Test submission of complete AEC-dump refactoring.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 547 matching lines @@
     : WebRtcVoiceEngine(adm, decoder_factory, audio_mixer, new VoEWrapper()) {
   audio_state_ =
       webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
-    : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
+    : worker_queue_("file_writer_task_queue_"),
+      adm_(adm),
+      decoder_factory_(decoder_factory),
+      voe_wrapper_(voe_wrapper) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
   RTC_DCHECK(decoder_factory);
 
   signal_thread_checker_.DetachFromThread();
 
   // Load our audio codec list.
   LOG(LS_INFO) << "Supported send codecs in order of preference:";
   send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
@@ skipping 413 matching lines @@
                                      int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
   if (!aec_dump_file_stream) {
     LOG(LS_ERROR) << "Could not open AEC dump file stream.";
     if (!rtc::ClosePlatformFile(file))
       LOG(LS_WARNING) << "Could not close file.";
     return false;
   }
   StopAecDump();
-  if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) !=
+  if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes,
+                                 &worker_queue_) !=
       webrtc::AudioProcessing::kNoError) {
     LOG_RTCERR0(StartDebugRecording);
     fclose(aec_dump_file_stream);
     return false;
   }
   is_dumping_aec_ = true;
   return true;
 }
 
 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (!is_dumping_aec_) {
     // Start dumping AEC when we are not dumping.
-    if (apm()->StartDebugRecording(filename.c_str(), -1) !=
+    if (apm()->StartDebugRecording(filename.c_str(), -1, &worker_queue_) !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR1(StartDebugRecording, filename.c_str());
     } else {
       is_dumping_aec_ = true;
     }
   }
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 1599 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE