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..ea3c60e2cd4b1a6f0af056844b2c9c6546f71d8a 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 +42,23 @@ bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
 }  // namespace
 
 namespace internal {
+
+class AudioSendStream::AdaptCodecTask : public rtc::QueuedTask {
+ public:
+  explicit AdaptCodecTask(const rtc::WeakPtr<AudioSendStream>& send_stream)
+      : send_stream_(std::move(send_stream)) {}
+
+ private:
+  bool Run() override {
+    if (send_stream_) {
+      send_stream_->AdaptCodec();
+    }
+    return true;
+  }
+
+  rtc::WeakPtr<AudioSendStream> send_stream_;
+};
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -53,7 +71,8 @@ AudioSendStream::AudioSendStream(
     : worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
-      bitrate_allocator_(bitrate_allocator) {
+      bitrate_allocator_(bitrate_allocator),
+      adapt_codec_task_started_(false) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
@@ -105,6 +124,8 @@ 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));
+      weak_ptr_ = weak_ptr_factory_->GetWeakPtr();
       bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
                                       config_.max_bitrate_bps, 0, true);
       thread_sync_event.Set();
@@ -124,6 +145,8 @@ 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);
+    adapt_codec_task_started_ = false;
     thread_sync_event.Set();
   });
   thread_sync_event.Wait(rtc::Event::kForever);
@@ -234,6 +257,7 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt,
                                            int64_t probing_interval_ms) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
   RTC_DCHECK_GE(bitrate_bps,
                 static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
@@ -244,6 +268,12 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
 
   channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms);
 
+  if (!adapt_codec_task_started_) {
+    // Starts adapt codec task, which calls AdaptCodec on a timely base.
+    worker_queue_->PostTask([this]() { AdaptCodec(); });
+    adapt_codec_task_started_ = true;
+  }
+
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   return 0;
@@ -386,5 +416,14 @@ bool AudioSendStream::SetupSendCodec() {
   return true;
 }
 
+void AudioSendStream::AdaptCodec() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  channel_proxy_->AdaptCodec();
+  constexpr uint32_t kAdaptCodecIntervalMs = 200;

[minyue-webrtc, 2016/12/13 17:21:27]

oh, this, how about making it a config_ field?

[michaelt, 2016/12/14 11:31:21]

Done.

+  worker_queue_->PostDelayedTask(
+      std::unique_ptr<rtc::QueuedTask>(new AdaptCodecTask(weak_ptr_)),
+      kAdaptCodecIntervalMs);
+}
+
 }  // namespace internal
 }  // namespace webrtc