Log Call {audio, video} stream deletions.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #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"
 
 namespace webrtc {
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{remote_ssrc: " << remote_ssrc;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
@@ skipping 15 matching lines @@
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
       RemoteBitrateEstimator* remote_bitrate_estimator,
       const webrtc::AudioReceiveStream::Config& config)
     : 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);
   for (const auto& ext : config.rtp.extensions) {
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     RTC_DCHECK_GE(ext.id, 1);
     RTC_DCHECK_LE(ext.id, 14);
     if (ext.name == RtpExtension::kAudioLevel) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, ext.id));
     } else if (ext.name == RtpExtension::kAbsSendTime) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, ext.id));
     } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
       RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, ext.id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 }
 
+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() {
 }
 
 void AudioReceiveStream::Stop() {
 }
 
 void AudioReceiveStream::SignalNetworkState(NetworkState state) {
 }
 
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
@@ skipping 17 matching lines @@
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc