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 b031762b4cecd173e21766e4ab3f91c070a37fd9..d31e7a1903b1f492059041102a31ca25586fd44a 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,
@@ -56,7 +74,6 @@ AudioSendStream::AudioSendStream(
   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_->SetRtcEventLog(event_log);
@@ -102,6 +119,8 @@ 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();
@@ -121,6 +140,7 @@ 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);
@@ -240,6 +260,10 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
 
   channel_proxy_->SetBitrate(bitrate_bps);
 
+  if (first_update_bitrate_()) {
+    AdaptCodec();
+  }
+
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   return 0;
@@ -382,5 +406,14 @@ bool AudioSendStream::SetupSendCodec() {
   return true;
 }
 
+void AudioSendStream::AdaptCodec() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  channel_proxy_->AdaptCodec();
+  constexpr uint32_t kAdaptCodecIntervalMs = 200;
+  worker_queue_->PostDelayedTask(
+      std::unique_ptr<rtc::QueuedTask>(new AdaptCodecTask(weak_ptr_)),
+      kAdaptCodecIntervalMs);
+}
+
 }  // namespace internal
 }  // namespace webrtc