Move setting of AudioFrame::timestamp_ into NetEq

webrtc/modules/audio_coding/neteq/neteq_impl.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 383 matching lines @@
   assert(vad_.get());
   vad_->Enable();
 }
 
 void NetEqImpl::DisableVad() {
   rtc::CritScope lock(&crit_sect_);
   assert(vad_.get());
   vad_->Disable();
 }
 
-rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() {
+rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
   rtc::CritScope lock(&crit_sect_);
   if (first_packet_) {
     // We don't have a valid RTP timestamp until we have decoded our first
     // RTP packet.
     return rtc::Optional<uint32_t>();
   }
   return rtc::Optional<uint32_t>(
       timestamp_scaler_->ToExternal(playout_timestamp_));
 }
 
@@ skipping 544 matching lines @@
     // would be moved "backwards".
     uint32_t temp_timestamp = sync_buffer_->end_timestamp() -
         static_cast<uint32_t>(sync_buffer_->FutureLength());
     if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {
       playout_timestamp_ = temp_timestamp;
     }
   } else {
     // Use dead reckoning to estimate the |playout_timestamp_|.
     playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
   }
+  // Set the timestamp in the audio frame to zero before the first packet has
+  // been inserted. Otherwise, subtract the frame size in samples to get the
+  // timestamp of the first sample in the frame (playout_timestamp_ is the
+  // last + 1).
+  audio_frame->timestamp_ =

[minyue-webrtc, 2016/04/06 11:16:26]

maybe too many lines here

[hlundin-webrtc, 2016/04/06 11:18:21]

This is what git cl format gave me, and I'd rather not fight it today. :)

+      first_packet_
+          ? 0
+          : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                static_cast<uint32_t>(audio_frame->samples_per_channel_);
 
   if (decode_return_value) return decode_return_value;
   return return_value;
 }
 
 int NetEqImpl::GetDecision(Operations* operation,
                            PacketList* packet_list,
                            DtmfEvent* dtmf_event,
                            bool* play_dtmf) {
   // Initialize output variables.
@@ skipping 1105 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc