Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender.

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 365 matching lines @@
           this,
           config.post_encode_callback,
           &stats_proxy_),
       vie_encoder_(num_cpu_cores,
                    module_process_thread_,
                    &stats_proxy_,
                    &overuse_detector_),
       encoder_feedback_(Clock::GetRealTimeClock(),
                         config.rtp.ssrcs,
                         &vie_encoder_),
+      post_encode_protection_(Clock::GetRealTimeClock(), this),
       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 14 matching lines @@
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(remb_);
 
 
   // RTP/RTCP initialization.
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     module_process_thread_->RegisterModule(rtp_rtcp);
     congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp);
   }
 
-  video_sender_->RegisterProtectionCallback(this);
-
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(id, 1);
     RTC_DCHECK_LE(id, 14);
     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
                           StringToRtpExtensionType(extension), id));
@@ skipping 127 matching lines @@
       // vie_encoder_.SetEncoder must be called before this.
       if (config_.suspend_below_min_bitrate)
         video_sender_->SuspendBelowMinBitrate();
 
       // Clear stats for disabled layers.
       for (size_t i = encoder_settings->streams.size();
            i < config_.rtp.ssrcs.size(); ++i) {
         stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
       }
 
+      post_encode_protection_.SetEncodingData(
+          encoder_settings->video_codec.startBitrate * 1000,
+          encoder_settings->video_codec.width,
+          encoder_settings->video_codec.height,
+          encoder_settings->video_codec.maxFramerate,
+          encoder_settings->streams.size(), payload_router_.MaxPayloadLength());

[stefan-webrtc, 2016/05/19 12:52:40]

num_layers should be the number of temporal layers. Now I think you are setting
it to number of spatial layers?

[perkj_webrtc, 2016/06/01 20:55:23]

Done.

+
       // 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)) {
       // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should
@@ skipping 36 matching lines @@
   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);
+  post_encode_protection_.UpdateWithEncodedData(encoded_image);
   int32_t return_value = payload_router_.Encoded(
       encoded_image, codec_specific_info, fragmentation);
 
   if (kEnableFrameRecording) {
     int layer = codec_specific_info->codecType == kVideoCodecVP8
                     ? codec_specific_info->codecSpecific.VP8.simulcastIdx
                     : 0;
     IvfFileWriter* file_writer;
     {
       if (file_writers_[layer] == nullptr) {
@@ skipping 59 matching lines @@
     rtp_rtcp->SetStorePacketsStatus(
         enable_protection_nack || congestion_controller_->pacer(),
         kMinSendSidePacketHistorySize);
     // Set FEC.
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
       rtp_rtcp->SetGenericFECStatus(enable_protection_fec, payload_type_red,
                                     payload_type_fec);
     }
   }
 
-  vie_encoder_.SetProtectionMethod(enable_protection_nack,
-                                   enable_protection_fec);
+  post_encode_protection_.SetProtectionMethod(enable_protection_fec,
+                                              enable_protection_nack);
 }
 
 void VideoSendStream::ConfigureSsrcs() {
   // Configure regular SSRCs.
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
     RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
     rtp_rtcp->SetSSRC(ssrc);
 
     // Restore RTP state if previous existed.
@@ skipping 57 matching lines @@
 }
 
 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);
+  // Get the encoder target rate. It is the estimated network rate -
+  // protection overhead.
+  uint32_t encoder_target_rate =
+      post_encode_protection_.SetTargetRates(bitrate_bps, fraction_loss, rtt);
+
+  vie_encoder_.OnBitrateUpdated(encoder_target_rate, 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