Change rtp_event_log2text to ignore webrtc::MediaType from proto.
BUG=webrtc:7538 Review-Url: https://codereview.webrtc.org/2894833003 Cr-Commit-Position: refs/heads/master@{#18210}
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perkj
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c52bd61f65
@ -73,6 +73,32 @@ bool ParseSsrc(std::string str) {
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return str.empty() || (!ss.fail() && ss.eof());
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}
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// Struct used for storing SSRCs used in a Stream.
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struct Stream {
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Stream(uint32_t ssrc,
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webrtc::MediaType media_type,
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webrtc::PacketDirection direction)
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: ssrc(ssrc), media_type(media_type), direction(direction) {}
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uint32_t ssrc;
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webrtc::MediaType media_type;
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webrtc::PacketDirection direction;
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};
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// All configured streams found in the event log.
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std::vector<Stream> global_streams;
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// Returns the MediaType for registered SSRCs. Search from the end to use last
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// registered types first.
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webrtc::MediaType GetMediaType(uint32_t ssrc,
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webrtc::PacketDirection direction) {
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for (auto rit = global_streams.rbegin(); rit != global_streams.rend();
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++rit) {
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if (rit->ssrc == ssrc && rit->direction == direction)
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return rit->media_type;
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}
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return webrtc::MediaType::ANY;
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}
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bool ExcludePacket(webrtc::PacketDirection direction,
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webrtc::MediaType media_type,
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uint32_t packet_ssrc) {
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@ -118,11 +144,11 @@ const char* StreamInfo(webrtc::PacketDirection direction,
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void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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webrtc::rtcp::SenderReport sr;
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if (!sr.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -133,11 +159,11 @@ void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
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void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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webrtc::rtcp::ReceiverReport rr;
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if (!rr.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -147,11 +173,11 @@ void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
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void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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webrtc::rtcp::ExtendedReports xr;
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if (!xr.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -161,20 +187,20 @@ void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
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void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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std::cout << log_timestamp << "\t"
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<< "RTCP_SDES" << StreamInfo(direction, media_type) << std::endl;
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<< "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
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<< std::endl;
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RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
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}
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void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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webrtc::rtcp::Bye bye;
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if (!bye.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -184,13 +210,14 @@ void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
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void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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switch (rtcp_block.fmt()) {
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case webrtc::rtcp::Nack::kFeedbackMessageType: {
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webrtc::rtcp::Nack nack;
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if (!nack.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(nack.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -202,6 +229,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::Tmmbr tmmbr;
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if (!tmmbr.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(tmmbr.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -213,6 +242,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::Tmmbn tmmbn;
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if (!tmmbn.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(tmmbn.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -224,6 +255,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::RapidResyncRequest sr_req;
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if (!sr_req.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(sr_req.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -235,6 +268,8 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::TransportFeedback transport_feedback;
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if (!transport_feedback.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(transport_feedback.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type,
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transport_feedback.sender_ssrc()))
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return;
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@ -250,13 +285,13 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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uint64_t log_timestamp,
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webrtc::PacketDirection direction,
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webrtc::MediaType media_type) {
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webrtc::PacketDirection direction) {
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switch (rtcp_block.fmt()) {
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case webrtc::rtcp::Pli::kFeedbackMessageType: {
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webrtc::rtcp::Pli pli;
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if (!pli.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -268,6 +303,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::Fir fir;
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if (!fir.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -279,6 +315,8 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
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webrtc::rtcp::Remb remb;
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if (!remb.Parse(rtcp_block))
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return;
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webrtc::MediaType media_type =
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GetMediaType(remb.sender_ssrc(), direction);
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if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
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return;
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std::cout << log_timestamp << "\t"
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@ -324,45 +362,75 @@ int main(int argc, char* argv[]) {
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}
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for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
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if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
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if (parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
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webrtc::VideoReceiveStream::Config config(nullptr);
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parsed_stream.GetVideoReceiveConfig(i, &config);
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
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<< "\tssrc=" << config.rtp.remote_ssrc
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<< "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
