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

webrtc/voice_engine/channel.cc

 /*
  *  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 3270 matching lines @@
                  ") != "
                  "fileSamples(%" PRIuS ")",
                  audioFrame.samples_per_channel_, fileSamples);
     return -1;
   }
 
   return 0;
 }
 
 void Channel::UpdatePlayoutTimestamp(bool rtcp) {
-  uint32_t playout_timestamp = 0;
+  rtc::Optional<uint32_t> playout_timestamp = audio_coding_->PlayoutTimestamp();
 
-  if (audio_coding_->PlayoutTimestamp(&playout_timestamp) == -1) {
+  if (!playout_timestamp) {
     // This can happen if this channel has not been received any RTP packet. In
     // this case, NetEq is not capable of computing playout timestamp.
     return;
   }
 
   uint16_t delay_ms = 0;
   if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::UpdatePlayoutTimestamp() failed to read playout"
                  " delay from the ADM");
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_RETRIEVE_VALUE, kTraceError,
         "UpdatePlayoutTimestamp() failed to retrieve playout delay");
     return;
   }
 
-  jitter_buffer_playout_timestamp_ = playout_timestamp;
+  jitter_buffer_playout_timestamp_ = *playout_timestamp;
 
   // Remove the playout delay.
-  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+  *playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
 
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
-               playout_timestamp);
+               *playout_timestamp);
 
   {
     rtc::CritScope lock(&video_sync_lock_);
     if (rtcp) {
-      playout_timestamp_rtcp_ = playout_timestamp;
+      playout_timestamp_rtcp_ = *playout_timestamp;
     } else {
-      playout_timestamp_rtp_ = playout_timestamp;
+      playout_timestamp_rtp_ = *playout_timestamp;
     }
     playout_delay_ms_ = delay_ms;
   }
 }
 
 // Called for incoming RTP packets after successful RTP header parsing.
 void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
                                 uint16_t sequence_number) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
@@ skipping 193 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc