Moved codec-specific audio packet splitting into decoders.

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 614 matching lines @@
   if (ret != PayloadSplitter::kOK) {
     PacketBuffer::DeleteAllPackets(&packet_list);
     switch (ret) {
       case PayloadSplitter::kUnknownPayloadType:
         return kUnknownRtpPayloadType;
       default:
         return kOtherError;
     }
   }
 
-  // Split payloads into smaller chunks. This also verifies that all payloads
-  // are of a known payload type.
-  ret = payload_splitter_->SplitAudio(&packet_list, *decoder_database_);
-  if (ret != PayloadSplitter::kOK) {
-    PacketBuffer::DeleteAllPackets(&packet_list);
-    switch (ret) {
-      case PayloadSplitter::kUnknownPayloadType:
-        return kUnknownRtpPayloadType;
-      case PayloadSplitter::kFrameSplitError:
-        return kFrameSplitError;
-      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 17 matching lines @@
       // Carry comfort noise packets along.
       parsed_packet_list.push_back(packet.release());
     } else {
       std::vector<AudioDecoder::ParseResult> results =
           info->GetDecoder()->ParsePayload(std::move(packet->payload),
                                            packet->header.timestamp,
                                            packet->primary);
       const RTPHeader& original_header = packet->header;
       for (auto& result : results) {
         RTC_DCHECK(result.frame);
-        // Reuse the packet if possible
+        // Reuse the packet if possible.
         if (!packet) {
           packet.reset(new Packet);
           packet->header = original_header;
         }
         packet->header.timestamp = result.timestamp;
         // TODO(ossu): Move from primary to some sort of priority level.
         packet->primary = result.primary;
         packet->frame = std::move(result.frame);
         parsed_packet_list.push_back(packet.release());
       }
@@ skipping 1401 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