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}
This commit is contained in:
Elad Alon
Commit Bot
parent
48d96c0bcc
commit
34f303cf58
@ -386,190 +386,260 @@ 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 (FLAG_config && FLAG_video && FLAG_incoming &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
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webrtc::rtclog::StreamConfig config =
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parsed_stream.GetVideoReceiveConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
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<< "\tssrc=" << config.remote_ssrc
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<< "\tfeedback_ssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
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}
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std::cout << "}" << std::endl;
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}
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if (FLAG_config && FLAG_video && FLAG_outgoing &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
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std::vector<webrtc::rtclog::StreamConfig> configs =
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parsed_stream.GetVideoSendConfig(i);
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for (const auto& config : configs) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
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std::cout << "\tssrcs=" << config.local_ssrc;
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std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
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}
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std::cout << "}" << std::endl;
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}
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}
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if (FLAG_config && FLAG_audio && FLAG_incoming &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
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webrtc::rtclog::StreamConfig config =
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parsed_stream.GetAudioReceiveConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
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<< "\tssrc=" << config.remote_ssrc
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<< "\tfeedback_ssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
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}
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std::cout << "}" << std::endl;
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}
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if (FLAG_config && FLAG_audio && FLAG_outgoing &&
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parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
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webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
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<< "\tssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type << "}";
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}
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std::cout << "}" << std::endl;
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}
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if (FLAG_rtp &&
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parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
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size_t header_length;
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size_t total_length;
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uint8_t header[IP_PACKET_SIZE];
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webrtc::PacketDirection direction;
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webrtc::RtpHeaderExtensionMap* extension_map = parsed_stream.GetRtpHeader(
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i, &direction, header, &header_length, &total_length);
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if (extension_map == nullptr)
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extension_map = &default_map;
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// Parse header to get SSRC and RTP time.
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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, extension_map);
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MediaType media_type =
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parsed_stream.GetMediaType(parsed_header.ssrc, direction);
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if (ExcludePacket(direction, media_type, parsed_header.ssrc))
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switch (parsed_stream.GetEventType(i)) {
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case webrtc::ParsedRtcEventLog::UNKNOWN_EVENT: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
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<< StreamInfo(direction, media_type)
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<< "\tssrc=" << parsed_header.ssrc
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<< "\ttimestamp=" << parsed_header.timestamp;
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if (parsed_header.extension.hasAbsoluteSendTime) {
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std::cout << "\tAbsSendTime="
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<< parsed_header.extension.absoluteSendTime;
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case webrtc::ParsedRtcEventLog::LOG_START: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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if (parsed_header.extension.hasVideoContentType) {
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std::cout << "\tContentType="
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<< static_cast<int>(parsed_header.extension.videoContentType);
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}
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if (parsed_header.extension.hasVideoRotation) {
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std::cout << "\tRotation="
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<< static_cast<int>(parsed_header.extension.videoRotation);
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}
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if (parsed_header.extension.hasTransportSequenceNumber) {
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std::cout << "\tTransportSeq="
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<< parsed_header.extension.transportSequenceNumber;
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}
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if (parsed_header.extension.hasTransmissionTimeOffset) {
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std::cout << "\tTransmTimeOffset="
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<< parsed_header.extension.transmissionTimeOffset;
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}
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if (parsed_header.extension.hasAudioLevel) {
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std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
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}
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std::cout << std::endl;
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}
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if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
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webrtc::ParsedRtcEventLog::RTCP_EVENT) {
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size_t length;
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uint8_t packet[IP_PACKET_SIZE];
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webrtc::PacketDirection direction;
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parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
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webrtc::rtcp::CommonHeader rtcp_block;
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const uint8_t* packet_end = packet + length;
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for (const uint8_t* next_block = packet; next_block != packet_end;
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next_block = rtcp_block.NextPacket()) {
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ptrdiff_t remaining_blocks_size = packet_end - next_block;
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RTC_DCHECK_GT(remaining_blocks_size, 0);
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if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
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break;
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case webrtc::ParsedRtcEventLog::LOG_END: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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case webrtc::ParsedRtcEventLog::RTP_EVENT: {
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if (FLAG_rtp) {
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size_t header_length;
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size_t total_length;
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uint8_t header[IP_PACKET_SIZE];
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webrtc::PacketDirection direction;
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webrtc::RtpHeaderExtensionMap* extension_map =
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parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
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&total_length);
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if (extension_map == nullptr)
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extension_map = &default_map;
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// Parse header to get SSRC and RTP time.
