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

webrtc/modules/audio_coding/acm2/audio_coding_module_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 884 matching lines @@
 }
 
 int AudioCodingModuleImpl::DisableOpusDtx() {
   rtc::CritScope lock(&acm_crit_sect_);
   if (!HaveValidEncoder("DisableOpusDtx")) {
     return -1;
   }
   return encoder_stack_->SetDtx(false) ? 0 : -1;
 }
 
-int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
+int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
+  rtc::Optional<uint32_t> ts = PlayoutTimestamp();
+  if (!ts)
+    return -1;
+  *timestamp = *ts;
+  return 0;
+}
+
+rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
+  return receiver_.GetPlayoutTimestamp();
 }
 
 bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
   if (!encoder_stack_) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "%s failed: No send codec is registered.", caller_name);
     return false;
   }
   return true;
 }
@@ skipping 19 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc