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 265 matching lines @@
     }
     return kFail;
   }
   return kOK;
 }
 
 int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
   rtc::CritScope lock(&crit_sect_);
   int ret = decoder_database_->Remove(rtp_payload_type);
   if (ret == DecoderDatabase::kOK) {
     packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type);

[hlundin-webrtc, 2016/09/20 07:02:24]

This is very courteous...

[ossu, 2016/09/20 13:51:56]

Thanks?

     return kOK;
   } else if (ret == DecoderDatabase::kDecoderNotFound) {
     error_code_ = kDecoderNotFound;
   } else {
     error_code_ = kOtherError;
   }
   return kFail;
 }
 
 bool NetEqImpl::SetMinimumDelay(int delay_ms) {
@@ skipping 225 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 21 matching lines @@
 
     // Trigger an update of sampling rate and the number of channels.
     update_sample_rate_and_channels = true;
   }
 
   // Update RTCP statistics, only for regular packets.
   rtcp_.Update(main_header, receive_timestamp);
 
   // Check for RED payload type, and separate payloads into several packets.
   if (decoder_database_->IsRed(main_header.payloadType)) {
-    if (payload_splitter_->SplitRed(&packet_list) != PayloadSplitter::kOK) {
+    if (!payload_splitter_->SplitRed(&packet_list)) {
       PacketBuffer::DeleteAllPackets(&packet_list);
       return kRedundancySplitError;
     }
     // Only accept a few RED payloads of the same type as the main data,
     // DTMF events and CNG.
     payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_);
     // Update the stored main payload header since the main payload has now
     // changed.
     memcpy(&main_header, &packet_list.front()->header, sizeof(main_header));
   }
@@ skipping 28 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 12 matching lines @@
       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(std::move(packet->payload),
-                                           packet->header.timestamp,
-                                           packet->primary);
+                                           packet->header.timestamp);
       const RTPHeader& original_header = packet->header;
+      const Packet::Priority original_priority = packet->priority;
       for (auto& result : results) {
         RTC_DCHECK(result.frame);
         // 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->priority.codec_level = result.priority;
+        packet->priority.red_level = original_priority.red_level;
         packet->frame = std::move(result.frame);
         parsed_packet_list.push_back(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(
+  const int ret = packet_buffer_->InsertPacketList(
       &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 1224 matching lines @@
       }
       prev_sequence_number = packet->header.sequenceNumber;
       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) {
+      // TODO(ossu): Is this the correct way to track Opus FEC packets?
+      if (packet->priority.codec_level > 0) {
         stats_.SecondaryDecodedSamples(rtc::checked_cast<int>(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 127 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