2001
From: "henrika@webrtc.org"
 <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Thu, 4 Apr 2013 08:39:09 +0000
Subject: [PATCH] TSan v2 reports data races in
 WebRTCAudioDeviceTest.FullDuplexAudioWithAGC

BUG=226044
TEST=content_unittests in Chrome with TSan v2 enabled

Review URL: https://webrtc-codereview.appspot.com/1201010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3760 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 .../main/source/audio_coding_module_impl.cc   | 28 ++++++++++++-------
 webrtc/voice_engine/transmit_mixer.cc         | 17 +++++++++--
 2 files changed, 33 insertions(+), 12 deletions(-)

diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
index ba7bde1254..27cce93bca 100644
--- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -2053,8 +2053,11 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
       return 0;
     }
   } else {
-    if (track_neteq_buffer_)
-      num_bytes_accumulated_ += payload_length;
+    {
+      CriticalSectionScoped lock(acm_crit_sect_);
+      if (track_neteq_buffer_)
+        num_bytes_accumulated_ += payload_length;
+    }
     return neteq_.RecIn(incoming_payload, payload_length, rtp_header);
   }
 }
@@ -2136,9 +2139,12 @@ WebRtc_Word32 AudioCodingModuleImpl::SetMinimumPlayoutDelay(
                  "Delay must be in the range of 0-10000 milliseconds.");
     return -1;
   }
-  // Don't let the extra delay modified while accumulating buffers in NetEq.
-  if (track_neteq_buffer_ && first_payload_received_)
-    return 0;
+  {
+    CriticalSectionScoped lock(acm_crit_sect_);
+    // Don't let the extra delay modified while accumulating buffers in NetEq.
+    if (track_neteq_buffer_ && first_payload_received_)
+      return 0;
+  }
   return neteq_.SetExtraDelay(time_ms);
 }
 
@@ -2665,12 +2671,14 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutTimestamp(
     WebRtc_UWord32* timestamp) {
   WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                "PlayoutTimestamp()");
-  if (track_neteq_buffer_) {
-    *timestamp = playout_ts_;
-    return 0;
-  } else {
-    return neteq_.PlayoutTimestamp(*timestamp);
+  {
+    CriticalSectionScoped lock(acm_crit_sect_);
+    if (track_neteq_buffer_) {
+      *timestamp = playout_ts_;
+      return 0;
+    }
   }
+  return neteq_.PlayoutTimestamp(*timestamp);
 }
 
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 242b965c9f..0e4717c274 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -56,7 +56,18 @@ TransmitMixer::OnPeriodicProcess()
     }
 #endif
 
-    if (_saturationWarning)
+    bool saturationWarning = false;
+    {
+      // Modify |_saturationWarning| under lock to avoid conflict with write op
+      // in ProcessAudio and also ensure that we don't hold the lock during the
+      // callback.
+      CriticalSectionScoped cs(&_critSect);
+      saturationWarning = _saturationWarning;
+      if (_saturationWarning)
+        _saturationWarning = false;
+    }
+
+    if (saturationWarning)
     {
         CriticalSectionScoped cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr)
@@ -66,7 +77,6 @@ TransmitMixer::OnPeriodicProcess()
                          " CallbackOnError(VE_SATURATION_WARNING)");
             _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
         }
-        _saturationWarning = false;
     }
 }
 
@@ -454,6 +464,7 @@ TransmitMixer::EncodeAndSend()
 
 WebRtc_UWord32 TransmitMixer::CaptureLevel() const
 {
+    CriticalSectionScoped cs(&_critSect);
     return _captureLevel;
 }
 
@@ -1311,6 +1322,8 @@ void TransmitMixer::ProcessAudio(int delay_ms, int clock_drift,
     LOG(LS_ERROR) << "ProcessStream() error: " << err;
   }
 
+  CriticalSectionScoped cs(&_critSect);
+
   // Store new capture level. Only updated when analog AGC is enabled.
   _captureLevel = agc->stream_analog_level();