Variable audio 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.
  */
 
@@ skipping 41 matching lines @@
   // TODO(solenberg): Encoder config.
   ss << ", cng_payload_type: " << cng_payload_type;
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
-    CongestionController* congestion_controller)
-    : config_(config), audio_state_(audio_state) {
+    CongestionController* congestion_controller,
+    BitrateAllocator* bitrate_allocator)
+    : config_(config),
+      audio_state_(audio_state),
+      bitrate_allocator_(bitrate_allocator) {
   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_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),
@@ skipping 23 matching lines @@
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
+    RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
+    bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
+                                    config_.max_bitrate_kbps * 1000, 0, true);
+  }
+
   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());
+  bitrate_allocator_->RemoveObserver(this);
   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;
   }
 }
 
 bool AudioSendStream::SendTelephoneEvent(int payload_type, int event,
                                          int duration_ms) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 94 matching lines @@
 }
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // 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) {
+  RTC_DCHECK_GE(bitrate_bps,
+                static_cast<uint32_t>(config_.min_bitrate_kbps * 1000));
+  // 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_kbps * 1000;
+  if (bitrate_bps > max_bitrate_bps)
+    bitrate_bps = max_bitrate_bps;
+
+  channel_proxy_->SetBitrate(bitrate_bps);
+
+  // 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_;
 }
 
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc