Log Call {audio, video} stream deletions.

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 99 matching lines @@
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
     : transport_adapter_(config.send_transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
       suspended_ssrcs_(suspended_ssrcs),
       module_process_thread_(module_process_thread),
       channel_group_(channel_group),
       channel_id_(channel_id),
       use_config_bitrate_(true),
       stats_proxy_(Clock::GetRealTimeClock(), config) {
+  LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
   RTC_DCHECK(!config_.rtp.ssrcs.empty());
   RTC_CHECK(channel_group->CreateSendChannel(
       channel_id_, &transport_adapter_, &stats_proxy_,
       config.pre_encode_callback, num_cpu_cores, config_));
   vie_channel_ = channel_group_->GetChannel(channel_id_);
   vie_encoder_ = channel_group_->GetEncoder(channel_id_);
 
   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;
@@ skipping 57 matching lines @@
     vie_encoder_->SuspendBelowMinBitrate();
 
   vie_channel_->RegisterSendChannelRtcpStatisticsCallback(&stats_proxy_);
   vie_channel_->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_);
   vie_channel_->RegisterRtcpPacketTypeCounterObserver(&stats_proxy_);
   vie_channel_->RegisterSendBitrateObserver(&stats_proxy_);
   vie_channel_->RegisterSendFrameCountObserver(&stats_proxy_);
 }
 
 VideoSendStream::~VideoSendStream() {
+  LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
   vie_channel_->RegisterSendFrameCountObserver(nullptr);
   vie_channel_->RegisterSendBitrateObserver(nullptr);
   vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
   vie_channel_->RegisterSendChannelRtpStatisticsCallback(nullptr);
   vie_channel_->RegisterSendChannelRtcpStatisticsCallback(nullptr);
 
   // Remove capture input (thread) so that it's not running after the current
   // channel is deleted.
   input_.reset();
 
@@ skipping 243 matching lines @@
   uint32_t jitter;
   int64_t rtt_ms;
   if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
                                           &extended_max_sequence_number,
                                           &jitter, &rtt_ms) == 0) {
     return rtt_ms;
   }
   return -1;
 }
 
+const VideoSendStream::Config& VideoSendStream::config() const {

[stefan-webrtc, 2015/10/14 10:30:16]

Not needed, right?

[pbos-webrtc, 2015/10/15 10:21:42]

Done.

+  return config_;
+}
+
 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
   if (video_codec.maxBitrate == 0) {
     // Unset max bitrate -> cap to one bit per pixel.
     video_codec.maxBitrate =
         (video_codec.width * video_codec.height * video_codec.maxFramerate) /
         1000;
   }
 
   if (video_codec.minBitrate < kViEMinCodecBitrate)
     video_codec.minBitrate = kViEMinCodecBitrate;
@@ skipping 26 matching lines @@
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc