Remove PayloadRouter dependency from ViEEncoder

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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 354 matching lines @@
       remb_(remb),
       encoder_thread_(EncoderThreadFunction, this, "EncoderThread"),
       encoder_wakeup_event_(false, false),
       stop_encoder_thread_(0),
       overuse_detector_(
           Clock::GetRealTimeClock(),
           GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
           this,
           config.post_encode_callback,
           &stats_proxy_),
-      vie_encoder_(
-          num_cpu_cores,
-          config_.rtp.ssrcs,
-          module_process_thread_,
-          &stats_proxy_,
-          config.pre_encode_callback,
-          &overuse_detector_,
-          congestion_controller_->pacer(),
-          &payload_router_,
-          config.post_encode_callback ? &encoded_frame_proxy_ : nullptr),
+      vie_encoder_(num_cpu_cores,
+                   config_.rtp.ssrcs,
+                   module_process_thread_,
+                   &stats_proxy_,
+                   config.pre_encode_callback,
+                   &overuse_detector_,
+                   congestion_controller_->pacer()),
       video_sender_(vie_encoder_.video_sender()),
       bandwidth_observer_(congestion_controller_->GetBitrateController()
                               ->CreateRtcpBandwidthObserver()),
       rtp_rtcp_modules_(CreateRtpRtcpModules(
           config.send_transport,
           &encoder_feedback_,
           bandwidth_observer_.get(),
           congestion_controller_->GetTransportFeedbackObserver(),
           call_stats_->rtcp_rtt_stats(),
           congestion_controller_->pacer(),
@@ skipping 114 matching lines @@
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     rtp_rtcp->IncomingRtcpPacket(packet, length);
   return true;
 }
 
 void VideoSendStream::Start() {
   if (payload_router_.active())
     return;
   TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
-  vie_encoder_.Pause();
   payload_router_.set_active(true);
   // Was not already started, trigger a keyframe.
   vie_encoder_.SendKeyFrame();
-  vie_encoder_.Restart();
+  vie_encoder_.Start();
 }
 
 void VideoSendStream::Stop() {
   if (!payload_router_.active())
     return;
   TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
-  // TODO(pbos): Make sure the encoder stops here.
+  vie_encoder_.Pause();
   payload_router_.set_active(false);
 }
 
 VideoCaptureInput* VideoSendStream::Input() {
   return &input_;
 }
 
 bool VideoSendStream::EncoderThreadFunction(void* obj) {
   static_cast<VideoSendStream*>(obj)->EncoderProcess();
   // We're done, return false to abort.
@@ skipping 17 matching lines @@
         encoder_settings = pending_encoder_settings_;
         pending_encoder_settings_ = rtc::Optional<EncoderSettings>();
       }
     }
     if (encoder_settings) {
       encoder_settings->video_codec.startBitrate =
           bitrate_allocator_->AddObserver(
               this, encoder_settings->video_codec.minBitrate * 1000,
               encoder_settings->video_codec.maxBitrate * 1000) /
           1000;
+
+      payload_router_.SetSendStreams(encoder_settings->streams);
       vie_encoder_.SetEncoder(encoder_settings->video_codec,
-                              encoder_settings->min_transmit_bitrate_bps);
+                              encoder_settings->min_transmit_bitrate_bps,
+                              payload_router_.MaxPayloadLength(), this);
       if (config_.suspend_below_min_bitrate) {
         video_sender_->SuspendBelowMinBitrate();
         bitrate_allocator_->EnforceMinBitrate(false);
       }
       // We might've gotten new settings while configuring the encoder settings,
       // restart from the top to see if that's the case before trying to encode
       // a frame (which might correspond to the last frame size).
       encoder_wakeup_event_.Set();
       continue;
     }
 
     VideoFrame frame;
     if (input_.GetVideoFrame(&frame))
       vie_encoder_.EncodeVideoFrame(frame);
   }
   vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
 }
 
 void VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size());
   VideoCodec video_codec = VideoEncoderConfigToVideoCodec(
       config, config_.encoder_settings.payload_name,
       config_.encoder_settings.payload_type);
   {
     rtc::CritScope lock(&encoder_settings_crit_);
     pending_encoder_settings_ = rtc::Optional<EncoderSettings>(
-        {video_codec, config.min_transmit_bitrate_bps});
+        {video_codec, config.min_transmit_bitrate_bps, config.streams});
   }
   encoder_wakeup_event_.Set();
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::OveruseDetected() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
 }
 
 void VideoSendStream::NormalUsage() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
 }
 
+int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image,
+                                 const CodecSpecificInfo* codec_specific_info,
+                                 const RTPFragmentationHeader* fragmentation) {
+  // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|;
+  encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info,
+                               fragmentation);
+  return payload_router_.Encoded(encoded_image, codec_specific_info,
+                                 fragmentation);
+}
+
 void VideoSendStream::ConfigureProtection() {
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
   bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1;
   // Payload types without picture ID cannot determine that a stream is complete
   // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is
   // a waste of bandwidth since FEC packets still have to be transmitted. Note
   // that this is not the case with FLEXFEC.
   if (enable_protection_nack &&
       !PayloadTypeSupportsSkippingFecPackets(
@@ skipping 101 matching lines @@
   return rtp_states;
 }
 
 int VideoSendStream::GetPaddingNeededBps() const {
   return vie_encoder_.GetPaddingNeededBps();
 }
 
 void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                        uint8_t fraction_loss,
                                        int64_t rtt) {
+  payload_router_.SetTargetSendBitrate(bitrate_bps);
   vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
 }
 
 int VideoSendStream::ProtectionRequest(const FecProtectionParams* delta_params,
                                        const FecProtectionParams* key_params,
                                        uint32_t* sent_video_rate_bps,
                                        uint32_t* sent_nack_rate_bps,
                                        uint32_t* sent_fec_rate_bps) {
   *sent_video_rate_bps = 0;
   *sent_nack_rate_bps = 0;
   *sent_fec_rate_bps = 0;
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     uint32_t not_used = 0;
     uint32_t module_video_rate = 0;
     uint32_t module_fec_rate = 0;
     uint32_t module_nack_rate = 0;
     rtp_rtcp->SetFecParameters(delta_params, key_params);
     rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc