Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'.

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 20 matching lines @@
 #include "webrtc/base/trace_event.h"
 #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_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 170 matching lines @@
   audio_state_ =
       webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(
     webrtc::AudioDeviceModule* adm,
     const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
-    : low_priority_worker_queue_("rtc-low-prio", rtc::TaskQueue::Priority::LOW),
-      adm_(adm),
+    : adm_(adm),
       encoder_factory_(encoder_factory),
       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();
 
@@ skipping 438 matching lines @@
 void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   auto it = std::find(channels_.begin(), channels_.end(), channel);
   RTC_DCHECK(it != channels_.end());
   channels_.erase(it);
 }
 
 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
                                      int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes,
-                                                 &low_priority_worker_queue_);
-  if (!aec_dump) {
+  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;
   }
-  apm()->AttachAecDump(std::move(aec_dump));
+  StopAecDump();
+  if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) !=
+      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());
-
-  auto aec_dump =
-      webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_);
-  if (aec_dump) {
-    apm()->AttachAecDump(std::move(aec_dump));
+  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;
+    }
   }
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  apm()->DetachAecDump();
+  if (is_dumping_aec_) {
+    // Stop dumping AEC when we are dumping.
+    if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
+      LOG_RTCERR0(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() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(adm_);
@@ skipping 1617 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE