Fix threading issues in the audio pipeline for WebRTC remoting protocol.

remoting/protocol/webrtc_audio_module.cc

 // Copyright 2016 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 "remoting/protocol/webrtc_audio_module.h"
 
 #include "base/bind.h"
+#include "base/memory/ptr_util.h"
 #include "base/stl_util.h"
 #include "base/threading/thread_task_runner_handle.h"
+#include "base/timer/timer.h"
 
 namespace remoting {
 namespace protocol {
 
 namespace {
 
 const int kSamplingRate = 48000;
 
 // Webrtc uses 10ms frames.
 const int kFrameLengthMs = 10;
@@ skipping 475 matching lines @@
 
 int WebrtcAudioModule::GetRecordAudioParameters(
     webrtc::AudioParameters* params) const {
   NOTREACHED();
   return -1;
 }
 #endif  // WEBRTC_IOS
 
 void WebrtcAudioModule::StartPlayoutOnAudioThread() {
   DCHECK(audio_task_runner_->BelongsToCurrentThread());
-  poll_timer_.Start(
+  poll_timer_ = base::MakeUnique<base::RepeatingTimer>();
+  poll_timer_->Start(
       FROM_HERE, kPollInterval,
       base::Bind(&WebrtcAudioModule::PollFromSource, base::Unretained(this)));
 }
 
 void WebrtcAudioModule::StopPlayoutOnAudioThread() {
   DCHECK(audio_task_runner_->BelongsToCurrentThread());
-  poll_timer_.Stop();
+  poll_timer_.reset();
 }
 
 void WebrtcAudioModule::PollFromSource() {
   DCHECK(audio_task_runner_->BelongsToCurrentThread());
 
   base::AutoLock lock(lock_);
   if (!audio_transport_)
     return;
 
   for (int i = 0; i < kPollInterval.InMilliseconds() / kFrameLengthMs; i++) {
     int64_t elapsed_time_ms = -1;
     int64_t ntp_time_ms = -1;
     char data[kBytesPerSample * kChannels * kSamplesPerFrame];
     audio_transport_->PullRenderData(kBytesPerSample * 8, kSamplingRate,
                                      kChannels, kSamplesPerFrame, data,
                                      &elapsed_time_ms, &ntp_time_ms);
   }
 }
 
 }  // namespace protocol
 }  // namespace remoting