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;
   }
 }
+
 }  // namespace
 
 PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
                              int payload_type)
     : active_(false),
       num_sending_modules_(1),
       rtp_modules_(rtp_modules),
       payload_type_(payload_type) {
   UpdateModuleSendingState();
 }
@@ skipping 11 matching lines @@
     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::SetSendStreams(const std::vector<VideoStream>& streams) {
+  RTC_DCHECK_LE(streams.size(), rtp_modules_.size());
   rtc::CritScope lock(&crit_);
-  num_sending_modules_ = num_sending_modules;
+  num_sending_modules_ = streams.size();
+  streams_ = streams;

[pbos-webrtc, 2016/05/02 00:45:44]

Perhaps this one should also reallocate bitrates, feel free to add a TODO here.

[perkj_webrtc, 2016/05/02 12:33:19]

Done.

   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_);
-  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]);
+  RTC_DCHECK_LE(streams_.size(), rtp_modules_.size());
+
+  // Allocate the bandwidth between the streams.

[pbos-webrtc, 2016/05/02 00:45:44]

I think this should be target_bitrate_bps for all streams except max, perhaps
this could go on top of https://codereview.webrtc.org/1913073002/ to have one
allocation function.

I don't think this is a regression though, but please put a TODO and feel free
to follow up. :)

[perkj_webrtc, 2016/05/02 12:33:19]

Done.

+  // TODO(perkj): Why are we allocating bitrates here as well as in the
+  // VCM. It should be done once, in the BitrateAllocator?
+  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);
   }
 }
 
 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