Log Call {audio, video} stream deletions.

webrtc/video/video_send_stream.cc

Index: webrtc/video/video_send_stream.cc
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index d55adf01252b5e4f368718c691e1571eb862c66a..f90e4877675bca422ce158b35ec3bb6858669df4 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -117,6 +117,7 @@ VideoSendStream::VideoSendStream(
       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_,
@@ -194,6 +195,7 @@ VideoSendStream::VideoSendStream(
 }
 
 VideoSendStream::~VideoSendStream() {
+  LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString();
   vie_channel_->RegisterSendFrameCountObserver(nullptr);
   vie_channel_->RegisterSendBitrateObserver(nullptr);
   vie_channel_->RegisterRtcpPacketTypeCounterObserver(nullptr);
@@ -457,6 +459,10 @@ int64_t VideoSendStream::GetRtt() const {
   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.