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

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 210313896004d2efd39484b2aaa52a43138b94d1..020d74d82636fd4c02e868b2f750ba52076f3437 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -38,6 +38,7 @@
 #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"
@@ -228,7 +229,8 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     VoEWrapper* voe_wrapper)
-    : adm_(adm),
+    : low_priority_worker_queue_("rtc-low-prio", rtc::TaskQueue::Priority::LOW),
+      adm_(adm),
       encoder_factory_(encoder_factory),
       decoder_factory_(decoder_factory),
       voe_wrapper_(voe_wrapper) {
@@ -687,46 +689,28 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
 bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
                                      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.";
+  auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes,
+                                                 &low_priority_worker_queue_);
+  if (!aec_dump) {
     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;
-  }
-  is_dumping_aec_ = true;
+  apm()->AttachAecDump(std::move(aec_dump));
   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;
-    }
+
+  auto aec_dump =
+      webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_);
+  if (aec_dump) {
+    apm()->AttachAecDump(std::move(aec_dump));
   }
 }
 
 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);
-    }
-    is_dumping_aec_ = false;
-  }
+  apm()->DetachAecDump();
 }
 
 int WebRtcVoiceEngine::CreateVoEChannel() {