diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc index ff9c4d922..4b9607439 100644 --- a/webrtc/modules/pacing/paced_sender.cc +++ b/webrtc/modules/pacing/paced_sender.cc @@ -22,7 +22,6 @@ #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" #include "webrtc/system_wrappers/interface/field_trial.h" #include "webrtc/system_wrappers/interface/logging.h" -#include "webrtc/system_wrappers/interface/trace_event.h" namespace { // Time limit in milliseconds between packet bursts. diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc index 8dc728c18..25a4756e6 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -394,9 +394,8 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser, if (rtcpPacketType == RTCPUtility::kRtcpSrCode) { - TRACE_EVENT_INSTANT2("webrtc_rtp", "SR", - "remote_ssrc", remoteSSRC, - "ssrc", main_ssrc_); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "SR", + "remote_ssrc", remoteSSRC, "ssrc", main_ssrc_); if (_remoteSSRC == remoteSSRC) // have I received RTP packets from this party { @@ -425,9 +424,8 @@ RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser, } } else { - TRACE_EVENT_INSTANT2("webrtc_rtp", "RR", - "remote_ssrc", remoteSSRC, - "ssrc", main_ssrc_); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR", + "remote_ssrc", remoteSSRC, "ssrc", main_ssrc_); rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr; } @@ -555,7 +553,8 @@ void RTCPReceiver::HandleReportBlock( reportBlock->numAverageCalcs++; } - TRACE_COUNTER_ID1("webrtc_rtp", "RR_RTT", rb.SSRC, RTT); + TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR_RTT", rb.SSRC, + RTT); rtcpPacketInformation.AddReportInfo(*reportBlock); } @@ -986,7 +985,7 @@ void RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser, RTCPPacketInformation& rtcpPacketInformation) { const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); if (main_ssrc_ == rtcpPacket.PLI.MediaSSRC) { - TRACE_EVENT_INSTANT0("webrtc_rtp", "PLI"); + TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PLI"); ++packet_type_counter_.pli_packets; // Received a signal that we need to send a new key frame. diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index 95256ba8c..7bf38a6f9 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc @@ -1764,9 +1764,11 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, } else if (buildVal == -2) { return position; } - TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::PLI"); + TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCPSender::PLI"); ++packet_type_counter_.pli_packets; - TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_PLICount", _SSRC, + TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCP_PLICount", _SSRC, packet_type_counter_.pli_packets); } if(rtcpPacketTypeFlags & kRtcpFir) @@ -1777,9 +1779,11 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, } else if (buildVal == -2) { return position; } - TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::FIR"); + TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCPSender::FIR"); ++packet_type_counter_.fir_packets; - TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_FIRCount", _SSRC, + TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCP_FIRCount", _SSRC, packet_type_counter_.fir_packets); } if(rtcpPacketTypeFlags & kRtcpSli) @@ -1813,7 +1817,8 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, } else if (buildVal == -2) { return position; } - TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::REMB"); + TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCPSender::REMB"); } if(rtcpPacketTypeFlags & kRtcpBye) { @@ -1861,10 +1866,12 @@ int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state, } else if (buildVal == -2) { return position; } - TRACE_EVENT_INSTANT1("webrtc_rtp", "RTCPSender::NACK", - "nacks", TRACE_STR_COPY(nackString.c_str())); + TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCPSender::NACK", "nacks", + TRACE_STR_COPY(nackString.c_str())); ++packet_type_counter_.nack_packets; - TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_NACKCount", _SSRC, + TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTCP_NACKCount", _SSRC, packet_type_counter_.nack_packets); } if(rtcpPacketTypeFlags & kRtcpXrVoipMetric) diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc index 0235635ed..dc5241a79 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc @@ -187,9 +187,9 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, size_t payload_length, int64_t timestamp_ms, bool is_first_packet) { - TRACE_EVENT2("webrtc_rtp", "Audio::ParseRtp", - "seqnum", rtp_header->header.sequenceNumber, - "timestamp", rtp_header->header.timestamp); + TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::ParseRtp", + "seqnum", rtp_header->header.sequenceNumber, "timestamp", + rtp_header->header.timestamp); rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs; num_energy_ = rtp_header->type.Audio.numEnergy; if (rtp_header->type.Audio.numEnergy > 0 && diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc index 8c50b025a..90dfc9231 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -54,11 +54,8 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, size_t payload_length, int64_t timestamp_ms, bool is_first_packet) { - TRACE_EVENT2("webrtc_rtp", - "Video::ParseRtp", - "seqnum", - rtp_header->header.sequenceNumber, - "timestamp", + TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp", + "seqnum", rtp_header->header.sequenceNumber, "timestamp", rtp_header->header.timestamp); rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index ae814a75c..fe1af00a3 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -685,8 +685,9 @@ bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) { if (transport_) { bytes_sent = transport_->SendPacket(id_, packet, size); } - TRACE_EVENT_INSTANT2("webrtc_rtp", "RTPSender::SendPacketToNetwork", - "size", size, "sent", bytes_sent); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTPSender::SendPacketToNetwork", "size", size, "sent", + bytes_sent); // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer. if (bytes_sent <= 0) { LOG(LS_WARNING) << "Transport failed to send packet"; @@ -709,8 +710,9 @@ int RTPSender::SetSelectiveRetransmissions(uint8_t settings) { void RTPSender::OnReceivedNACK(const std::list& nack_sequence_numbers, int64_t avg_rtt) { - TRACE_EVENT2("webrtc_rtp", "RTPSender::OnReceivedNACK", - "num_seqnum", nack_sequence_numbers.size(), "avg_rtt", avg_rtt); + TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "RTPSender::OnReceivedNACK", "num_seqnum", + nack_sequence_numbers.size(), "avg_rtt", avg_rtt); const int64_t now = clock_->TimeInMilliseconds(); uint32_t bytes_re_sent = 0; uint32_t target_bitrate = GetTargetBitrate(); @@ -840,12 +842,13 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer, RTPHeader rtp_header; rtp_parser.Parse(rtp_header); if (!is_retransmit && rtp_header.markerBit) { - TRACE_EVENT_ASYNC_END0("webrtc_rtp", "PacedSend", capture_time_ms); + TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend", + capture_time_ms); } - TRACE_EVENT_INSTANT2("webrtc_rtp", "PrepareAndSendPacket", - "timestamp", rtp_header.timestamp, - "seqnum", rtp_header.sequenceNumber); + TRACE_EVENT_INSTANT2( + TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PrepareAndSendPacket", + "timestamp", rtp_header.timestamp, "seqnum", rtp_header.sequenceNumber); uint8_t data_buffer_rtx[IP_PACKET_SIZE]; if (send_over_rtx) { @@ -969,7 +972,8 @@ int32_t RTPSender::SendToNetwork( if (last_capture_time_ms_sent_ == 0 || corrected_time_ms > last_capture_time_ms_sent_) { last_capture_time_ms_sent_ = corrected_time_ms; - TRACE_EVENT_ASYNC_BEGIN1("webrtc_rtp", "PacedSend", corrected_time_ms, + TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "PacedSend", corrected_time_ms, "capture_time_ms", corrected_time_ms); } // We can't send the packet right now. diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc index 7ec8aa2fc..cdf545009 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -473,10 +473,10 @@ RTPSenderAudio::SendTelephoneEventPacket(bool ended, dtmfbuffer[13] = E|R|volume; RtpUtility::AssignUWord16ToBuffer(dtmfbuffer + 14, duration); - TRACE_EVENT_INSTANT2("webrtc_rtp", - "Audio::SendTelephoneEvent", - "timestamp", dtmfTimeStamp, - "seqnum", _rtpSender->SequenceNumber()); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "Audio::SendTelephoneEvent", "timestamp", + dtmfTimeStamp, "seqnum", + _rtpSender->SequenceNumber()); retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1, kAllowRetransmission, PacedSender::kHighPriority); diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index a164e5d5c..f82496a2d 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -108,12 +108,9 @@ int32_t RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer, RedPacket* red_packet = producer_fec_.BuildRedPacket( data_buffer, payload_length, rtp_header_length, _payloadTypeRED); - TRACE_EVENT_INSTANT2("webrtc_rtp", - "Video::PacketRed", - "timestamp", - capture_timestamp, - "seqnum", - _rtpSender.SequenceNumber()); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "Video::PacketRed", "timestamp", capture_timestamp, + "seqnum", _rtpSender.SequenceNumber()); // Sending the media packet with RED header. int packet_success = _rtpSender.SendToNetwork(red_packet->data(), @@ -148,12 +145,9 @@ int32_t RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer, if (_retransmissionSettings & kRetransmitFECPackets) { storage = kAllowRetransmission; } - TRACE_EVENT_INSTANT2("webrtc_rtp", - "Video::PacketFec", - "timestamp", - capture_timestamp, - "seqnum", - _rtpSender.SequenceNumber()); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "Video::PacketFec", "timestamp", capture_timestamp, + "seqnum", _rtpSender.SequenceNumber()); // Sending FEC packet with RED header. int packet_success = _rtpSender.SendToNetwork(red_packet->data(), @@ -175,12 +169,9 @@ int32_t RTPSenderVideo::SendVideoPacket(uint8_t* data_buffer, _fecOverheadRate.Update(fec_overhead_sent); return ret; } - TRACE_EVENT_INSTANT2("webrtc_rtp", - "Video::PacketNormal", - "timestamp", - capture_timestamp, - "seqnum", - _rtpSender.SequenceNumber()); + TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "Video::PacketNormal", "timestamp", capture_timestamp, + "seqnum", _rtpSender.SequenceNumber()); int ret = _rtpSender.SendToNetwork(data_buffer, payload_length, rtp_header_length, @@ -206,9 +197,8 @@ int32_t RTPSenderVideo::SendRTPIntraRequest() { RtpUtility::AssignUWord32ToBuffer(data + 4, _rtpSender.SSRC()); - TRACE_EVENT_INSTANT1("webrtc_rtp", - "Video::IntraRequest", - "seqnum", + TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), + "Video::IntraRequest", "seqnum", _rtpSender.SequenceNumber()); return _rtpSender.SendToNetwork( data, 0, length, -1, kDontStore, PacedSender::kNormalPriority); diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc index 7b4e21d6f..05b6c45dd 100644 --- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc +++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc @@ -950,7 +950,8 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) { for (uint16_t i = latest_received_sequence_number_ + 1; IsNewerSequenceNumber(sequence_number, i); ++i) { missing_sequence_numbers_.insert(missing_sequence_numbers_.end(), i); - TRACE_EVENT_INSTANT1("webrtc", "AddNack", "seqnum", i); + TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "AddNack", + "seqnum", i); } if (TooLargeNackList() && !HandleTooLargeNackList()) { LOG(LS_WARNING) << "Requesting key frame due to too large NACK list."; @@ -963,7 +964,8 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) { } } else { missing_sequence_numbers_.erase(sequence_number); - TRACE_EVENT_INSTANT1("webrtc", "RemoveNack", "seqnum", sequence_number); + TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RemoveNack", + "seqnum", sequence_number); } return true; } diff --git a/webrtc/modules/video_coding/main/source/video_coding_impl.cc b/webrtc/modules/video_coding/main/source/video_coding_impl.cc index b7c72da52..7289e2f0b 100644 --- a/webrtc/modules/video_coding/main/source/video_coding_impl.cc +++ b/webrtc/modules/video_coding/main/source/video_coding_impl.cc @@ -16,7 +16,6 @@ #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/modules/video_coding/main/source/video_coding_impl.h" #include "webrtc/system_wrappers/interface/clock.h" -#include "webrtc/system_wrappers/interface/trace_event.h" namespace webrtc { namespace vcm { diff --git a/webrtc/system_wrappers/interface/trace_event.h b/webrtc/system_wrappers/interface/trace_event.h index 23b16c794..f97fd12fc 100644 --- a/webrtc/system_wrappers/interface/trace_event.h +++ b/webrtc/system_wrappers/interface/trace_event.h @@ -147,6 +147,10 @@ #define TRACE_STR_COPY(str) \ webrtc::trace_event_internal::TraceStringWithCopy(str) +// This will mark the trace event as disabled by default. The user will need +// to explicitly enable the event. +#define TRACE_DISABLED_BY_DEFAULT(name) "disabled-by-default-" name + // By default, uint64 ID argument values are not mangled with the Process ID in // TRACE_EVENT_ASYNC macros. Use this macro to force Process ID mangling. #define TRACE_ID_MANGLE(id) \