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 21 matching lines @@
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/engine/adm_helpers.h"
 #include "webrtc/media/engine/apm_helpers.h"
 #include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 
 namespace cricket {
 namespace {
 
 constexpr size_t kMaxUnsignaledRecvStreams = 1;
@@ skipping 506 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) !=
-      webrtc::AudioProcessing::kNoError) {
-    LOG_RTCERR0(StartDebugRecording);
-    fclose(aec_dump_file_stream);
-    return false;
-  }
+  apm()->StartDebugRecording(webrtc::AecDumpFactory::Create(
+      aec_dump_file_stream, max_size_bytes, &worker_queue_));
   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) !=
-        webrtc::AudioProcessing::kNoError) {
-      LOG_RTCERR1(StartDebugRecording, filename.c_str());
-    } else {
-      is_dumping_aec_ = true;
-    }
+    apm()->StartDebugRecording(
+        webrtc::AecDumpFactory::Create(filename.c_str(), -1, &worker_queue_));
+    is_dumping_aec_ = true;
   }
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
-    if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
-      LOG_RTCERR0(StopDebugRecording);
-    }
+    apm()->StopDebugRecording();
     is_dumping_aec_ = false;
   }
 }
 
 int WebRtcVoiceEngine::CreateVoEChannel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->base()->CreateChannel(channel_config_);
 }
 
 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
@@ skipping 1584 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE