NetEq: Use TickTimer in PacketBuffer

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 518 matching lines @@
     // is not defined outside of this block.
     Packet* packet = new Packet;
     packet->header.markerBit = false;
     packet->header.payloadType = rtp_header.header.payloadType;
     packet->header.sequenceNumber = rtp_header.header.sequenceNumber;
     packet->header.timestamp = rtp_header.header.timestamp;
     packet->header.ssrc = rtp_header.header.ssrc;
     packet->header.numCSRCs = 0;
     packet->payload_length = payload.size();
     packet->primary = true;
-    packet->waiting_time = 0;
+    // Waiting time will be set upon inserting the packet in the buffer.
+    RTC_DCHECK(!packet->waiting_time);
     packet->payload = new uint8_t[packet->payload_length];
     packet->sync_packet = is_sync_packet;
     if (!packet->payload) {
       LOG_F(LS_ERROR) << "Payload pointer is NULL.";
     }
     assert(!payload.empty());  // Already checked above.
     memcpy(packet->payload, payload.data(), packet->payload_length);
     // Insert packet in a packet list.
     packet_list.push_back(packet);
     // Save main payloads header for later.
@@ skipping 445 matching lines @@
 
 int NetEqImpl::GetDecision(Operations* operation,
                            PacketList* packet_list,
                            DtmfEvent* dtmf_event,
                            bool* play_dtmf) {
   // Initialize output variables.
   *play_dtmf = false;
   *operation = kUndefined;
 
   // Increment time counters.
-  packet_buffer_->IncrementWaitingTimes();
   stats_.IncreaseCounter(output_size_samples_, fs_hz_);
 
   assert(sync_buffer_.get());
   uint32_t end_timestamp = sync_buffer_->end_timestamp();
   if (!new_codec_) {
     const uint32_t five_seconds_samples = 5 * fs_hz_;
     packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples);
   }
   const RTPHeader* header = packet_buffer_->NextRtpHeader();
 
@@ skipping 908 matching lines @@
     size_t discard_count = 0;
     Packet* packet = packet_buffer_->GetNextPacket(&discard_count);
     // |header| may be invalid after the |packet_buffer_| operation.
     header = NULL;
     if (!packet) {
       LOG(LS_ERROR) << "Should always be able to extract a packet here";
       assert(false);  // Should always be able to extract a packet here.
       return -1;
     }
     stats_.PacketsDiscarded(discard_count);
-    // Store waiting time in ms; packets->waiting_time is in "output blocks".
-    stats_.StoreWaitingTime(packet->waiting_time * kOutputSizeMs);
+    stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
     assert(packet->payload_length > 0);
     packet_list->push_back(packet);  // Store packet in list.
 
     if (first_packet) {
       first_packet = false;
       if (nack_enabled_) {
         RTC_DCHECK(nack_);
         // TODO(henrik.lundin): Should we update this for all decoded packets?
         nack_->UpdateLastDecodedPacket(packet->header.sequenceNumber,
                                        packet->header.timestamp);
@@ skipping 160 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