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

webrtc/modules/audio_coding/acm2/acm_receiver.cc

Index: webrtc/modules/audio_coding/acm2/acm_receiver.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 5649f07b2bc46efdeab6b8150b56d055588000b1..61808976e52e7a73f8ad1e986cc4b89cf5c9a386 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -196,9 +196,9 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
   // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
   // |audio_frame|.
   // TODO(henrik.lundin) Move setting of audio_frame->timestamp_ inside NetEq.
-  uint32_t playout_timestamp = 0;
-  if (GetPlayoutTimestamp(&playout_timestamp)) {
-    audio_frame->timestamp_ = playout_timestamp -
+  rtc::Optional<uint32_t> playout_timestamp = GetPlayoutTimestamp();
+  if (playout_timestamp) {
+    audio_frame->timestamp_ = *playout_timestamp -
         static_cast<uint32_t>(audio_frame->samples_per_channel_);
   } else {
     // Remain 0 until we have a valid |playout_timestamp|.
@@ -318,8 +318,8 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
   return 0;
 }
 
-bool AcmReceiver::GetPlayoutTimestamp(uint32_t* timestamp) {
-  return neteq_->GetPlayoutTimestamp(timestamp);
+rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
+  return neteq_->GetPlayoutTimestamp();
 }
 
 int AcmReceiver::LastAudioCodec(CodecInst* codec) const {