Variable audio 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 d08bfeaa1ef70c43de1de0fd57f1e252966542ec..17979d5760c0a2ca9ceecf9b7ad0cb5e545d3acc 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -59,8 +59,11 @@ 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());
@@ -104,6 +107,12 @@ AudioSendStream::~AudioSendStream() {
 
 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) {
@@ -113,6 +122,7 @@ void AudioSendStream::Start() {
 
 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) {
@@ -227,6 +237,24 @@ bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   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_;