zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Stop simulations when a LOG_END event is reached.

Browse Source 
When a LOG_END event is reached, it makes no sense to continue simulating NetEq.

Bug: webrtc:9667
Change-Id: Ie4f6811cdec0d0632f6e7906059e0e74e9f10438
Reviewed-on: https://webrtc-review.googlesource.com/c/105643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25176}
This commit is contained in:
Ivo Creusen
2018-10-15 09:55:51 +02:00
committed by Commit Bot
parent 961dbeac82
commit ed04912ccd
1 changed files with 14 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
modules/audio_coding/neteq/tools/rtc_event_log_source.cc
View File

@ -68,9 +68,16 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
if (!parsed_log.ParseFile(file_name)) if (!parsed_log.ParseFile(file_name))
return false; return false;
const auto first_log_end_time_us =
parsed_log.stop_log_events().empty()
? std::numeric_limits<int64_t>::max()
: parsed_log.stop_log_events().front().log_time_us();
auto handle_rtp_packet = auto handle_rtp_packet =
[this](const webrtc::LoggedRtpPacketIncoming& incoming) { [this,
if (!filter_.test(incoming.rtp.header.payloadType)) { first_log_end_time_us](const webrtc::LoggedRtpPacketIncoming& incoming) {
if (!filter_.test(incoming.rtp.header.payloadType) &&
incoming.log_time_us() < first_log_end_time_us) {
rtp_packets_.emplace_back(absl::make_unique<Packet>( rtp_packets_.emplace_back(absl::make_unique<Packet>(
incoming.rtp.header, incoming.rtp.total_length, incoming.rtp.header, incoming.rtp.total_length,
incoming.rtp.total_length - incoming.rtp.header_length, incoming.rtp.total_length - incoming.rtp.header_length,
@ -79,8 +86,11 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
}; };
auto handle_audio_playout = auto handle_audio_playout =
[this](const webrtc::LoggedAudioPlayoutEvent& audio_playout) { [this, first_log_end_time_us](
audio_outputs_.emplace_back(audio_playout.log_time_ms()); const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
if (audio_playout.log_time_us() < first_log_end_time_us) {
audio_outputs_.emplace_back(audio_playout.log_time_ms());
}
}; };
// This wouldn't be needed if we knew that there was at most one audio stream. // This wouldn't be needed if we knew that there was at most one audio stream.