Test submission of complete AEC-dump refactoring.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 8fcda89962ee768153b9d6337a862309b7e67aab..57f06da3c175220e698d3b01b73a9c2e35d15d26 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -565,7 +565,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
     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);
@@ -999,7 +1002,8 @@ bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile 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);
@@ -1013,7 +1017,7 @@ 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 {