Refactor AudioInputDevice to remove race conditions and allow more flexible calling sequences.

content/renderer/media/webrtc_audio_device_impl.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 "content/renderer/media/webrtc_audio_device_impl.h"
 
 #include "base/string_util.h"
 #include "media/audio/audio_util.h"
 
 // TODO(henrika): come up with suitable value(s) for all platforms.
@@ skipping 371 matching lines @@
   if (!audio_transport_callback_) {
     LOG(ERROR) << "Audio transport is missing";
     return -1;
   }
   if (recording_) {
     // webrtc::VoiceEngine assumes that it is OK to call Start() twice and
     // that the call is ignored the second time.
     LOG(WARNING) << "Recording is already active";
     return 0;
   }
-  recording_ = audio_input_device_->Start();
-  return (recording_ ? 0 : -1);
+  audio_input_device_->Start();
+  recording_ = true;
+  return 0;
 }
 
 int32_t WebRtcAudioDeviceImpl::StopRecording() {
   VLOG(1) << "StopRecording()";
   DCHECK(audio_input_device_);
   if (!recording_) {
     // webrtc::VoiceEngine assumes that it is OK to call Stop() just in case.
     LOG(WARNING) << "Recording was already stopped";
     return 0;
   }
@@ skipping 319 matching lines @@
     size_t buffer_size, float sample_rate) const {
   const int samples_per_sec = static_cast<int>(sample_rate);
   const int samples_per_10_msec = (samples_per_sec / 100);
   bool size_is_valid = (((buffer_size % samples_per_10_msec) == 0) &&
                          (buffer_size <= kMaxBufferSize));
   DLOG_IF(WARNING, !size_is_valid) << "Size of buffer must be and even "
                                    << "multiple of 10 ms and less than "
                                    << kMaxBufferSize;
   return size_is_valid;
 }