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Simplifies RtcEventProcessor interface.

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Bug: webrtc:10170
Change-Id: Ie643e47c55b8c35ca9b8ef31eda5b1673f19d7b3
Reviewed-on: https://webrtc-review.googlesource.com/c/116066
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26160}
This commit is contained in:
Sebastian Jansson
2019-01-08 15:31:06 +01:00
committed by Commit Bot
parent 37d18485dd
commit 03fbf1eb4b
7 changed files with 105 additions and 91 deletions
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12
modules/audio_coding/neteq/tools/rtc_event_log_source.cc
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@ -109,21 +109,13 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
if (ShouldSkipStream(media_type, rtp_packets.ssrc, ssrc_filter)) {
continue;
}
auto rtp_view = absl::make_unique<
webrtc::ProcessableEventList<webrtc::LoggedRtpPacketIncoming>>(
rtp_packets.incoming_packets.begin(),
rtp_packets.incoming_packets.end(), handle_rtp_packet);
event_processor.AddEvents(std::move(rtp_view));
event_processor.AddEvents(rtp_packets.incoming_packets, handle_rtp_packet);
}
for (const auto& audio_playouts : parsed_log.audio_playout_events()) {
if (ssrc_filter.has_value() && audio_playouts.first != *ssrc_filter)
continue;
auto audio_view = absl::make_unique<
webrtc::ProcessableEventList<webrtc::LoggedAudioPlayoutEvent>>(
audio_playouts.second.begin(), audio_playouts.second.end(),
handle_audio_playout);
event_processor.AddEvents(std::move(audio_view));
event_processor.AddEvents(audio_playouts.second, handle_audio_playout);
}
// Fills in rtp_packets_ and audio_outputs_.