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global_streams.emplace_back(config.rtp.remote_ssrc,
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webrtc::MediaType::VIDEO,
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webrtc::kIncomingPacket);
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global_streams.emplace_back(config.rtp.local_ssrc,
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webrtc::MediaType::VIDEO,
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webrtc::kOutgoingPacket);
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if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
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<< "\tssrc=" << config.rtp.remote_ssrc
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<< "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
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}
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}
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if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
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if (parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
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webrtc::VideoSendStream::Config config(nullptr);
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parsed_stream.GetVideoSendConfig(i, &config);
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
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std::cout << "\tssrcs=";
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for (const auto& ssrc : config.rtp.ssrcs)
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std::cout << ssrc << ',';
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std::cout << "\trtx_ssrcs=";
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for (const auto& ssrc : config.rtp.rtx.ssrcs)
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std::cout << ssrc << ',';
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std::cout << std::endl;
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for (uint32_t ssrc : config.rtp.ssrcs) {
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global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
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webrtc::kOutgoingPacket);
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}
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for (uint32_t ssrc : config.rtp.rtx.ssrcs) {
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global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
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webrtc::kOutgoingPacket);
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}
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if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
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std::cout << "\tssrcs=";
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for (const auto& ssrc : config.rtp.ssrcs)
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std::cout << ssrc << ',';
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std::cout << "\trtx_ssrcs=";
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for (const auto& ssrc : config.rtp.rtx.ssrcs)
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std::cout << ssrc << ',';
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std::cout << std::endl;
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}
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}
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if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
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if (parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
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webrtc::AudioReceiveStream::Config config;
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parsed_stream.GetAudioReceiveConfig(i, &config);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
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<< "\tssrc=" << config.rtp.remote_ssrc
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<< "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
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global_streams.emplace_back(config.rtp.remote_ssrc,
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webrtc::MediaType::AUDIO,
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webrtc::kIncomingPacket);
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global_streams.emplace_back(config.rtp.local_ssrc,
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webrtc::MediaType::AUDIO,
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webrtc::kOutgoingPacket);
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if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
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<< "\tssrc=" << config.rtp.remote_ssrc
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<< "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
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}
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}
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if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
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if (parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
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webrtc::AudioSendStream::Config config(nullptr);
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parsed_stream.GetAudioSendConfig(i, &config);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
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<< "\tssrc=" << config.rtp.ssrc << std::endl;
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global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
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webrtc::kOutgoingPacket);
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if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
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<< "\tssrc=" << config.rtp.ssrc << std::endl;
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}
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}
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if (!FLAGS_nortp &&
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parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
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@ -378,6 +446,7 @@ int main(int argc, char* argv[]) {
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webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
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webrtc::RTPHeader parsed_header;
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rtp_parser.Parse(&parsed_header);
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media_type = GetMediaType(parsed_header.ssrc, direction);
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if (ExcludePacket(direction, media_type, parsed_header.ssrc))
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continue;
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@ -409,26 +478,25 @@ int main(int argc, char* argv[]) {
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uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
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switch (rtcp_block.type()) {
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case webrtc::rtcp::SenderReport::kPacketType:
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PrintSenderReport(rtcp_block, log_timestamp, direction, media_type);
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PrintSenderReport(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::ReceiverReport::kPacketType:
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PrintReceiverReport(rtcp_block, log_timestamp, direction,
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media_type);
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PrintReceiverReport(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Sdes::kPacketType:
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PrintSdes(rtcp_block, log_timestamp, direction, media_type);
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PrintSdes(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::ExtendedReports::kPacketType:
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PrintXr(rtcp_block, log_timestamp, direction, media_type);
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PrintXr(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Bye::kPacketType:
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PrintBye(rtcp_block, log_timestamp, direction, media_type);
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PrintBye(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Rtpfb::kPacketType:
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PrintRtpFeedback(rtcp_block, log_timestamp, direction, media_type);
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PrintRtpFeedback(rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Psfb::kPacketType:
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PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
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PrintPsFeedback(rtcp_block, log_timestamp, direction);
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break;
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default:
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break;
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