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

webrtc/modules/audio_coding/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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 177 matching lines @@
   // Return value             : 0 if OK.
   //                           -1 if an error occurred.
   //
   int RemoveCodec(uint8_t payload_type);
 
   //
   // Remove all registered codecs.
   //
   int RemoveAllCodecs();
 
-  //
-  // Gets the RTP timestamp of the last sample delivered by GetAudio().
-  // Returns true if the RTP timestamp is valid, otherwise false.
-  //
-  bool GetPlayoutTimestamp(uint32_t* timestamp);
+  // Returns the RTP timestamp for the last sample delivered by GetAudio().
+  // The return value will be empty if no valid timestamp is available.
+  rtc::Optional<uint32_t> GetPlayoutTimestamp();
 
   //
   // Get the audio codec associated with the last non-CNG/non-DTMF received
   // payload. If no non-CNG/non-DTMF packet is received -1 is returned,
   // otherwise return 0.
   //
   int LastAudioCodec(CodecInst* codec) const;
 
   //
   // Get a decoder given its registered payload-type.
@@ skipping 59 matching lines @@
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
   rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_