Log Call {audio, video} stream deletions.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 064749c61ebf226624b3a62ac536e8eac00fef19..7f7bde1a1241c2144629b503f0a2f1b84f70fb6a 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -48,6 +48,7 @@ AudioReceiveStream::AudioReceiveStream(
     : remote_bitrate_estimator_(remote_bitrate_estimator),
       config_(config),
       rtp_header_parser_(RtpHeaderParser::Create()) {
+  LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK(config.voe_channel_id != -1);
   RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
   RTC_DCHECK(rtp_header_parser_ != nullptr);
@@ -70,10 +71,18 @@ AudioReceiveStream::AudioReceiveStream(
   }
 }
 
+AudioReceiveStream::~AudioReceiveStream() {
+  LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
+}
+
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return webrtc::AudioReceiveStream::Stats();
 }
 
+const webrtc::AudioReceiveStream::Config AudioReceiveStream::config() const {
+  return config_;
+}
+
 void AudioReceiveStream::Start() {
 }