Detect errors in audio output and report them upstream.

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/bind.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
 #include "content/renderer/media/audio_hardware.h"
@@ skipping 100 matching lines @@
         output_buffer_.get(),
         audio_data[channel_index],
         channels,
         channel_index,
         bytes_per_sample_,
         number_of_frames);
   }
   return number_of_frames;
 }
 
+void WebRtcAudioDeviceImpl::OnError() {
+  // TODO(henrika): Implement error handling.
+}
+
 void WebRtcAudioDeviceImpl::Capture(
     const std::vector<float*>& audio_data,
     size_t number_of_frames,
     size_t audio_delay_milliseconds) {
   DCHECK_LE(number_of_frames, input_buffer_size_);
 
   int output_delay_ms = 0;
   {
     base::AutoLock auto_lock(lock_);
     // Store the reported audio delay locally.
@@ skipping 847 matching lines @@
 }
 
 int32_t WebRtcAudioDeviceImpl::GetLoudspeakerStatus(bool* enabled) const {
   NOTIMPLEMENTED();
   return -1;
 }
 
 void WebRtcAudioDeviceImpl::SetSessionId(int session_id) {
   session_id_ = session_id;
 }