Remove SendPacer from ViEEncoder

webrtc/call/call.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 648 matching lines @@
 }
 
 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
   if (first_packet_sent_ms_ == -1)
     first_packet_sent_ms_ = clock_->TimeInMilliseconds();
   congestion_controller_->OnSentPacket(sent_packet);
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
                             int64_t rtt_ms) {
-  uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
-      target_bitrate_bps, fraction_loss, rtt_ms);
+  bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
+                                       rtt_ms);
 
   int pad_up_to_bitrate_bps = 0;
   {
     ReadLockScoped read_lock(*send_crit_);
     // No need to update as long as we're not sending.
     if (video_send_streams_.empty())
       return;
 
     for (VideoSendStream* stream : video_send_streams_)
       pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
   }
-  // Allocated bitrate might be higher than bitrate estimate if enforcing min
-  // bitrate, or lower if estimate is higher than the sum of max bitrates, so
-  // set the pacer bitrate to the maximum of the two.
-  uint32_t pacer_bitrate_bps =
-      std::max(target_bitrate_bps, allocated_bitrate_bps);
+
+  uint32_t min_send_bitrate = bitrate_allocator_->GetMinSendBitrate();

[stefan-webrtc, 2016/04/29 10:48:38]

I think this can race with the call on 669. Probably doesn't matter, but if
there's no other benefit with this API we could return it as before.

[perkj_webrtc, 2016/05/02 11:29:10]

discussed off line. Reverting.

   {
     rtc::CritScope lock(&bitrate_crit_);
     // We only update these stats if we have send streams, and assume that
     // OnNetworkChanged is called roughly with a fixed frequency.
     estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
-    pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
+
+    // Allocated bitrate might be higher than bitrate estimate if enforcing min
+    // bitrate, or lower if estimate is higher than the sum of max bitrates, so
+    // set the pacer bitrate to the maximum of the two.
+    pacer_bitrate_sum_kbits_ +=
+        std::max(target_bitrate_bps, min_send_bitrate) / 1000;
     ++num_bitrate_updates_;
   }
-  congestion_controller_->UpdatePacerBitrate(
-      target_bitrate_bps / 1000,
-      PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
-      pad_up_to_bitrate_bps / 1000);
+  congestion_controller_->UpdatePaddingBitrate(pad_up_to_bitrate_bps);

[stefan-webrtc, 2016/04/29 10:48:38]

I think we still have to set the target_bitrate_bps.

[perkj_webrtc, 2016/05/02 11:29:09]

See new architecture.

+
+  // TODO(perkj): Refactor setting the minimum send bitrate.
+  // It changes when a stream is added or removed based on settings.
+  // Not when the network change.
+  congestion_controller_->SetMinimumSendBitrate(min_send_bitrate);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (voice_engine() == nullptr || sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
   // Find existing audio stream.
   const auto it = sync_stream_mapping_.find(sync_group);
@@ skipping 124 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc