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..c725e37477af5f36b6a9a18c3556475d12b94b3e 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -13,6 +13,7 @@
 #include <string>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 
@@ -48,6 +49,7 @@ 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 +72,18 @@ 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() {
 }