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Update to the RtcEventLog protobuf to remove the DebugEvent message.

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This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.

This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1348113003 .

Cr-Commit-Position: refs/heads/master@{#10221}
This commit is contained in:
Ivo Creusen
2015-10-08 18:07:41 +02:00
parent 8ac544e811
commit 301aaed813
4 changed files with 64 additions and 72 deletions
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19
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
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@ -47,16 +47,16 @@ const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
return &rtp_packet;
}
const rtclog::DebugEvent* GetAudioOutputEvent(const rtclog::Event& event) {
if (!event.has_type() || event.type() != rtclog::Event::DEBUG_EVENT)
const rtclog::AudioPlayoutEvent* GetAudioPlayoutEvent(
const rtclog::Event& event) {
if (!event.has_type() || event.type() != rtclog::Event::AUDIO_PLAYOUT_EVENT)
return nullptr;
if (!event.has_timestamp_us() || !event.has_debug_event())
if (!event.has_timestamp_us() || !event.has_audio_playout_event())
return nullptr;
const rtclog::DebugEvent& debug_event = event.debug_event();
if (!debug_event.has_type() ||
debug_event.type() != rtclog::DebugEvent::AUDIO_PLAYOUT)
const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
if (!playout_event.has_local_ssrc())
return nullptr;
return &debug_event;
return &playout_event;
}
} // namespace
@ -107,9 +107,10 @@ Packet* RtcEventLogSource::NextPacket() {
int64_t RtcEventLogSource::NextAudioOutputEventMs() {
while (audio_output_index_ < event_log_->stream_size()) {
const rtclog::Event& event = event_log_->stream(audio_output_index_);
const rtclog::DebugEvent* debug_event = GetAudioOutputEvent(event);
const rtclog::AudioPlayoutEvent* playout_event =
GetAudioPlayoutEvent(event);
audio_output_index_++;
if (debug_event)
if (playout_event)
return event.timestamp_us() / 1000;
}
return std::numeric_limits<int64_t>::max();