Remove PayloadRouter dependency from ViEEncoder

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 66 matching lines @@
       rtp->codec = kRtpVideoH264;
       return;
     case kVideoCodecGeneric:
       rtp->codec = kRtpVideoGeneric;
       rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
       return;
     default:
       return;
   }
 }
+
+std::vector<uint32_t> AllocateStreamBitrates(
+    uint32_t total_bitrate,
+    const SimulcastStream* stream_configs,
+    size_t number_of_streams) {
+  if (number_of_streams == 0) {
+    std::vector<uint32_t> stream_bitrates(1, 0);
+    stream_bitrates[0] = total_bitrate;
+    return stream_bitrates;
+  }
+  std::vector<uint32_t> stream_bitrates(number_of_streams, 0);
+  uint32_t bitrate_remainder = total_bitrate;
+  for (size_t i = 0; i < stream_bitrates.size() && bitrate_remainder > 0; ++i) {
+    if (stream_configs[i].maxBitrate * 1000 > bitrate_remainder) {
+      stream_bitrates[i] = bitrate_remainder;
+    } else {
+      stream_bitrates[i] = stream_configs[i].maxBitrate * 1000;
+    }
+    bitrate_remainder -= stream_bitrates[i];
+  }
+  return stream_bitrates;
+}
+
 }  // namespace
 
 PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
                              int payload_type)
     : active_(false),
+      codec_(),
       num_sending_modules_(1),
       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;
 }
 
 void PayloadRouter::set_active(bool active) {
   rtc::CritScope lock(&crit_);
   if (active_ == active)
     return;
   active_ = active;
   UpdateModuleSendingState();
 }
 
 bool PayloadRouter::active() {
   rtc::CritScope lock(&crit_);
   return active_ && !rtp_modules_.empty();
 }
 
-void PayloadRouter::SetSendingRtpModules(size_t num_sending_modules) {
-  RTC_DCHECK_LE(num_sending_modules, rtp_modules_.size());
+void PayloadRouter::SetSendCodec(const VideoCodec& codec) {
+  RTC_DCHECK_LE(codec.numberOfSimulcastStreams, rtp_modules_.size());
   rtc::CritScope lock(&crit_);
-  num_sending_modules_ = num_sending_modules;
+  num_sending_modules_ = codec.numberOfSimulcastStreams;
+  codec_ = codec;
   UpdateModuleSendingState();
 }
 
 void PayloadRouter::UpdateModuleSendingState() {
   for (size_t i = 0; i < num_sending_modules_; ++i) {
     rtp_modules_[i]->SetSendingStatus(active_);
     rtp_modules_[i]->SetSendingMediaStatus(active_);
   }
   // Disable inactive modules.
   for (size_t i = num_sending_modules_; i < rtp_modules_.size(); ++i) {
@@ skipping 24 matching lines @@
   if (rtp_video_header.simulcastIdx >= num_sending_modules_)
     return -1;
   stream_idx = rtp_video_header.simulcastIdx;
 
   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::SetTargetSendBitrates(
-    const std::vector<uint32_t>& stream_bitrates) {
+void PayloadRouter::SetTargetSendBitrate(uint32_t bitrate_bps) {
   rtc::CritScope lock(&crit_);
+  // Allocate the bandwidth between the streams.
+  std::vector<uint32_t> stream_bitrates = AllocateStreamBitrates(
+      bitrate_bps, codec_.simulcastStream, codec_.numberOfSimulcastStreams);
+
   RTC_DCHECK_LE(stream_bitrates.size(), rtp_modules_.size());
   for (size_t i = 0; i < stream_bitrates.size(); ++i) {
     rtp_modules_[i]->SetTargetSendBitrate(stream_bitrates[i]);
   }
 }
 
 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