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

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
 #include "webrtc/config.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/apm_helpers.h"
 #include "webrtc/media/engine/webrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/pc/channel.h"
@@ skipping 68 matching lines @@
   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   void StartAecDump(const std::string& filename);
   int CreateVoEChannel();
+
+  rtc::TaskQueue low_priority_worker_queue_;
+
   webrtc::AudioDeviceModule* adm();
   webrtc::AudioProcessing* apm();
   webrtc::voe::TransmitMixer* transmit_mixer();
 
   AudioCodecs CollectCodecs(
       const std::vector<webrtc::AudioCodecSpec>& specs) const;
 
   rtc::ThreadChecker signal_thread_checker_;
   rtc::ThreadChecker worker_thread_checker_;
 
@@ skipping 171 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
       send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_