Work-in-progress upload to add worker task queue to PC factory.

webrtc/media/base/mediaengine.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_BASE_MEDIAENGINE_H_
 #define WEBRTC_MEDIA_BASE_MEDIAENGINE_H_
 
 #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
 #include <CoreAudio/CoreAudio.h>
 #endif
 
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/sigslotrepeater.h"
+#include "webrtc/base/task_queue.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videocommon.h"
 
 #if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
 #define DISABLE_MEDIA_ENGINE_FACTORY
 #endif
 
+namespace rtc {
+class TaskQueue;
+}
+
 namespace webrtc {
 class AudioDeviceModule;
 class AudioMixer;
 class Call;
 }
 
 namespace cricket {
 
 struct RtpCapabilities {
   std::vector<webrtc::RtpExtension> header_extensions;
@@ skipping 65 matching lines @@
 // CompositeMediaEngine constructs a MediaEngine from separate
 // voice and video engine classes.
 template<class VOICE, class VIDEO>
 class CompositeMediaEngine : public MediaEngineInterface {
  public:
   CompositeMediaEngine(webrtc::AudioDeviceModule* adm,
                        const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
                            audio_encoder_factory,
                        const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
                            audio_decoder_factory,
-                       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
-      : voice_(adm, audio_encoder_factory, audio_decoder_factory, audio_mixer) {
-  }
+                       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
+                       rtc::TaskQueue* low_priority_worker_queue)
+      : owned_low_priority_worker_queue_(
+            low_priority_worker_queue
+                ? nullptr
+                : new rtc::TaskQueue("low_prio_worker_queue",
+                                     rtc::TaskQueue::Priority::LOW)),
+        voice_(adm,
+               audio_encoder_factory,
+               audio_decoder_factory,
+               audio_mixer,
+               low_priority_worker_queue
+                   ? owned_low_priority_worker_queue_.get()
+                   : low_priority_worker_queue) {}
   virtual ~CompositeMediaEngine() {}
   virtual bool Init() {
     video_.Init();
     return true;
   }
 
   virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
     return voice_.GetAudioState();
   }
   virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
@@ skipping 25 matching lines @@
   }
 
   virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
     return voice_.StartAecDump(file, max_size_bytes);
   }
 
   virtual void StopAecDump() {
     voice_.StopAecDump();
   }
 
+ private:
+  std::unique_ptr<rtc::TaskQueue> owned_low_priority_worker_queue_;
+
  protected:
   VOICE voice_;
   VIDEO video_;
 };
 
 enum DataChannelType { DCT_NONE = 0, DCT_RTP = 1, DCT_SCTP = 2, DCT_QUIC = 3 };
 
 class DataEngineInterface {
  public:
   virtual ~DataEngineInterface() {}
   virtual DataMediaChannel* CreateChannel(const MediaConfig& config) = 0;
   virtual const std::vector<DataCodec>& data_codecs() = 0;
 };
 
 webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIAENGINE_H_