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

webrtc/modules/audio_coding/acm2/acm_receiver.cc

 /*
  *  Copyright (c) 2013 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 178 matching lines @@
   memcpy(last_audio_buffer_.get(), audio_frame->data_,
          sizeof(int16_t) * audio_frame->samples_per_channel_ *
              audio_frame->num_channels_);
 
   call_stats_.DecodedByNetEq(audio_frame->speech_type_);
 
   // Computes the RTP timestamp of the first sample in |audio_frame| from
   // |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 -

[minyue-webrtc, 2016/04/04 15:53:50]

"*playout_timestamp -" may be moved to follow after "audio_frame->timestamp_ ="

[hlundin-webrtc, 2016/04/04 21:02:54]

Done. The extra line break is what git cl format gave me, but I do agree that it
is prettier to keep it together on one line.

         static_cast<uint32_t>(audio_frame->samples_per_channel_);
   } else {
     // Remain 0 until we have a valid |playout_timestamp|.
     audio_frame->timestamp_ = 0;
   }
 
   return 0;
 }
 
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
@@ skipping 99 matching lines @@
     return -1;
   }
   if (last_audio_decoder_ == &it->second) {
     last_audio_decoder_ = nullptr;
     last_packet_sample_rate_hz_ = rtc::Optional<int>();
   }
   decoders_.erase(it);
   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 {
   rtc::CritScope lock(&crit_sect_);
   if (!last_audio_decoder_) {
     return -1;
   }
   *codec = *RentACodec::CodecInstById(
       *RentACodec::CodecIdFromIndex(last_audio_decoder_->acm_codec_id));
   codec->pltype = last_audio_decoder_->payload_type;
@@ skipping 91 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc