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Don't add empty extension list in event log parser.

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This allows the fall back list to be used instead.

Bug: webrtc:9718
Change-Id: Ie17a4b740fef60385c6019ea167c73eff07e8ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/111246
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25676}
This commit is contained in:
Sebastian Jansson 2018-11-16 18:40:09 +01:00 committed by Commit Bot
parent 1eebec9808
commit 8c1e73b024
1 changed files with 25 additions and 17 deletions
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42
logging/rtc_event_log/rtc_event_log_parser_new.cc
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@ -1014,13 +1014,15 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: { case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetVideoReceiveConfig(event); rtclog::StreamConfig config = GetVideoReceiveConfig(event);
video_recv_configs_.emplace_back(GetTimestamp(event), config); video_recv_configs_.emplace_back(GetTimestamp(event), config);
incoming_rtp_extensions_maps_[config.remote_ssrc] = if (!config.rtp_extensions.empty()) {
RtpHeaderExtensionMap(config.rtp_extensions); incoming_rtp_extensions_maps_[config.remote_ssrc] =
// TODO(terelius): I don't understand the reason for configuring header RtpHeaderExtensionMap(config.rtp_extensions);
// extensions for the local SSRC. I think it should be removed, but for // TODO(terelius): I don't understand the reason for configuring header
// now I want to preserve the previous functionality. // extensions for the local SSRC. I think it should be removed, but for
incoming_rtp_extensions_maps_[config.local_ssrc] = // now I want to preserve the previous functionality.
RtpHeaderExtensionMap(config.rtp_extensions); incoming_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
incoming_video_ssrcs_.insert(config.remote_ssrc); incoming_video_ssrcs_.insert(config.remote_ssrc);
incoming_video_ssrcs_.insert(config.rtx_ssrc); incoming_video_ssrcs_.insert(config.rtx_ssrc);
incoming_rtx_ssrcs_.insert(config.rtx_ssrc); incoming_rtx_ssrcs_.insert(config.rtx_ssrc);
@ -1030,10 +1032,12 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event); std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
video_send_configs_.emplace_back(GetTimestamp(event), configs); video_send_configs_.emplace_back(GetTimestamp(event), configs);
for (const auto& config : configs) { for (const auto& config : configs) {
outgoing_rtp_extensions_maps_[config.local_ssrc] = if (!config.rtp_extensions.empty()) {
RtpHeaderExtensionMap(config.rtp_extensions); outgoing_rtp_extensions_maps_[config.local_ssrc] =
outgoing_rtp_extensions_maps_[config.rtx_ssrc] = RtpHeaderExtensionMap(config.rtp_extensions);
RtpHeaderExtensionMap(config.rtp_extensions); outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
outgoing_video_ssrcs_.insert(config.local_ssrc); outgoing_video_ssrcs_.insert(config.local_ssrc);
outgoing_video_ssrcs_.insert(config.rtx_ssrc); outgoing_video_ssrcs_.insert(config.rtx_ssrc);
outgoing_rtx_ssrcs_.insert(config.rtx_ssrc); outgoing_rtx_ssrcs_.insert(config.rtx_ssrc);
@ -1043,18 +1047,22 @@ void ParsedRtcEventLogNew::StoreParsedLegacyEvent(const rtclog::Event& event) {
case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: { case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioReceiveConfig(event); rtclog::StreamConfig config = GetAudioReceiveConfig(event);
audio_recv_configs_.emplace_back(GetTimestamp(event), config); audio_recv_configs_.emplace_back(GetTimestamp(event), config);
incoming_rtp_extensions_maps_[config.remote_ssrc] = if (!config.rtp_extensions.empty()) {
RtpHeaderExtensionMap(config.rtp_extensions); incoming_rtp_extensions_maps_[config.remote_ssrc] =
incoming_rtp_extensions_maps_[config.local_ssrc] = RtpHeaderExtensionMap(config.rtp_extensions);
RtpHeaderExtensionMap(config.rtp_extensions); incoming_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
incoming_audio_ssrcs_.insert(config.remote_ssrc); incoming_audio_ssrcs_.insert(config.remote_ssrc);
break; break;
} }
case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: { case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT: {
rtclog::StreamConfig config = GetAudioSendConfig(event); rtclog::StreamConfig config = GetAudioSendConfig(event);
audio_send_configs_.emplace_back(GetTimestamp(event), config); audio_send_configs_.emplace_back(GetTimestamp(event), config);
outgoing_rtp_extensions_maps_[config.local_ssrc] = if (!config.rtp_extensions.empty()) {
RtpHeaderExtensionMap(config.rtp_extensions); outgoing_rtp_extensions_maps_[config.local_ssrc] =
RtpHeaderExtensionMap(config.rtp_extensions);
}
outgoing_audio_ssrcs_.insert(config.local_ssrc); outgoing_audio_ssrcs_.insert(config.local_ssrc);
break; break;
} }