Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
index aa973811082593c01217c9915e8b2116f78c1e95..b4dfc9986dace2a1b94bf8545785ce83a83c12bd 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc |
@@ -401,15 +401,15 @@ void NetEqImpl::DisableVad() { |
vad_->Disable(); |
} |
-bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) { |
+rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() { |
rtc::CritScope lock(&crit_sect_); |
if (first_packet_) { |
// We don't have a valid RTP timestamp until we have decoded our first |
// RTP packet. |
- return false; |
+ return rtc::Optional<uint32_t>(); |
} |
- *timestamp = timestamp_scaler_->ToExternal(playout_timestamp_); |
- return true; |
+ return rtc::Optional<uint32_t>( |
+ timestamp_scaler_->ToExternal(playout_timestamp_)); |
} |
int NetEqImpl::last_output_sample_rate_hz() const { |