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

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.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.
  */
 
@@ skipping 139 matching lines @@
 
   // Minimum playout delay.
   int SetMinimumPlayoutDelay(int time_ms) override;
 
   // Maximum playout delay.
   int SetMaximumPlayoutDelay(int time_ms) override;
 
   // Smallest latency NetEq will maintain.
   int LeastRequiredDelayMs() const override;
 
-  // Get playout timestamp.
-  int PlayoutTimestamp(uint32_t* timestamp) override;
+  RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
+
+  rtc::Optional<uint32_t> PlayoutTimestamp() override;
 
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
   int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
 
   /////////////////////////////////////////
   //   Statistics
   //
 
   int GetNetworkStatistics(NetworkStatistics* statistics) override;
@@ skipping 120 matching lines @@
   rtc::CriticalSection callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_