Wire up BitrateAllocation to be sent as RTCP TargetBitrate

webrtc/video/payload_router.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 83 matching lines @@
       payload_type_(payload_type) {
 }
 
 PayloadRouter::~PayloadRouter() {}
 
 size_t PayloadRouter::DefaultMaxPayloadLength() {
   const size_t kIpUdpSrtpLength = 44;
   return IP_PACKET_SIZE - kIpUdpSrtpLength;
 }
 
-void PayloadRouter::set_active(bool active) {
+void PayloadRouter::SetActive(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)
     return;
   active_ = active;
 
   for (auto& module : rtp_modules_) {
     module->SetSendingStatus(active_);
     module->SetSendingMediaStatus(active_);
   }
 }
 
-bool PayloadRouter::active() {
+bool PayloadRouter::IsActive() {
   rtc::CritScope lock(&crit_);
   return active_ && !rtp_modules_.empty();
 }
 
 EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
     const EncodedImage& encoded_image,
     const CodecSpecificInfo* codec_specific_info,
     const RTPFragmentationHeader* fragmentation) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK(!rtp_modules_.empty());
@@ skipping 24 matching lines @@
   size_t min_payload_length = DefaultMaxPayloadLength();
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < rtp_modules_.size(); ++i) {
     size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
     if (module_payload_length < min_payload_length)
       min_payload_length = module_payload_length;
   }
   return min_payload_length;
 }
 
+void PayloadRouter::OnBitrateAllocationUpdated(
+    const BitrateAllocation& bitrate) {
+  rtc::CritScope lock(&crit_);
+  if (IsActive()) {
+    if (rtp_modules_.size() == 1) {
+      // If spatial scalability is enabled, it is covered by a single stream.
+      rtp_modules_[0]->SetVideoBitrateAllocation(bitrate);
+    } else {
+      // Simulcast is in use, split the BitrateAllocation into one struct per
+      // rtp stream, moving over the temporal layer allocation.
+      for (size_t si = 0; si < rtp_modules_.size(); ++si) {
+        BitrateAllocation layer_bitrate;
+        for (int tl = 0; tl < kMaxTemporalStreams; ++tl)
+          layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl));
+        rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate);
+      }
+    }
+  }
+}
+
 }  // namespace webrtc