Extract common simulcast logic from VP8 wrapper and simulcast adapter

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 67 matching lines @@
       return;
     case kVideoCodecGeneric:
       rtp->codec = kRtpVideoGeneric;
       rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
       return;
     default:
       return;
   }
 }
 
+VideoCodec DefaultCodec() {
+  VideoCodec codec;
+  memset(&codec, 0, sizeof(VideoCodec));
+  return codec;
+}
+
 }  // namespace
 
 PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
                              int payload_type)
     : active_(false),
       num_sending_modules_(1),
+      simulcast_state_(DefaultCodec()),
       rtp_modules_(rtp_modules),
       payload_type_(payload_type) {
   UpdateModuleSendingState();
 }
 
 PayloadRouter::~PayloadRouter() {}
 
 size_t PayloadRouter::DefaultMaxPayloadLength() {
   const size_t kIpUdpSrtpLength = 44;
   return IP_PACKET_SIZE - kIpUdpSrtpLength;
@@ skipping 59 matching lines @@
   return rtp_modules_[stream_idx]->SendOutgoingData(
       encoded_image._frameType, payload_type_, encoded_image._timeStamp,
       encoded_image.capture_time_ms_, encoded_image._buffer,
       encoded_image._length, fragmentation, &rtp_video_header);
 }
 
 void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
   rtc::CritScope lock(&crit_);
   RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
 
-  // TODO(sprang): Rebase https://codereview.webrtc.org/1913073002/ on top of
-  // this.
-  int bitrate_remainder = bitrate_bps;
-  for (size_t i = 0; i < streams_.size() && bitrate_remainder > 0; ++i) {
-    int stream_bitrate = 0;
-    if (streams_[i].max_bitrate_bps > bitrate_remainder) {
-      stream_bitrate = bitrate_remainder;
-    } else {
-      stream_bitrate = streams_[i].max_bitrate_bps;
-    }
-    bitrate_remainder -= stream_bitrate;
-    rtp_modules_[i]->SetTargetSendBitrate(stream_bitrate);
+  if (simulcast_state_.NumStreams() > 0) {
+    simulcast_state_.AllocateBitrate(bitrate_bps);
+    for (auto& stream : simulcast_state_.Streams())
+      rtp_modules_[stream.idx]->SetTargetSendBitrate(stream.allocated_rate_bps);
+  } else {
+    rtp_modules_[0]->SetTargetSendBitrate(bitrate_bps);
   }
 }
 
+void PayloadRouter::UpdateSimulcastState(const SimulcastState& state) {
+  rtc::CritScope lock(&crit_);
+  simulcast_state_ = state;
+}
+
 size_t PayloadRouter::MaxPayloadLength() const {
   size_t min_payload_length = DefaultMaxPayloadLength();
   rtc::CritScope lock(&crit_);
   for (size_t i = 0; i < num_sending_modules_; ++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;
 }
 
 }  // namespace webrtc