Let ViEEncoder tell VideoSendStream about reconfigurations.

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 72 matching lines @@
     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();
 }
 
 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();
+
+  for (auto& module : rtp_modules_) {
+    module->SetSendingStatus(active_);
+    module->SetSendingMediaStatus(active_);
+  }
 }
 
 bool PayloadRouter::active() {
   rtc::CritScope lock(&crit_);
   return active_ && !rtp_modules_.empty();
 }
 
-void PayloadRouter::SetSendStreams(const std::vector<VideoStream>& streams) {
-  RTC_DCHECK_LE(streams.size(), rtp_modules_.size());
-  rtc::CritScope lock(&crit_);
-  num_sending_modules_ = streams.size();
-  streams_ = streams;
-  // TODO(perkj): Should SetSendStreams also call SetTargetSendBitrate?
-  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) {
-    rtp_modules_[i]->SetSendingStatus(false);
-    rtp_modules_[i]->SetSendingMediaStatus(false);
-  }
-}
-
 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());
-  if (!active_ || num_sending_modules_ == 0)
+  if (!active_)
     return Result(Result::ERROR_SEND_FAILED);
 
   int stream_index = 0;
 
   RTPVideoHeader rtp_video_header;
   memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
   if (codec_specific_info)
     CopyCodecSpecific(codec_specific_info, &rtp_video_header);
   rtp_video_header.rotation = encoded_image.rotation_;
   rtp_video_header.playout_delay = encoded_image.playout_delay_;
-
-  RTC_DCHECK_LT(rtp_video_header.simulcastIdx, rtp_modules_.size());
-  // The simulcast index might actually be larger than the number of modules
-  // in case the encoder was processing a frame during a codec reconfig.
-  if (rtp_video_header.simulcastIdx >= num_sending_modules_)
-    return Result(Result::ERROR_SEND_FAILED);
   stream_index = rtp_video_header.simulcastIdx;
 
   uint32_t frame_id;
   int send_result = rtp_modules_[stream_index]->SendOutgoingData(
       encoded_image._frameType, payload_type_, encoded_image._timeStamp,
       encoded_image.capture_time_ms_, encoded_image._buffer,
       encoded_image._length, fragmentation, &rtp_video_header, &frame_id);
 
+  RTC_DCHECK_LT(rtp_video_header.simulcastIdx, rtp_modules_.size());
   if (send_result < 0)
     return Result(Result::ERROR_SEND_FAILED);
 
   return Result(Result::OK, frame_id);
 }
 
 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) {
+  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;
 }
 
 }  // namespace webrtc