Moved Opus-specific payload splitting into AudioDecoderOpus.

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 510 matching lines @@
     // directly once it's been inserted in the packet list. This way, |packet|
     // 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.SetData(payload.data(), payload.size());
-    packet->primary = true;
     // Waiting time will be set upon inserting the packet in the buffer.
     RTC_DCHECK(!packet->waiting_time);
     // Insert packet in a packet list.
     packet_list.push_back(packet);
     // Save main payloads header for later.
     memcpy(&main_header, &packet->header, sizeof(main_header));
   }
 
   bool update_sample_rate_and_channels = false;
   // Reinitialize NetEq if it's needed (changed SSRC or first call).
@@ skipping 70 matching lines @@
         PacketBuffer::DeleteAllPackets(&packet_list);
         return kDtmfInsertError;
       }
       delete current_packet;
       it = packet_list.erase(it);
     } else {
       ++it;
     }
   }
 
-  // Check for FEC in packets, and separate payloads into several packets.
-  int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get());
-  if (ret != PayloadSplitter::kOK) {
-    PacketBuffer::DeleteAllPackets(&packet_list);
-    switch (ret) {
-      case PayloadSplitter::kUnknownPayloadType:
-        return kUnknownRtpPayloadType;
-      default:
-        return kOtherError;
-    }
-  }
-
   // Update bandwidth estimate, if the packet is not comfort noise.
   if (!packet_list.empty() &&
       !decoder_database_->IsComfortNoise(main_header.payloadType)) {
     // The list can be empty here if we got nothing but DTMF payloads.
     AudioDecoder* decoder =
         decoder_database_->GetDecoder(main_header.payloadType);
     assert(decoder);  // Should always get a valid object, since we have
     // already checked that the payload types are known.
     decoder->IncomingPacket(packet_list.front()->payload.data(),
                             packet_list.front()->payload.size(),
@@ skipping 11 matching lines @@
     if (!info) {
       LOG(LS_WARNING) << "SplitAudio unknown payload type";
       return kUnknownRtpPayloadType;
     }
 
     if (info->IsComfortNoise()) {
       // Carry comfort noise packets along.
       parsed_packet_list.push_back(packet.release());
     } else {
       std::vector<AudioDecoder::ParseResult> results =
-          info->GetDecoder()->ParsePayload(
-              &packet->payload, packet->header.timestamp, packet->primary);
+          info->GetDecoder()->ParsePayload(&packet->payload,
+                                           packet->header.timestamp);
 
       // Reuse the packet if possible
       if (results.size() == 1) {
         auto& result = results[0];
         RTC_DCHECK(result.frame);
         packet->header.timestamp = result.timestamp;
-        packet->primary = result.primary;
+        packet->priority.codec_level = result.priority;
         packet->frame = std::move(result.frame);
         parsed_packet_list.push_back(packet.release());
       } else {
         for (auto& result : results) {
           // If you can't parse, return an empty list instead!
           RTC_DCHECK(result.frame);
           std::unique_ptr<Packet> parsed_packet(new Packet);
           parsed_packet->header = packet->header;
           parsed_packet->header.timestamp = result.timestamp;
-          // TODO(ossu): Move from primary to some sort of priority level.
-          parsed_packet->primary = result.primary;
+          parsed_packet->priority.codec_level = result.priority;
+          parsed_packet->priority.red_level = packet->priority.red_level;
           parsed_packet->frame = std::move(result.frame);
           parsed_packet_list.push_back(parsed_packet.release());
         }
       }
     }
   }
 
   if (nack_enabled_) {
     RTC_DCHECK(nack_);
     if (update_sample_rate_and_channels) {
       nack_->Reset();
     }
     nack_->UpdateLastReceivedPacket(
         parsed_packet_list.front()->header.sequenceNumber,
         parsed_packet_list.front()->header.timestamp);
   }
 
   // Insert packets in buffer.
   const size_t buffer_length_before_insert =
       packet_buffer_->NumPacketsInBuffer();
-  ret = packet_buffer_->InsertPacketList(
+  int ret = packet_buffer_->InsertPacketList(

[kwiberg-webrtc, 2016/09/19 11:07:49]

const?

[ossu, 2016/09/19 11:41:01]

Acknowledged.

       &parsed_packet_list, *decoder_database_, &current_rtp_payload_type_,
       &current_cng_rtp_payload_type_);
   if (ret == PacketBuffer::kFlushed) {
     // Reset DSP timestamp etc. if packet buffer flushed.
     new_codec_ = true;
     update_sample_rate_and_channels = true;
   } else if (ret != PacketBuffer::kOK) {
     PacketBuffer::DeleteAllPackets(&parsed_packet_list);
     return kOtherError;
   }
@@ skipping 1226 matching lines @@
       prev_timestamp = packet->header.timestamp;
       prev_payload_type = packet->header.payloadType;
     }
 
     // Store number of extracted samples.
     size_t packet_duration = 0;
     if (packet->frame) {
       packet_duration = packet->frame->Duration();
       // TODO(ossu): Is this the correct way to track samples decoded from a
       // redundant packet?
-      if (packet_duration > 0 && !packet->primary) {
+      if (packet->priority != Packet::kHighestPriority) {
         stats_.SecondaryDecodedSamples(packet_duration);
       }
     } else if (!decoder_database_->IsComfortNoise(packet->header.payloadType)) {
       LOG(LS_WARNING) << "Unknown payload type "
                       << static_cast<int>(packet->header.payloadType);
       RTC_NOTREACHED();
     }
 
     if (packet_duration == 0) {
       // Decoder did not return a packet duration. Assume that the packet
@@ skipping 128 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(),
       tick_timer_.get()));
 }
 }  // namespace webrtc