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_;

[the sun, 2017/05/22 22:38:00]

On some platforms this means we're adding yet another thread. What other options
have you considered?

[aleloi, 2017/05/23 11:05:49]

The file logging has to be done on *some* thread different from the real-time
one. Using a TaskQueue seemed the best solution, because

* using a custom (possibly existing) thread would require adding signaling
mechanisms for passing the thread down to APM/AecDump and communicating with it.
Compare the RTC event log:
https://cs.chromium.org/chromium/src/third_party/webrtc/logging/rtc_event_log...
* using a task queue could limit the total number of threads if the task queue
was passed from elsewhere. Other tasks could be run on the same task queue.

The current plan is to move the task queue in PeerConnectionFactory
created/destroyed by the signaling thread. This TQ could then be used for
similar tasks in different sub-modules (I think RTC event logging).

There is a dependent CL which attempts to do just that:
https://codereview.webrtc.org/2888303003. A few problems still remain, mainly
the case of using sub-components of WebRTC separately from
PeerConnectionFactory. That happens in internal projects.

[the sun, 2017/05/23 20:47:29]

Sorry about the confusion - that's not what I'm suggesting. Rather something
like what you reference below - sharing a thread/task queue for low prio tasks.

I'm looping in Danil and Tommi for thoughts on how TaskQueue is implemented -
IIRC in Chromium it is overridden with the Chromium impl - does that mean that
there is more thread pooling going on behind the scenes than is the case with
the pure WebRTC impl?

+
   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_