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

webrtc/modules/audio_coding/include/audio_coding_module.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_INCLUDE_AUDIO_CODING_MODULE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
 
 #include <string>
 #include <vector>
 
+#include "webrtc/base/deprecation.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
@@ skipping 613 matching lines @@
   virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
 
   //
   // The shortest latency, in milliseconds, required by jitter buffer. This
   // is computed based on inter-arrival times and playout mode of NetEq. The
   // actual delay is the maximum of least-required-delay and the minimum-delay
   // specified by SetMinumumPlayoutDelay() API.
   //
   virtual int LeastRequiredDelayMs() const = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
   // audio of the packet in question. This function returns the timestamp of
   // the latest audio obtained by calling PlayoutData10ms().
   //
   // Input:
   //   -timestamp          : a reference to a uint32_t to receive the
   //                         timestamp.
   // Return value:
   //    0 if the output is a correct timestamp.
   //   -1 if failed to output the correct timestamp.
   //
-  // TODO(tlegrand): Change function to return the timestamp.
-  virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
+  RTC_DEPRECATED virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
+
+  ///////////////////////////////////////////////////////////////////////////
+  // int32_t PlayoutTimestamp()
+  // The send timestamp of an RTP packet is associated with the decoded
+  // audio of the packet in question. This function returns the timestamp of
+  // the latest audio obtained by calling PlayoutData10ms(), or empty if no
+  // valid timestamp is available.
+  //
+  virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling
   // frequency. ACM will perform a resampling if required.
   //
   // Input:
   //   -desired_freq_hz    : the desired sampling frequency, in Hertz, of the
   //                         output audio. If set to -1, the function returns
   //                         the audio at the current sampling frequency.
@@ skipping 115 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_