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Print media types for streams using the new event log format.

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This CL moves the code that stores SSRCs used by different media types
so that it will be used by the new format too. This is sufficient to
get the correct media types printed in e.g. event_log_visualizer.

Bug: webrtc:8111
Change-Id: Ife11bc49b2af7577c7b5326c0b0fadd2e5b48b94
Reviewed-on: https://webrtc-review.googlesource.com/c/113942
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25965}
This commit is contained in:
Bjorn Terelius
2018-12-11 13:46:19 +01:00
committed by Commit Bot
parent 698d6c4f30
commit ae786b82b6
2 changed files with 20 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

28
logging/rtc_event_log/rtc_event_log_parser_new.cc
View File

@ -942,6 +942,24 @@ bool ParsedRtcEventLogNew::ParseStream(
Clear();
bool success = ParseStreamInternal(stream);
// Cache the configured SSRCs.
for (const auto& video_recv_config : video_recv_configs()) {
incoming_video_ssrcs_.insert(video_recv_config.config.remote_ssrc);
incoming_video_ssrcs_.insert(video_recv_config.config.rtx_ssrc);
incoming_rtx_ssrcs_.insert(video_recv_config.config.rtx_ssrc);
}
for (const auto& video_send_config : video_send_configs()) {
outgoing_video_ssrcs_.insert(video_send_config.config.local_ssrc);
outgoing_video_ssrcs_.insert(video_send_config.config.rtx_ssrc);
outgoing_rtx_ssrcs_.insert(video_send_config.config.rtx_ssrc);
}
for (const auto& audio_recv_config : audio_recv_configs()) {
incoming_audio_ssrcs_.insert(audio_recv_config.config.remote_ssrc);
}
for (const auto& audio_send_config : audio_send_configs()) {
outgoing_audio_ssrcs_.insert(audio_send_config.config.local_ssrc);
}
// ParseStreamInternal stores the RTP packets in a map indexed by SSRC.
// Since we dont need rapid lookup based on SSRC after parsing, we move the
// packets_streams from map to vector.
@ -998,10 +1016,8 @@ bool ParsedRtcEventLogNew::ParseStream(
// stream configurations and starting/stopping the log.
// TODO(terelius): Figure out if we actually need to find the first and last
// timestamp in the parser. It seems like this could be done by the caller.
first_timestamp_ = std::numeric_limits<int64_t>::max();
last_timestamp_ = std::numeric_limits<int64_t>::min();
StoreFirstAndLastTimestamp(alr_state_events());
for (const auto& audio_stream : audio_playout_events()) {
// Audio playout events are grouped by SSRC.
@ -1131,9 +1147,6 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
incoming_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
incoming_video_ssrcs_.insert(config.remote_ssrc);
incoming_video_ssrcs_.insert(config.rtx_ssrc);
incoming_rtx_ssrcs_.insert(config.rtx_ssrc);
break;
}
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT: {
@ -1145,9 +1158,6 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
outgoing_video_ssrcs_.insert(config.local_ssrc);
outgoing_video_ssrcs_.insert(config.rtx_ssrc);
outgoing_rtx_ssrcs_.insert(config.rtx_ssrc);
break;
}
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: {
@ -1157,7 +1167,6 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
incoming_rtp_extensions_maps_[config.remote_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
incoming_audio_ssrcs_.insert(config.remote_ssrc);
break;
}
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: {
@ -1167,7 +1176,6 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
outgoing_audio_ssrcs_.insert(config.local_ssrc);
break;
}
case rtclog::Event::RTP_EVENT: {

2
logging/rtc_event_log/rtc_event_log_parser_new.h
View File

@ -1080,6 +1080,8 @@ class ParsedRtcEventLogNew {
// The extension maps are mutable to allow us to insert the default
// configuration when parsing an RTP header for an unconfigured stream.
// TODO(terelius): This is only used for the legacy format. Remove once we've
// fully transitioned to the new format.
mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>
incoming_rtp_extensions_maps_;
mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>