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:
+  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,
     rtc::TaskQueue* worker_queue,
     PacketRouter* packet_router,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats)
     : 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());
   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);
   channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->RegisterSenderCongestionControlObjects(
@@ skipping 31 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));
+      weak_ptr_ = weak_ptr_factory_->GetWeakPtr();
       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);
+    adapt_codec_task_started_ = false;
     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 90 matching lines @@
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 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
   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
   const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_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;
 }
 
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;
 }
 
@@ skipping 122 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();
+  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