Move setting of AudioFrame::timestamp_ into NetEq

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 173 matching lines @@
     resampled_last_output_frame_ = false;
     // We might end up here ONLY if codec is changed.
   }
 
   // Store current audio in |last_audio_buffer_| for next time.
   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.
-  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|.
-    audio_frame->timestamp_ = 0;
-  }
-
   return 0;
 }
 
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
                               size_t channels,
                               int sample_rate_hz,
                               AudioDecoder* audio_decoder,
                               const std::string& name) {
   const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
@@ skipping 206 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc