Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc |
index 14e105123e5bb5aabab03a4734664d174679e1e9..9e3b94b6d7861d3f05af9b7e45cef5f3155f69b1 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc |
@@ -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(); |