remove the race related to output_delay_ms_ in ADM and races in the unittests.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager.h"
@@ skipping 89 matching lines @@
   }
   virtual ~WebRTCMediaProcessImpl() {}
 
   // TODO(henrika): Refactor in WebRTC and convert to Chrome coding style.
   virtual void Process(const int channel,
                        const webrtc::ProcessingTypes type,
                        WebRtc_Word16 audio_10ms[],
                        const int length,
                        const int sampling_freq,
                        const bool is_stereo) {
+    base::AutoLock auto_lock(lock_);
     channel_id_ = channel;
     type_ = type;
     packet_size_ = length;
     sample_rate_ = sampling_freq;
     channels_ = (is_stereo ? 2 : 1);
     if (event_) {
       // Signal that a new callback has been received.
       event_->Signal();
     }
   }
 
-  int channel_id() const { return channel_id_; }
-  int type() const { return type_; }
-  int packet_size() const { return packet_size_; }
-  int sample_rate() const { return sample_rate_; }
-  int channels() const { return channels_; }
+  int channel_id() const {
+    base::AutoLock auto_lock(lock_);
+    return channel_id_;
+  }
+
+  int type() const {
+    base::AutoLock auto_lock(lock_);
+    return type_;
+  }
+
+  int packet_size() const {
+    base::AutoLock auto_lock(lock_);
+    return packet_size_;
+  }
+
+  int sample_rate() const {
+    base::AutoLock auto_lock(lock_);
+    return sample_rate_;
+  }
+
+  int channels() const {
+    base::AutoLock auto_lock(lock_);
+    return channels_;
+  }
 
  private:
   base::WaitableEvent* event_;
   int channel_id_;
   webrtc::ProcessingTypes type_;
   int packet_size_;
   int sample_rate_;
   int channels_;
+  mutable base::Lock lock_;
   DISALLOW_COPY_AND_ASSIGN(WebRTCMediaProcessImpl);
 };
 
 }  // end namespace
 
 // Utility class to delete the AudioManager.
 // TODO(tommi): Remove when we've fixed issue 105249.
 class AutoAudioManagerCleanup {
  public:
   AutoAudioManagerCleanup() {
@@ skipping 301 matching lines @@
                                 new MessageLoop::QuitTask(),
                                 TestTimeouts::action_timeout_ms());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }