Add task queue to Call.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/task_queue.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
@@ skipping 22 matching lines @@
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    rtc::TaskQueue* worker_queue,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator)
-    : config_(config),
+    : worker_queue_(worker_queue),
+      config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(congestion_controller);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->RegisterSenderCongestionControlObjects(
@@ skipping 27 matching lines @@
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
     RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
-    bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
-                                    config_.max_bitrate_kbps * 1000, 0, true);
+    rtc::Event thread_sync_event(false /* manual_reset */, false);
+    worker_queue_->PostTask([this, &thread_sync_event] {
+      bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
+                                      config_.max_bitrate_kbps * 1000, 0, true);
+      thread_sync_event.Set();
+    });
+    thread_sync_event.Wait(rtc::Event::kForever);
   }
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StartSend(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
   }
 }
 
 void AudioSendStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  bitrate_allocator_->RemoveObserver(this);
+  rtc::Event thread_sync_event(false /* manual_reset */, false);
+  worker_queue_->PostTask([this, &thread_sync_event] {
+    bitrate_allocator_->RemoveObserver(this);
+    thread_sync_event.Set();
+  });
+  thread_sync_event.Wait(rtc::Event::kForever);
+
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StopSend(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioSendStream::Stop failed with error: " << error;
   }
 }
 
 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
                                          int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 126 matching lines @@
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc