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Only use GetAudio events that correspond to an ssrc matching at least one incoming packet.

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Using GetAudio events from SSRCs without incoming packets doesn't make sense, and should be prevented.

Bug: b/116685514
Change-Id: I48e38bb780549c71cb5f68d370a6819634ad487d
Reviewed-on: https://webrtc-review.googlesource.com/c/114321
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26017}
This commit is contained in:
Ivo Creusen
2018-12-14 14:43:11 +01:00
committed by Commit Bot
parent 24779d8229
commit 94f107454e
1 changed files with 19 additions and 5 deletions
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22
modules/audio_coding/neteq/tools/rtc_event_log_source.cc
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@ -13,6 +13,7 @@
#include <string.h>
#include <iostream>
#include <limits>
#include <set>
#include <utility>
#include "logging/rtc_event_log/rtc_event_processor.h"
@ -73,23 +74,30 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
? std::numeric_limits<int64_t>::max()
: parsed_log.stop_log_events().front().log_time_us();
std::set<uint32_t> packet_ssrcs;
auto handle_rtp_packet =
[this,
first_log_end_time_us](const webrtc::LoggedRtpPacketIncoming& incoming) {
[this, first_log_end_time_us,
&packet_ssrcs](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>(
incoming.rtp.header, incoming.rtp.total_length,
incoming.rtp.total_length - incoming.rtp.header_length,
static_cast<double>(incoming.log_time_ms())));
packet_ssrcs.insert(rtp_packets_.back()->header().ssrc);
}
};
std::set<uint32_t> ignored_ssrcs;
auto handle_audio_playout =
[this, first_log_end_time_us](
const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
[this, first_log_end_time_us, &packet_ssrcs,
&ignored_ssrcs](const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
if (audio_playout.log_time_us() < first_log_end_time_us) {
if (packet_ssrcs.count(audio_playout.ssrc) > 0) {
audio_outputs_.emplace_back(audio_playout.log_time_ms());
} else {
ignored_ssrcs.insert(audio_playout.ssrc);
}
}
};
@ -121,6 +129,12 @@ bool RtcEventLogSource::OpenFile(const std::string& file_name,
// Fills in rtp_packets_ and audio_outputs_.
event_processor.ProcessEventsInOrder();
for (const auto& ssrc : ignored_ssrcs) {
std::cout << "Ignoring GetAudio events from SSRC 0x" << std::hex << ssrc
<< " because no packets were found with a matching SSRC."
<< std::endl;
}
return true;
}