Make rtc_event_log2text handle all events [1/2]
rtc_event_log2text doesn't currently handle all possible RtcEvent-s.
1. This CL - to make sure events are not forgotten in the future, change the succession of if-statements to a switch, so that the compiler would complain if events are ever added, but are not handled here.
2. Next CL - add handling of currently-unhandled events.
BUG=webrtc:8111
NOPRESUBMIT=True
Change-Id: Ia4459b4e760eb0208823fdab69996de0e8420703
Reviewed-on: https://webrtc-review.googlesource.com/1242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19861}diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index c7cad48..4d96711 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/logging/rtc_event_log/rtc_event_log2text.cc
@@ -386,190 +386,260 @@
}
for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
- if (FLAG_config && FLAG_video && FLAG_incoming &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
- webrtc::rtclog::StreamConfig config =
- parsed_stream.GetVideoReceiveConfig(i);
- std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
- << "\tssrc=" << config.remote_ssrc
- << "\tfeedback_ssrc=" << config.local_ssrc;
- std::cout << "\textensions={";
- for (const auto& extension : config.rtp_extensions) {
- std::cout << extension.ToString() << ",";
- }
- std::cout << "}";
- std::cout << "\tcodecs={";
- for (const auto& codec : config.codecs) {
- std::cout << "{name: " << codec.payload_name
- << ", payload_type: " << codec.payload_type
- << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
- }
- std::cout << "}" << std::endl;
- }
- if (FLAG_config && FLAG_video && FLAG_outgoing &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
- std::vector<webrtc::rtclog::StreamConfig> configs =
- parsed_stream.GetVideoSendConfig(i);
- for (const auto& config : configs) {
- std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
- std::cout << "\tssrcs=" << config.local_ssrc;
- std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
- std::cout << "\textensions={";
- for (const auto& extension : config.rtp_extensions) {
- std::cout << extension.ToString() << ",";
- }
- std::cout << "}";
- std::cout << "\tcodecs={";
- for (const auto& codec : config.codecs) {
- std::cout << "{name: " << codec.payload_name
- << ", payload_type: " << codec.payload_type
- << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
- }
- std::cout << "}" << std::endl;
- }
- }
- if (FLAG_config && FLAG_audio && FLAG_incoming &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
- webrtc::rtclog::StreamConfig config =
- parsed_stream.GetAudioReceiveConfig(i);
- std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
- << "\tssrc=" << config.remote_ssrc
- << "\tfeedback_ssrc=" << config.local_ssrc;
- std::cout << "\textensions={";
- for (const auto& extension : config.rtp_extensions) {
- std::cout << extension.ToString() << ",";
- }
- std::cout << "}";
- std::cout << "\tcodecs={";
- for (const auto& codec : config.codecs) {
- std::cout << "{name: " << codec.payload_name
- << ", payload_type: " << codec.payload_type
- << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
- }
- std::cout << "}" << std::endl;
- }
- if (FLAG_config && FLAG_audio && FLAG_outgoing &&
- parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
- webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
- std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
- << "\tssrc=" << config.local_ssrc;
- std::cout << "\textensions={";
- for (const auto& extension : config.rtp_extensions) {
- std::cout << extension.ToString() << ",";
- }
- std::cout << "}";
- std::cout << "\tcodecs={";
- for (const auto& codec : config.codecs) {
- std::cout << "{name: " << codec.payload_name
- << ", payload_type: " << codec.payload_type
- << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
- }
- std::cout << "}" << std::endl;
- }
- if (FLAG_rtp &&
- parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
- size_t header_length;
- size_t total_length;
- uint8_t header[IP_PACKET_SIZE];
- webrtc::PacketDirection direction;
- webrtc::RtpHeaderExtensionMap* extension_map = parsed_stream.GetRtpHeader(
- i, &direction, header, &header_length, &total_length);
-
- if (extension_map == nullptr)
- extension_map = &default_map;
-
- // Parse header to get SSRC and RTP time.
- webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
- webrtc::RTPHeader parsed_header;
- rtp_parser.Parse(&parsed_header, extension_map);
- MediaType media_type =
- parsed_stream.GetMediaType(parsed_header.ssrc, direction);
-
- if (ExcludePacket(direction, media_type, parsed_header.ssrc))
+ switch (parsed_stream.GetEventType(i)) {
+ case webrtc::ParsedRtcEventLog::UNKNOWN_EVENT: {
+ // TODO(eladalon): Implement in new CL.
continue;
+ }
- std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
- << StreamInfo(direction, media_type)
- << "\tssrc=" << parsed_header.ssrc
- << "\ttimestamp=" << parsed_header.timestamp;
- if (parsed_header.extension.hasAbsoluteSendTime) {
- std::cout << "\tAbsSendTime="
- << parsed_header.extension.absoluteSendTime;
+ case webrtc::ParsedRtcEventLog::LOG_START: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
}
- if (parsed_header.extension.hasVideoContentType) {
- std::cout << "\tContentType="
- << static_cast<int>(parsed_header.extension.videoContentType);
- }
- if (parsed_header.extension.hasVideoRotation) {
- std::cout << "\tRotation="
- << static_cast<int>(parsed_header.extension.videoRotation);
- }
- if (parsed_header.extension.hasTransportSequenceNumber) {
- std::cout << "\tTransportSeq="
- << parsed_header.extension.transportSequenceNumber;
- }
- if (parsed_header.extension.hasTransmissionTimeOffset) {
- std::cout << "\tTransmTimeOffset="
- << parsed_header.extension.transmissionTimeOffset;
- }
- if (parsed_header.extension.hasAudioLevel) {
- std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
- }
- std::cout << std::endl;
- }
- if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
- webrtc::ParsedRtcEventLog::RTCP_EVENT) {
- size_t length;
- uint8_t packet[IP_PACKET_SIZE];
- webrtc::PacketDirection direction;
- parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
- webrtc::rtcp::CommonHeader rtcp_block;
- const uint8_t* packet_end = packet + length;
- for (const uint8_t* next_block = packet; next_block != packet_end;
- next_block = rtcp_block.NextPacket()) {
- ptrdiff_t remaining_blocks_size = packet_end - next_block;
- RTC_DCHECK_GT(remaining_blocks_size, 0);
- if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
- break;
+ case webrtc::ParsedRtcEventLog::LOG_END: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::RTP_EVENT: {
+ if (FLAG_rtp) {
+ size_t header_length;
+ size_t total_length;
+ uint8_t header[IP_PACKET_SIZE];
+ webrtc::PacketDirection direction;
+ webrtc::RtpHeaderExtensionMap* extension_map =
+ parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
+ &total_length);
+
+ if (extension_map == nullptr)
+ extension_map = &default_map;
+
+ // Parse header to get SSRC and RTP time.
+ webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
+ webrtc::RTPHeader parsed_header;
+ rtp_parser.Parse(&parsed_header, extension_map);
+ MediaType media_type =
+ parsed_stream.GetMediaType(parsed_header.ssrc, direction);
+
+ if (ExcludePacket(direction, media_type, parsed_header.ssrc))
+ continue;
+
+ std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
+ << StreamInfo(direction, media_type)
+ << "\tssrc=" << parsed_header.ssrc
+ << "\ttimestamp=" << parsed_header.timestamp;
+ if (parsed_header.extension.hasAbsoluteSendTime) {
+ std::cout << "\tAbsSendTime="
+ << parsed_header.extension.absoluteSendTime;
+ }
+ if (parsed_header.extension.hasVideoContentType) {
+ std::cout << "\tContentType="
+ << static_cast<int>(
+ parsed_header.extension.videoContentType);
+ }
+ if (parsed_header.extension.hasVideoRotation) {
+ std::cout << "\tRotation="
+ << static_cast<int>(
+ parsed_header.extension.videoRotation);
+ }
+ if (parsed_header.extension.hasTransportSequenceNumber) {
+ std::cout << "\tTransportSeq="
+ << parsed_header.extension.transportSequenceNumber;
+ }
+ if (parsed_header.extension.hasTransmissionTimeOffset) {
+ std::cout << "\tTransmTimeOffset="
+ << parsed_header.extension.transmissionTimeOffset;
+ }
+ if (parsed_header.extension.hasAudioLevel) {
+ std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
+ }
+ std::cout << std::endl;
}
+ continue;
+ }
- uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
- switch (rtcp_block.type()) {
- case webrtc::rtcp::SenderReport::kPacketType:
- PrintSenderReport(parsed_stream, rtcp_block, log_timestamp,
- direction);
- break;
- case webrtc::rtcp::ReceiverReport::kPacketType:
- PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
+ case webrtc::ParsedRtcEventLog::RTCP_EVENT: {
+ if (FLAG_rtcp) {
+ size_t length;
+ uint8_t packet[IP_PACKET_SIZE];
+ webrtc::PacketDirection direction;
+ parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
+
+ webrtc::rtcp::CommonHeader rtcp_block;
+ const uint8_t* packet_end = packet + length;
+ for (const uint8_t* next_block = packet; next_block != packet_end;
+ next_block = rtcp_block.NextPacket()) {
+ ptrdiff_t remaining_blocks_size = packet_end - next_block;
+ RTC_DCHECK_GT(remaining_blocks_size, 0);
+ if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
+ break;
+ }
+
+ uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
+ switch (rtcp_block.type()) {
+ case webrtc::rtcp::SenderReport::kPacketType:
+ PrintSenderReport(parsed_stream, rtcp_block, log_timestamp,
+ direction);
+ break;
+ case webrtc::rtcp::ReceiverReport::kPacketType:
+ PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
+ direction);
+ break;
+ case webrtc::rtcp::Sdes::kPacketType:
+ PrintSdes(rtcp_block, log_timestamp, direction);
+ break;
+ case webrtc::rtcp::ExtendedReports::kPacketType:
+ PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
+ break;
+ case webrtc::rtcp::Bye::kPacketType:
+ PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
+ break;
+ case webrtc::rtcp::Rtpfb::kPacketType:
+ PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
+ direction);
+ break;
+ case webrtc::rtcp::Psfb::kPacketType:
+ PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
direction);
- break;
- case webrtc::rtcp::Sdes::kPacketType:
- PrintSdes(rtcp_block, log_timestamp, direction);
- break;
- case webrtc::rtcp::ExtendedReports::kPacketType:
- PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
- break;
- case webrtc::rtcp::Bye::kPacketType:
- PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
- break;
- case webrtc::rtcp::Rtpfb::kPacketType:
- PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
- direction);
- break;
- case webrtc::rtcp::Psfb::kPacketType:
- PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
- direction);
- break;
- default:
- break;
+ break;
+ default:
+ break;
+ }
+ }
}
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
+ if (FLAG_config && FLAG_video && FLAG_incoming) {
+ webrtc::rtclog::StreamConfig config =
+ parsed_stream.GetVideoReceiveConfig(i);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
+ << "\tssrc=" << config.remote_ssrc
+ << "\tfeedback_ssrc=" << config.local_ssrc;
+ std::cout << "\textensions={";
+ for (const auto& extension : config.rtp_extensions) {
+ std::cout << extension.ToString() << ",";
+ }
+ std::cout << "}";
+ std::cout << "\tcodecs={";
+ for (const auto& codec : config.codecs) {
+ std::cout << "{name: " << codec.payload_name
+ << ", payload_type: " << codec.payload_type
+ << ", rtx_payload_type: " << codec.rtx_payload_type
+ << "}";
+ }
+ std::cout << "}" << std::endl;
+ }
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
+ if (FLAG_config && FLAG_video && FLAG_outgoing) {
+ std::vector<webrtc::rtclog::StreamConfig> configs =
+ parsed_stream.GetVideoSendConfig(i);
+ for (const auto& config : configs) {
+ std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
+ std::cout << "\tssrcs=" << config.local_ssrc;
+ std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
+ std::cout << "\textensions={";
+ for (const auto& extension : config.rtp_extensions) {
+ std::cout << extension.ToString() << ",";
+ }
+ std::cout << "}";
+ std::cout << "\tcodecs={";
+ for (const auto& codec : config.codecs) {
+ std::cout << "{name: " << codec.payload_name
+ << ", payload_type: " << codec.payload_type
+ << ", rtx_payload_type: " << codec.rtx_payload_type
+ << "}";
+ }
+ std::cout << "}" << std::endl;
+ }
+ }
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
+ if (FLAG_config && FLAG_audio && FLAG_incoming) {
+ webrtc::rtclog::StreamConfig config =
+ parsed_stream.GetAudioReceiveConfig(i);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
+ << "\tssrc=" << config.remote_ssrc
+ << "\tfeedback_ssrc=" << config.local_ssrc;
+ std::cout << "\textensions={";
+ for (const auto& extension : config.rtp_extensions) {
+ std::cout << extension.ToString() << ",";
+ }
+ std::cout << "}";
+ std::cout << "\tcodecs={";
+ for (const auto& codec : config.codecs) {
+ std::cout << "{name: " << codec.payload_name
+ << ", payload_type: " << codec.payload_type
+ << ", rtx_payload_type: " << codec.rtx_payload_type
+ << "}";
+ }
+ std::cout << "}" << std::endl;
+ }
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
+ if (FLAG_config && FLAG_audio && FLAG_outgoing) {
+ webrtc::rtclog::StreamConfig config =
+ parsed_stream.GetAudioSendConfig(i);
+ std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
+ << "\tssrc=" << config.local_ssrc;
+ std::cout << "\textensions={";
+ for (const auto& extension : config.rtp_extensions) {
+ std::cout << extension.ToString() << ",";
+ }
+ std::cout << "}";
+ std::cout << "\tcodecs={";
+ for (const auto& codec : config.codecs) {
+ std::cout << "{name: " << codec.payload_name
+ << ", payload_type: " << codec.payload_type
+ << ", rtx_payload_type: " << codec.rtx_payload_type
+ << "}";
+ }
+ std::cout << "}" << std::endl;
+ }
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
+ }
+
+ case webrtc::ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
+ // TODO(eladalon): Implement in new CL.
+ continue;
}
}
+
+ RTC_NOTREACHED();
}
return 0;
}
