Video/Screenshare loopback tool.

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 348 matching lines @@
   rtc::TaskQueue worker_queue_;
 
   // The config mask set by SetBitrateConfigMask.
   // 0 <= min <= start <= max
   Config::BitrateConfigMask bitrate_config_mask_;
 
   // The config set by SetBitrateConfig.
   // min >= 0, start != 0, max == -1 || max > 0
   Config::BitrateConfig base_bitrate_config_;
 
+  void PrintDebugStuff() override;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 std::string Call::Stats::ToString(int64_t time_ms) const {
   std::stringstream ss;
   ss << "Call stats: " << time_ms << ", {";
   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
   ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
@@ skipping 1054 matching lines @@
   }
   // For audio, we only support send side BWE.
   if (media_type == MediaType::VIDEO ||
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
+void Call::PrintDebugStuff() {
+  transport_send_->send_side_cc()->PrintDebugStuff();
+  print_ = true;
+}
+
 }  // namespace internal
 
 }  // namespace webrtc