Change rtp_event_log2text to ignore webrtc::MediaType from proto.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2894833003
Cr-Original-Commit-Position: refs/heads/master@{#18210}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: c52bd61f65d04b364e90a672691963d089ca5141
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index b2d537e..1fa694d 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -73,6 +73,32 @@
return str.empty() || (!ss.fail() && ss.eof());
}
+// Struct used for storing SSRCs used in a Stream.
+struct Stream {
+ Stream(uint32_t ssrc,
+ webrtc::MediaType media_type,
+ webrtc::PacketDirection direction)
+ : ssrc(ssrc), media_type(media_type), direction(direction) {}
+ uint32_t ssrc;
+ webrtc::MediaType media_type;
+ webrtc::PacketDirection direction;
+};
+
+// All configured streams found in the event log.
+std::vector<Stream> global_streams;
+
+// Returns the MediaType for registered SSRCs. Search from the end to use last
+// registered types first.
+webrtc::MediaType GetMediaType(uint32_t ssrc,
+ webrtc::PacketDirection direction) {
+ for (auto rit = global_streams.rbegin(); rit != global_streams.rend();
+ ++rit) {
+ if (rit->ssrc == ssrc && rit->direction == direction)
+ return rit->media_type;
+ }
+ return webrtc::MediaType::ANY;
+}
+
bool ExcludePacket(webrtc::PacketDirection direction,
webrtc::MediaType media_type,
uint32_t packet_ssrc) {
@@ -118,11 +144,11 @@
void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
webrtc::rtcp::SenderReport sr;
if (!sr.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -133,11 +159,11 @@
void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
webrtc::rtcp::ReceiverReport rr;
if (!rr.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -147,11 +173,11 @@
void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
webrtc::rtcp::ExtendedReports xr;
if (!xr.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -161,20 +187,20 @@
void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
std::cout << log_timestamp << "\t"
- << "RTCP_SDES" << StreamInfo(direction, media_type) << std::endl;
+ << "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
+ << std::endl;
RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
}
void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
webrtc::rtcp::Bye bye;
if (!bye.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -184,13 +210,14 @@
void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
switch (rtcp_block.fmt()) {
case webrtc::rtcp::Nack::kFeedbackMessageType: {
webrtc::rtcp::Nack nack;
if (!nack.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(nack.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -202,6 +229,8 @@
webrtc::rtcp::Tmmbr tmmbr;
if (!tmmbr.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(tmmbr.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -213,6 +242,8 @@
webrtc::rtcp::Tmmbn tmmbn;
if (!tmmbn.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(tmmbn.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -224,6 +255,8 @@
webrtc::rtcp::RapidResyncRequest sr_req;
if (!sr_req.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(sr_req.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -235,6 +268,8 @@
webrtc::rtcp::TransportFeedback transport_feedback;
if (!transport_feedback.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(transport_feedback.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type,
transport_feedback.sender_ssrc()))
return;
@@ -250,13 +285,13 @@
void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
uint64_t log_timestamp,
- webrtc::PacketDirection direction,
- webrtc::MediaType media_type) {
+ webrtc::PacketDirection direction) {
switch (rtcp_block.fmt()) {
case webrtc::rtcp::Pli::kFeedbackMessageType: {
webrtc::rtcp::Pli pli;
if (!pli.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -268,6 +303,7 @@
webrtc::rtcp::Fir fir;
if (!fir.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -279,6 +315,8 @@
webrtc::rtcp::Remb remb;
if (!remb.Parse(rtcp_block))
return;
+ webrtc::MediaType media_type =
+ GetMediaType(remb.sender_ssrc(), direction);
if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
return;
std::cout << log_timestamp << "\t"
@@ -324,45 +362,75 @@
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
- if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
+ if (parsed_stream.GetEventType(i) ==
+ webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
webrtc::VideoReceiveStream::Config config(nullptr);
parsed_stream.GetVideoReceiveConfig(i, &config);
- std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
