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

webrtc/modules/audio_coding/neteq/include/neteq.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 
 #include <string.h>  // Provide access to size_t.
 
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class AudioFrame;
 struct WebRtcRTPHeader;
 
@@ skipping 207 matching lines @@
   // Same as RtcpStatistics(), but does not reset anything.
   virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
 
   // Enables post-decode VAD. When enabled, GetAudio() will return
   // kOutputVADPassive when the signal contains no speech.
   virtual void EnableVad() = 0;
 
   // Disables post-decode VAD.
   virtual void DisableVad() = 0;
 
-  // Gets the RTP timestamp for the last sample delivered by GetAudio().
-  // Returns true if the RTP timestamp is valid, otherwise false.
-  virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
+  // Returns the RTP timestamp for the last sample delivered by GetAudio().
+  // The return value will be empty if no valid timestamp is available.
+  virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() = 0;
 
   // Returns the sample rate in Hz of the audio produced in the last GetAudio
   // call. If GetAudio has not been called yet, the configured sample rate
   // (Config::sample_rate_hz) is returned.
   virtual int last_output_sample_rate_hz() const = 0;
 
   // Not implemented.
   virtual int SetTargetNumberOfChannels() = 0;
 
   // Not implemented.
@@ skipping 29 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_