Signal target bitrate only for screenshare streams

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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 599 matching lines @@
       &send_stream_, &thread_sync_event_, &stats_proxy_, vie_encoder_.get(),
       module_process_thread, call_stats, congestion_controller, packet_router,
       bitrate_allocator, send_delay_stats, remb, event_log, &config_,
       encoder_config.max_bitrate_bps, suspended_ssrcs)));
 
   // Wait for ConstructionTask to complete so that |send_stream_| can be used.
   // |module_process_thread| must be registered and deregistered on the thread
   // it was created on.
   thread_sync_event_.Wait(rtc::Event::kForever);
   send_stream_->RegisterProcessThread(module_process_thread);
-  vie_encoder_->SetBitrateObserver(send_stream_.get());
+  // TODO(sprang): Enable this also for regular video calls if it works well.
+  if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen) {
+    // Only signal target bitrate for screenshare streams, for now.
+    vie_encoder_->SetBitrateObserver(send_stream_.get());
+  }
   vie_encoder_->RegisterProcessThread(module_process_thread);
 
   ReconfigureVideoEncoder(std::move(encoder_config));
 }
 
 VideoSendStream::~VideoSendStream() {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   RTC_DCHECK(!send_stream_);
 }
 
@@ skipping 663 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc