Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 383 matching lines @@
   assert(vad_.get());
   vad_->Enable();
 }
 
 void NetEqImpl::DisableVad() {
   rtc::CritScope lock(&crit_sect_);
   assert(vad_.get());
   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 {
   rtc::CritScope lock(&crit_sect_);
   return last_output_sample_rate_hz_;
 }
 
 int NetEqImpl::SetTargetNumberOfChannels() {
   return kNotImplemented;
 }
@@ skipping 1661 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc