Added logging for audio send/receive stream configs.

webrtc/call/rtc_event_log.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.
  */
 
@@ skipping 35 matching lines @@
   ~RtcEventLogImpl() override;
 
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override;
   bool StartLogging(rtc::PlatformFile platform_file,
                     int64_t max_size_bytes) override;
   void StopLogging() override;
   void LogVideoReceiveStreamConfig(
       const VideoReceiveStream::Config& config) override;
   void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override;
+  void LogAudioReceiveStreamConfig(
+      const AudioReceiveStream::Config& config) override;
+  void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override;
   void LogRtpHeader(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override;
   void LogRtcpPacket(PacketDirection direction,
                      MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override;
   void LogAudioPlayout(uint32_t ssrc) override;
   void LogBwePacketLossEvent(int32_t bitrate,
@@ skipping 212 matching lines @@
     sender_config->add_rtx_ssrcs(rtx_ssrc);
   }
   sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
 
   rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
   encoder->set_name(config.encoder_settings.payload_name);
   encoder->set_payload_type(config.encoder_settings.payload_type);
   StoreEvent(&event);
 }
 
+void RtcEventLogImpl::LogAudioReceiveStreamConfig(
+    const AudioReceiveStream::Config& config) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
+
+  rtclog::AudioReceiveConfig* receiver_config =
+      event->mutable_audio_receiver_config();
+  receiver_config->set_remote_ssrc(config.rtp.remote_ssrc);
+  receiver_config->set_local_ssrc(config.rtp.local_ssrc);
+
+  for (const auto& e : config.rtp.extensions) {
+    rtclog::RtpHeaderExtension* extension =
+        receiver_config->add_header_extensions();
+    extension->set_name(e.uri);
+    extension->set_id(e.id);
+  }
+  StoreEvent(&event);
+}
+
+void RtcEventLogImpl::LogAudioSendStreamConfig(
+    const AudioSendStream::Config& config) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
+
+  rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config();
+
+  sender_config->set_ssrc(config.rtp.ssrc);
+
+  for (const auto& e : config.rtp.extensions) {
+    rtclog::RtpHeaderExtension* extension =
+        sender_config->add_header_extensions();
+    extension->set_name(e.uri);
+    extension->set_id(e.id);
+  }
+
+  StoreEvent(&event);
+}
+
 void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
                                    MediaType media_type,
                                    const uint8_t* header,
                                    size_t packet_length) {
   // Read header length (in bytes) from packet data.
   if (packet_length < 12u) {
     return;  // Don't read outside the packet.
   }
   const bool x = (header[0] & 0x10) != 0;
   const uint8_t cc = header[0] & 0x0f;
@@ skipping 138 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc