Update smoothed bitrate.

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index aff605fe6a6845b995fd2082d458d8e5b9f45f4c..e30f1631b85839f3502d64a0612e0b468b6f5138 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -11,6 +11,7 @@
 #include "webrtc/audio/audio_send_stream.h"
 
 #include <string>
+#include <utility>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
@@ -41,6 +42,28 @@ bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
 }  // 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,
@@ -105,6 +128,13 @@ void AudioSendStream::Start() {
     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();
@@ -124,6 +154,7 @@ void AudioSendStream::Stop() {
   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);
@@ -386,5 +417,10 @@ bool AudioSendStream::SetupSendCodec() {
   return true;
 }
 
+void AudioSendStream::AdaptCodec() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  channel_proxy_->AdaptCodec();
+}
+
 }  // namespace internal
 }  // namespace webrtc