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Replace rtcp parser in rtc event log handlers.

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RTCPUtility is going away.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2395383002
Cr-Commit-Position: refs/heads/master@{#14574}
This commit is contained in:
danilchap 2016-10-07 07:39:54 -07:00 committed by Commit bot
parent 05f3ec1356
commit bf369fe3dd
2 changed files with 39 additions and 46 deletions
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47
webrtc/logging/rtc_event_log/rtc_event_log.cc
View File

@ -22,7 +22,16 @@
#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"
@ -323,42 +332,34 @@ void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
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:
// 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.

38
webrtc/tools/event_log_visualizer/analyzer.cc
View File

@ -29,7 +29,7 @@
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
@ -392,35 +392,27 @@ EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
&total_length);
RtpUtility::RtpHeaderParser rtp_parser(packet, total_length);
RTPHeader parsed_header;
RTC_CHECK(rtp_parser.ParseRtcp(&parsed_header));
uint32_t ssrc = parsed_header.ssrc;
RTCPUtility::RTCPParserV2 rtcp_parser(packet, total_length, true);
RTC_CHECK(rtcp_parser.IsValid());
RTCPUtility::RTCPPacketTypes packet_type = rtcp_parser.Begin();
while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
switch (packet_type) {
case RTCPUtility::RTCPPacketTypes::kTransportFeedback: {
// Currently feedback is logged twice, both for audio and video.
// Only act on one of them.
if (media_type == MediaType::VIDEO) {
std::unique_ptr<rtcp::RtcpPacket> rtcp_packet(
rtcp_parser.ReleaseRtcpPacket());
// Currently feedback is logged twice, both for audio and video.
// Only act on one of them.
if (media_type == MediaType::VIDEO) {
rtcp::CommonHeader header;
const uint8_t* packet_end = packet + total_length;
for (const uint8_t* block = packet; block < packet_end;
block = header.NextPacket()) {
RTC_CHECK(header.Parse(block, packet_end - block));
if (header.type() == rtcp::TransportFeedback::kPacketType &&
header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
new rtcp::TransportFeedback());
if (rtcp_packet->Parse(header)) {
uint32_t ssrc = rtcp_packet->sender_ssrc();
StreamId stream(ssrc, direction);
uint64_t timestamp = parsed_log_.GetTimestamp(i);
rtcp_packets_[stream].push_back(LoggedRtcpPacket(
timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
}
break;
}
default:
break;
}
rtcp_parser.Iterate();
packet_type = rtcp_parser.PacketType();
}
break;
}