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

Review URL: http://codereview.chromium.org/8799011

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@114002 0039d316-1c4b-4281-b951-d872f2087c98

content/renderer/media/webrtc_audio_device_impl.cc

@@ -65,8 +65,11 @@ void WebRtcAudioDeviceImpl::Render(
     size_t audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, output_buffer_size_);
 
-  // Store the reported audio delay locally.
-  output_delay_ms_ = audio_delay_milliseconds;
+  {
+    base::AutoLock auto_lock(lock_);
+    // Store the reported audio delay locally.
+    output_delay_ms_ = audio_delay_milliseconds;
+  }
 
   const int channels = audio_data.size();
   DCHECK_LE(channels, output_channels_);
@@ -119,8 +122,13 @@ void WebRtcAudioDeviceImpl::Capture(
     size_t audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, input_buffer_size_);
 
-  // Store the reported audio delay locally.
-  input_delay_ms_ = audio_delay_milliseconds;
+  int output_delay_ms = 0;
+  {
+    base::AutoLock auto_lock(lock_);
+    // Store the reported audio delay locally.
+    input_delay_ms_ = audio_delay_milliseconds;
+    output_delay_ms = output_delay_ms_;
+  }
 
   const int channels = audio_data.size();
   DCHECK_LE(channels, input_channels_);
@@ -156,7 +164,7 @@ void WebRtcAudioDeviceImpl::Capture(
         bytes_per_sample_,
         channels,
         samples_per_sec,
-        input_delay_ms_ + output_delay_ms_,
+        input_delay_ms_ + output_delay_ms,
         0,  // clock_drift
         0,  // current_mic_level
         new_mic_level);  // not used
@@ -642,12 +650,17 @@ int32_t WebRtcAudioDeviceImpl::StopRecording() {
   DVLOG(1) << "StopRecording()";
   DCHECK(audio_input_device_);
 
-  base::AutoLock auto_lock(lock_);
-  if (!recording_) {
-    // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case.
-    return 0;
+  {
+    base::AutoLock auto_lock(lock_);
+    if (!recording_) {
+      // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case.
+      return 0;
+    }
   }
+
   audio_input_device_->Stop();
+
+  base::AutoLock auto_lock(lock_);
   recording_ = false;
   return 0;
 }
@@ -890,12 +903,14 @@ int32_t WebRtcAudioDeviceImpl::PlayoutBuffer(BufferType* type,
 
 int32_t WebRtcAudioDeviceImpl::PlayoutDelay(uint16_t* delay_ms) const {
   // Report the cached output delay value.
+  base::AutoLock auto_lock(lock_);
   *delay_ms = static_cast<uint16_t>(output_delay_ms_);
   return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::RecordingDelay(uint16_t* delay_ms) const {
   // Report the cached output delay value.
+  base::AutoLock auto_lock(lock_);
   *delay_ms = static_cast<uint16_t>(input_delay_ms_);
   return 0;
 }

content/renderer/media/webrtc_audio_device_impl.h

@@ -321,8 +321,8 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // on the input/capture side.
   int session_id_;
 
-  // Protects |recording_|.
-  base::Lock lock_;
+  // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|.
+  mutable base::Lock lock_;
 
   int bytes_per_sample_;
 

content/renderer/media/webrtc_audio_device_unittest.cc

@@ -107,6 +107,7 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
                        const int length,
                        const int sampling_freq,
                        const bool is_stereo) {
+    base::AutoLock auto_lock(lock_);
     channel_id_ = channel;
     type_ = type;
     packet_size_ = length;
@@ -118,11 +119,30 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
     }
   }
 
-  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_;
@@ -131,6 +151,7 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
   int packet_size_;
   int sample_rate_;
   int channels_;
+  mutable base::Lock lock_;
   DISALLOW_COPY_AND_ASSIGN(WebRTCMediaProcessImpl);
 };