Update smoothed bitrate.

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 <utility>
 
 #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"
@@ skipping 10 matching lines @@
 namespace {
 
 constexpr char kOpusCodecName[] = "opus";
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (_stricmp(codec.plname, ref_name) == 0);
 }
 }  // namespace
 
 namespace internal {
+
+class AudioSendStream::AdaptCodecTask : public rtc::QueuedTask {
+ public:
+  AdaptCodecTask(rtc::WeakPtr<AudioSendStream> send_stream,
+                 int64_t adapt_codec_interval_ms)
+      : send_stream_(std::move(send_stream)),
+        adapt_codec_interval_ms_(adapt_codec_interval_ms) {}
+
+ private:
+  bool Run() override {
+    if (send_stream_) {

[ossu, 2016/12/20 13:58:09]

I'm not too well versed in the TaskQueue, but I expect as-is, this would just
continue to get called every adapt_codec_interval_ms_ even after Stop() is
called, right? Perhaps only re-post the task if(send_stream_), and otherwise
return true? It still doesn't seem foolproof though, for fast turn-around
between Start() and Stop().

[michaelt, 2016/12/20 15:13:03]

You are right. 
Done

I think a fast turn-around between Start() and Stop() should not be a problem
since the Start and Stop runs in the same task queue. Which means that the task
can not overlap. the send_stream_ pointer will be invalidated in stop. 

+      send_stream_->AdaptCodec();
+    }
+    rtc::TaskQueue::Current()->PostDelayedTask(
+        std::unique_ptr<rtc::QueuedTask>(this), adapt_codec_interval_ms_);
+    return false;
+  }
+
+  rtc::WeakPtr<AudioSendStream> send_stream_;
+  int64_t adapt_codec_interval_ms_;
+};
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     PacketRouter* packet_router,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
     : worker_queue_(worker_queue),
@@ skipping 44 matching lines @@
   channel_proxy_->SetRtcEventLog(nullptr);
   channel_proxy_->SetRtcpRttStats(nullptr);
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
     rtc::Event thread_sync_event(false /* manual_reset */, false);
     worker_queue_->PostTask([this, &thread_sync_event] {
+      weak_ptr_factory_.reset(new rtc::WeakPtrFactory<AudioSendStream>(this));
+
+      // Starts adapt codec task, which calls AdaptCodec on a timely base.
+      worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(
+          new AdaptCodecTask(weak_ptr_factory_->GetWeakPtr(),
+                             config_.adapt_codec_interval_ms)));
+
       bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
                                       config_.max_bitrate_bps, 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());
   rtc::Event thread_sync_event(false /* manual_reset */, false);
   worker_queue_->PostTask([this, &thread_sync_event] {
     bitrate_allocator_->RemoveObserver(this);
+    weak_ptr_factory_.reset(nullptr);
     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;
   }
 }
@@ skipping 242 matching lines @@
       // interaction between VAD and Opus FEC.
       if (codec->SetVADStatus(channel, true) != 0) {
         LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
   }
   return true;
 }
 
+void AudioSendStream::AdaptCodec() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  channel_proxy_->AdaptCodec();
+}
+
 }  // namespace internal
 }  // namespace webrtc