- << "\tssrc=" << config.rtp.remote_ssrc
- << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+
+ global_streams.emplace_back(config.rtp.remote_ssrc,
+ webrtc::MediaType::VIDEO,
+ webrtc::kIncomingPacket);
+ global_streams.emplace_back(config.rtp.local_ssrc,
+ webrtc::MediaType::VIDEO,
+ webrtc::kOutgoingPacket);
+
+ if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
+ << "\tssrc=" << config.rtp.remote_ssrc
+ << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+ }
}
- if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
+ if (parsed_stream.GetEventType(i) ==
+ webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
webrtc::VideoSendStream::Config config(nullptr);
parsed_stream.GetVideoSendConfig(i, &config);
- std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
- std::cout << "\tssrcs=";
- for (const auto& ssrc : config.rtp.ssrcs)
- std::cout << ssrc << ',';
- std::cout << "\trtx_ssrcs=";
- for (const auto& ssrc : config.rtp.rtx.ssrcs)
- std::cout << ssrc << ',';
- std::cout << std::endl;
+
+ for (uint32_t ssrc : config.rtp.ssrcs) {
+ global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
+ webrtc::kOutgoingPacket);
+ }
+ for (uint32_t ssrc : config.rtp.rtx.ssrcs) {
+ global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
+ webrtc::kOutgoingPacket);
+ }
+
+ if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
+ std::cout << "\tssrcs=";
+ for (const auto& ssrc : config.rtp.ssrcs)
+ std::cout << ssrc << ',';
+ std::cout << "\trtx_ssrcs=";
+ for (const auto& ssrc : config.rtp.rtx.ssrcs)
+ std::cout << ssrc << ',';
+ std::cout << std::endl;
+ }
}
- if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
+ if (parsed_stream.GetEventType(i) ==
+ webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
webrtc::AudioReceiveStream::Config config;
parsed_stream.GetAudioReceiveConfig(i, &config);
- std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
- << "\tssrc=" << config.rtp.remote_ssrc
- << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+ global_streams.emplace_back(config.rtp.remote_ssrc,
+ webrtc::MediaType::AUDIO,
+ webrtc::kIncomingPacket);
+ global_streams.emplace_back(config.rtp.local_ssrc,
+ webrtc::MediaType::AUDIO,
+ webrtc::kOutgoingPacket);
+ if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
+ << "\tssrc=" << config.rtp.remote_ssrc
+ << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+ }
}
- if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
+ if (parsed_stream.GetEventType(i) ==
+ webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
webrtc::AudioSendStream::Config config(nullptr);
parsed_stream.GetAudioSendConfig(i, &config);
- std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
- << "\tssrc=" << config.rtp.ssrc << std::endl;
+ global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
+ webrtc::kOutgoingPacket);
+ if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
+ << "\tssrc=" << config.rtp.ssrc << std::endl;
+ }
}
if (!FLAGS_nortp &&
parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
@@ -378,6 +446,7 @@
webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
webrtc::RTPHeader parsed_header;
rtp_parser.Parse(&parsed_header);
+ media_type = GetMediaType(parsed_header.ssrc, direction);
if (ExcludePacket(direction, media_type, parsed_header.ssrc))
continue;
@@ -409,26 +478,25 @@
uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
switch (rtcp_block.type()) {
case webrtc::rtcp::SenderReport::kPacketType:
- PrintSenderReport(rtcp_block, log_timestamp, direction, media_type);
+ PrintSenderReport(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::ReceiverReport::kPacketType:
- PrintReceiverReport(rtcp_block, log_timestamp, direction,
- media_type);
+ PrintReceiverReport(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Sdes::kPacketType:
- PrintSdes(rtcp_block, log_timestamp, direction, media_type);
+ PrintSdes(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::ExtendedReports::kPacketType:
- PrintXr(rtcp_block, log_timestamp, direction, media_type);
+ PrintXr(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Bye::kPacketType:
- PrintBye(rtcp_block, log_timestamp, direction, media_type);
+ PrintBye(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Rtpfb::kPacketType:
- PrintRtpFeedback(rtcp_block, log_timestamp, direction, media_type);
+ PrintRtpFeedback(rtcp_block, log_timestamp, direction);
break;
case webrtc::rtcp::Psfb::kPacketType:
- PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
+ PrintPsFeedback(rtcp_block, log_timestamp, direction);
break;
default:
break;
