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;
+  if (!base::subtle::NoBarrier_Load(&have_received_packets_)) return false;

[Alpha Left Google, 2013/11/15 19:40:39]

We very rarely use atomicops. What is the reason that we need them here? I
though this object is non-threadsafe so what are the threads that use this
object?

[pwestin, 2013/11/15 19:48:06]

I was looking for other options but did not find any simpler approach. 

GetRawAudioFrame is called from the cast audio decoder thread and
IncomingParsedRtpPacket is called from the Main cast thread

 
   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);
-  have_received_packets_ = true;
+  base::subtle::NoBarrier_Store(&have_received_packets_, true);
 }
 
 }  // namespace cast
 }  // namespace media