Replace rtcp parser in rtc event log handlers.

webrtc/logging/rtc_event_log/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/logging/rtc_event_log/rtc_event_log.h"
 
 #include <limits>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.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/modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
@@ skipping 280 matching lines @@
 void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
                                     MediaType media_type,
                                     const uint8_t* packet,
                                     size_t length) {
   std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
   rtcp_event->set_timestamp_us(clock_->TimeInMicroseconds());
   rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
   rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
   rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
 
-  RTCPUtility::RtcpCommonHeader header;
+  rtcp::CommonHeader 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)) {
+    if (!header.Parse(block_begin, packet_end - block_begin)) {
       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:
+    const uint8_t* next_block = header.NextPacket();
+    uint32_t block_size = next_block - block_begin;
+    switch (header.type()) {
+      case rtcp::SenderReport::kPacketType:
+      case rtcp::ReceiverReport::kPacketType:
+      case rtcp::Bye::kPacketType:
+      case rtcp::ExtendedJitterReport::kPacketType:
+      case rtcp::Rtpfb::kPacketType:
+      case rtcp::Psfb::kPacketType:
+      case rtcp::ExtendedReports::kPacketType:

[terelius, 2016/10/07 12:39:25]

There is no need for annotating the fall-through?

[danilchap, 2016/10/07 12:45:18]

shouldn't be needed when the block is empty.
FALLTHROUGH is required when you go to next case in addition to doing something
else.

         // 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();
+      case rtcp::Sdes::kPacketType:
+      case rtcp::App::kPacketType:
       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);
   StoreEvent(&rtcp_event);
@@ skipping 62 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