| 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 {
|
|
|