Print media types for streams using the new event log format.
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}diff --git a/logging/rtc_event_log/rtc_event_log_parser_new.cc b/logging/rtc_event_log/rtc_event_log_parser_new.cc
index 6dab853..b8beff4 100644
--- a/logging/rtc_event_log/rtc_event_log_parser_new.cc
+++ b/logging/rtc_event_log/rtc_event_log_parser_new.cc
@@ -942,6 +942,24 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
- outgoing_audio_ssrcs_.insert(config.local_ssrc);
break;
}
case rtclog::Event::RTP_EVENT: {
diff --git a/logging/rtc_event_log/rtc_event_log_parser_new.h b/logging/rtc_event_log/rtc_event_log_parser_new.h
index b215601..903993f 100644
--- a/logging/rtc_event_log/rtc_event_log_parser_new.h
+++ b/logging/rtc_event_log/rtc_event_log_parser_new.h
@@ -1080,6 +1080,8 @@
// 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>
