Skip logging RTCP messages of type SDES and APP.

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.
  */
 
 #include "webrtc/call/rtc_event_log.h"
 
 #include <deque>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
 #else
 #include "webrtc/call/rtc_event_log.pb.h"
 #endif
@@ skipping 231 matching lines @@
   rtclog::Event event;
   event.set_timestamp_us(clock_->TimeInMicroseconds());
   event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
 
   rtclog::VideoReceiveConfig* receiver_config =
       event.mutable_video_receiver_config();
   receiver_config->set_remote_ssrc(config.rtp.remote_ssrc);
   receiver_config->set_local_ssrc(config.rtp.local_ssrc);
 
   receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode));
-
-  receiver_config->set_receiver_reference_time_report(
-      config.rtp.rtcp_xr.receiver_reference_time_report);
   receiver_config->set_remb(config.rtp.remb);
 
   for (const auto& kv : config.rtp.rtx) {
     rtclog::RtxMap* rtx = receiver_config->add_rtx_map();
     rtx->set_payload_type(kv.first);
     rtx->mutable_config()->set_rtx_ssrc(kv.second.ssrc);
     rtx->mutable_config()->set_rtx_payload_type(kv.second.payload_type);
   }
 
   for (const auto& e : config.rtp.extensions) {
@@ skipping 30 matching lines @@
         sender_config->add_header_extensions();
     extension->set_name(e.name);
     extension->set_id(e.id);
   }
 
   for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) {
     sender_config->add_rtx_ssrcs(rtx_ssrc);
   }
   sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
 
-  sender_config->set_c_name(config.rtp.c_name);
-
   rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
   encoder->set_name(config.encoder_settings.payload_name);
   encoder->set_payload_type(config.encoder_settings.payload_type);
   HandleEvent(&event);
 }
 
 void RtcEventLogImpl::LogRtpHeader(bool incoming,
                                    MediaType media_type,
                                    const uint8_t* header,
                                    size_t packet_length) {
@@ skipping 27 matching lines @@
 void RtcEventLogImpl::LogRtcpPacket(bool incoming,
                                     MediaType media_type,
                                     const uint8_t* packet,
                                     size_t length) {
   rtc::CritScope lock(&crit_);
   rtclog::Event rtcp_event;
   rtcp_event.set_timestamp_us(clock_->TimeInMicroseconds());
   rtcp_event.set_type(rtclog::Event::RTCP_EVENT);
   rtcp_event.mutable_rtcp_packet()->set_incoming(incoming);
   rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
-  rtcp_event.mutable_rtcp_packet()->set_packet_data(packet, length);
+
+  RTCPUtility::RtcpCommonHeader header;
+  const uint8_t* block_begin = packet;
+  const uint8_t* packet_end = packet + length;
+  RTC_DCHECK(length <= IP_PACKET_SIZE);
+  uint8_t buffer[IP_PACKET_SIZE];
+  uint32_t buffer_length = 0;
+  while (block_begin < packet_end) {
+    if (!RtcpParseCommonHeader(block_begin, packet_end - block_begin,
+                               &header)) {
+      break;  // Incorrect message header.
+    }
+    uint32_t block_size = header.BlockSize();
+    switch (header.packet_type) {
+      case RTCPUtility::PT_SR:
+        FALLTHROUGH();
+      case RTCPUtility::PT_RR:
+        FALLTHROUGH();
+      case RTCPUtility::PT_BYE:
+        FALLTHROUGH();
+      case RTCPUtility::PT_IJ:
+        FALLTHROUGH();
+      case RTCPUtility::PT_RTPFB:
+        FALLTHROUGH();
+      case RTCPUtility::PT_PSFB:
+        FALLTHROUGH();
+      case RTCPUtility::PT_XR:
+        // We log sender reports, receiver reports, bye messages
+        // inter-arrival jitter, third-party loss reports, payload-specific
+        // feedback and extended reports.
+        memcpy(buffer + buffer_length, block_begin, block_size);
+        buffer_length += block_size;
+        break;
+      case RTCPUtility::PT_SDES:
+        FALLTHROUGH();
+      case RTCPUtility::PT_APP:
+        FALLTHROUGH();
+      default:
+        // We don't log sender descriptions, application defined messages
+        // or message blocks of unknown type.
+        break;
+    }
+
+    block_begin += block_size;
+  }
+  rtcp_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
   HandleEvent(&rtcp_event);
 }
 
 void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
   rtc::CritScope lock(&crit_);
   rtclog::Event event;
   event.set_timestamp_us(clock_->TimeInMicroseconds());
   event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
   auto playout_event = event.mutable_audio_playout_event();
   playout_event->set_local_ssrc(ssrc);
@@ skipping 87 matching lines @@
 }
 
 #endif  // ENABLE_RTC_EVENT_LOG
 
 // RtcEventLog member functions.
 rtc::scoped_ptr<RtcEventLog> RtcEventLog::Create() {
   return rtc::scoped_ptr<RtcEventLog>(new RtcEventLogImpl());
 }
 
 }  // namespace webrtc