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

webrtc/modules/audio_coding/include/audio_coding_module.h

Index: webrtc/modules/audio_coding/include/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 305d8ea6d3c09acf2dc0f1098da7b0e37c86bdb2..7077cfa9c334ceaf4f5027e1c9632f1925b14b07 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -651,17 +651,10 @@ class AudioCodingModule {
   // 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().
+  // the latest audio obtained by calling PlayoutData10ms(), or empty if no
+  // valid timestamp is available.
   //
-  // 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;
+  virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
 
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(