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

webrtc/modules/audio_coding/test/APITest.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 648 matching lines @@
 }
 
 // Set Min delay, get delay, playout timestamp
 void APITest::TestDelay(char side) {
   AudioCodingModule* myACM;
   Channel* myChannel;
   int32_t* myMinDelay;
   EventTimerWrapper* myEvent = EventTimerWrapper::Create();
 
   uint32_t inTimestamp = 0;
-  uint32_t outTimestamp = 0;
   double estimDelay = 0;
 
   double averageEstimDelay = 0;
   double averageDelay = 0;
 
   CircularBuffer estimDelayCB(100);
   estimDelayCB.SetArithMean(true);
 
   if (side == 'A') {
     myACM = _acmA.get();
     myChannel = _channel_B2A;
     myMinDelay = &_minDelayA;
   } else {
     myACM = _acmB.get();
     myChannel = _channel_A2B;
     myMinDelay = &_minDelayB;
   }
 
   CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
 
   inTimestamp = myChannel->LastInTimestamp();
-  CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
+  rtc::Optional<uint32_t> outTimestamp = myACM->PlayoutTimestamp();
+  CHECK_ERROR_MT(outTimestamp ? 0 : -1);
 
   if (!_randomTest) {
     myEvent->StartTimer(true, 30);
     int n = 0;
     int settlePoint = 5000;
     while (n < settlePoint + 400) {
       myEvent->Wait(1000);
 
       inTimestamp = myChannel->LastInTimestamp();
-      CHECK_ERROR_MT(myACM->PlayoutTimestamp(&outTimestamp));
+      outTimestamp = myACM->PlayoutTimestamp();
+      CHECK_ERROR_MT(outTimestamp ? 0 : -1);
 
       //std::cout << outTimestamp << std::endl << std::flush;
-      estimDelay = (double) ((uint32_t)(inTimestamp - outTimestamp))
-          / ((double) myACM->ReceiveFrequency() / 1000.0);
+      estimDelay = (double)((uint32_t)(inTimestamp - *outTimestamp)) /
+                   ((double)myACM->ReceiveFrequency() / 1000.0);
 
       estimDelayCB.Update(estimDelay);
 
       estimDelayCB.ArithMean(averageEstimDelay);
       //printf("\n %6.1f \n", estimDelay);
       //std::cout << " " << std::flush;
 
       if (_verbose) {
         fprintf(stdout,
                 "\rExpected: %4d,    retreived: %6.1f,   measured: %6.1f",
@@ skipping 379 matching lines @@
   CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
   myChannel->ResetStats();
   {
     WriteLockScoped wl(_apiTestRWLock);
     *thereIsEncoder = true;
   }
   Wait(500);
 }
 
 }  // namespace webrtc