Cast: Removed unnecessary ref counters.

media/cast/audio_receiver/audio_decoder.cc

 // Copyright 2013 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/logging.h"
 #include "media/cast/audio_receiver/audio_decoder.h"
 
 #include "third_party/webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 #include "third_party/webrtc/modules/interface/module_common_types.h"
 
@@ skipping 34 matching lines @@
   audio_decoder_->SetMaximumPlayoutDelay(audio_config.rtp_max_delay_ms);
   audio_decoder_->SetPlayoutMode(webrtc::streaming);
 }
 
 AudioDecoder::~AudioDecoder() {}
 
 bool AudioDecoder::GetRawAudioFrame(int number_of_10ms_blocks,
                                     int desired_frequency,
                                     PcmAudioFrame* audio_frame,
                                     uint32* rtp_timestamp) {
-  if (!have_received_packets_) return false;
+  // Get a local copy under lock.

[Alpha Left Google, 2013/11/22 20:03:53]

Do you expect to have a race condition using the lock this way? Or this is a
race condition that we do not care?

[pwestin, 2013/11/22 21:16:36]

We don't care about the race where a thread ask for audio and the packet come in
at the same time. We will get the data in the next get request. 

+  lock_.Acquire();
+  bool have_received_packets = have_received_packets_;
+  lock_.Release();
+
+  if (!have_received_packets) return false;
 
   audio_frame->samples.clear();
 
   for (int i = 0; i < number_of_10ms_blocks; ++i) {
     webrtc::AudioFrame webrtc_audio_frame;
     if (0 != audio_decoder_->PlayoutData10Ms(desired_frequency,
                                              &webrtc_audio_frame)) {
       return false;
     }
     if (webrtc_audio_frame.speech_type_ == webrtc::AudioFrame::kPLCCNG ||
@@ skipping 19 matching lines @@
   }
   return true;
 }
 
 void AudioDecoder::IncomingParsedRtpPacket(const uint8* payload_data,
                                            size_t payload_size,
                                            const RtpCastHeader& rtp_header) {
   DCHECK_LE(payload_size, kIpPacketSize);
   audio_decoder_->IncomingPacket(payload_data, static_cast<int32>(payload_size),
                                  rtp_header.webrtc);
+
+  lock_.Acquire();
   have_received_packets_ = true;
+  lock_.Release();
 }
 
 }  // namespace cast
 }  // namespace media