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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, extension_map);
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MediaType media_type =
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parsed_stream.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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std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
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<< StreamInfo(direction, media_type)
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<< "\tssrc=" << parsed_header.ssrc
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<< "\ttimestamp=" << parsed_header.timestamp;
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if (parsed_header.extension.hasAbsoluteSendTime) {
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std::cout << "\tAbsSendTime="
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<< parsed_header.extension.absoluteSendTime;
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}
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if (parsed_header.extension.hasVideoContentType) {
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std::cout << "\tContentType="
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<< static_cast<int>(
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parsed_header.extension.videoContentType);
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}
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if (parsed_header.extension.hasVideoRotation) {
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std::cout << "\tRotation="
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<< static_cast<int>(
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parsed_header.extension.videoRotation);
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}
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if (parsed_header.extension.hasTransportSequenceNumber) {
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std::cout << "\tTransportSeq="
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<< parsed_header.extension.transportSequenceNumber;
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}
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if (parsed_header.extension.hasTransmissionTimeOffset) {
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std::cout << "\tTransmTimeOffset="
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<< parsed_header.extension.transmissionTimeOffset;
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}
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if (parsed_header.extension.hasAudioLevel) {
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std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
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}
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std::cout << std::endl;
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}
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continue;
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}
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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(parsed_stream, rtcp_block, log_timestamp,
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direction);
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break;
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case webrtc::rtcp::ReceiverReport::kPacketType:
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PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
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case webrtc::ParsedRtcEventLog::RTCP_EVENT: {
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if (FLAG_rtcp) {
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size_t length;
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uint8_t packet[IP_PACKET_SIZE];
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webrtc::PacketDirection direction;
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parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
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webrtc::rtcp::CommonHeader rtcp_block;
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const uint8_t* packet_end = packet + length;
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for (const uint8_t* next_block = packet; next_block != packet_end;
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next_block = rtcp_block.NextPacket()) {
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ptrdiff_t remaining_blocks_size = packet_end - next_block;
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RTC_DCHECK_GT(remaining_blocks_size, 0);
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if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
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break;
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}
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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(parsed_stream, rtcp_block, log_timestamp,
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direction);
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break;
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case webrtc::rtcp::ReceiverReport::kPacketType:
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PrintReceiverReport(parsed_stream, rtcp_block, log_timestamp,
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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);
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break;
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case webrtc::rtcp::ExtendedReports::kPacketType:
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PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Bye::kPacketType:
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PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Rtpfb::kPacketType:
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PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
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direction);
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break;
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case webrtc::rtcp::Psfb::kPacketType:
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PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
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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);
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break;
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case webrtc::rtcp::ExtendedReports::kPacketType:
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PrintXr(parsed_stream, rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Bye::kPacketType:
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PrintBye(parsed_stream, rtcp_block, log_timestamp, direction);
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break;
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case webrtc::rtcp::Rtpfb::kPacketType:
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PrintRtpFeedback(parsed_stream, rtcp_block, log_timestamp,
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direction);
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break;
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case webrtc::rtcp::Psfb::kPacketType:
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PrintPsFeedback(parsed_stream, rtcp_block, log_timestamp,
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direction);
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break;
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default:
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break;
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break;
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default:
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break;
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}
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}
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}
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continue;
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}
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case webrtc::ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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case webrtc::ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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case webrtc::ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
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// TODO(eladalon): Implement in new CL.
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continue;
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}
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case webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
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if (FLAG_config && FLAG_video && FLAG_incoming) {
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webrtc::rtclog::StreamConfig config =
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parsed_stream.GetVideoReceiveConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
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<< "\tssrc=" << config.remote_ssrc
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<< "\tfeedback_ssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type
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<< "}";
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}
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std::cout << "}" << std::endl;
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}
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continue;
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}
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case webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
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if (FLAG_config && FLAG_video && FLAG_outgoing) {
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std::vector<webrtc::rtclog::StreamConfig> configs =
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parsed_stream.GetVideoSendConfig(i);
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for (const auto& config : configs) {
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std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
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std::cout << "\tssrcs=" << config.local_ssrc;
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std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type
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<< "}";
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}
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std::cout << "}" << std::endl;
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}
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}
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continue;
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}
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case webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
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if (FLAG_config && FLAG_audio && FLAG_incoming) {
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webrtc::rtclog::StreamConfig config =
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parsed_stream.GetAudioReceiveConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
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<< "\tssrc=" << config.remote_ssrc
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<< "\tfeedback_ssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", payload_type: " << codec.payload_type
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<< ", rtx_payload_type: " << codec.rtx_payload_type
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<< "}";
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}
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std::cout << "}" << std::endl;
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}
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continue;
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}
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case webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
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if (FLAG_config && FLAG_audio && FLAG_outgoing) {
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webrtc::rtclog::StreamConfig config =
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parsed_stream.GetAudioSendConfig(i);
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std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
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<< "\tssrc=" << config.local_ssrc;
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std::cout << "\textensions={";
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for (const auto& extension : config.rtp_extensions) {
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std::cout << extension.ToString() << ",";
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}
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std::cout << "}";
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std::cout << "\tcodecs={";
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for (const auto& codec : config.codecs) {
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std::cout << "{name: " << codec.payload_name
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<< ", 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;
|
||||
}
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user