Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

webrtc/voice_engine/channel.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 14 matching lines @@
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
 namespace webrtc {
@@ skipping 1499 matching lines @@
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceWarning,
         "DeRegisterExternalTransport() external transport already "
         "disabled");
   }
   _externalTransport = false;
   _transportPtr = NULL;
   return 0;
 }
 
+// TODO(nisse): Delete this method together with ReceivedRTPPacket.
+// It's a temporary hack to support both ReceivedRTPPacket and
+// OnRtpPacket interfaces without too much code duplication.
+bool Channel::OnRtpPacketWithHeader(const uint8_t* received_packet,
+                                    size_t length,
+                                    RTPHeader *header) {
+  // Store playout timestamp for the received RTP packet
+  UpdatePlayoutTimestamp(false);
+
+  header->payload_type_frequency =
+      rtp_payload_registry_->GetPayloadTypeFrequency(header->payloadType);
+  if (header->payload_type_frequency < 0)
+    return false;
+  bool in_order = IsPacketInOrder(*header);
+  rtp_receive_statistics_->IncomingPacket(
+      *header, length, IsPacketRetransmitted(*header, in_order));
+  rtp_payload_registry_->SetIncomingPayloadType(*header);
+
+  return ReceivePacket(received_packet, length, *header, in_order);
+}
+
 int32_t Channel::ReceivedRTPPacket(const uint8_t* received_packet,
                                    size_t length,
                                    const PacketTime& packet_time) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::ReceivedRTPPacket()");
 
-  // Store playout timestamp for the received RTP packet
-  UpdatePlayoutTimestamp(false);
-
   RTPHeader header;
   if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
                  "Incoming packet: invalid RTP header");
     return -1;
   }
-  header.payload_type_frequency =
-      rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
-  if (header.payload_type_frequency < 0)
-    return -1;
-  bool in_order = IsPacketInOrder(header);
-  rtp_receive_statistics_->IncomingPacket(
-      header, length, IsPacketRetransmitted(header, in_order));
-  rtp_payload_registry_->SetIncomingPayloadType(header);
+  return OnRtpPacketWithHeader(received_packet, length, &header) ? 0 : -1;
+}
 
-  return ReceivePacket(received_packet, length, header, in_order) ? 0 : -1;
+void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
+  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::ReceivedRTPPacket()");
+
+  RTPHeader header;
+  packet.GetHeader(&header);
+  OnRtpPacketWithHeader(packet.data(), packet.size(), &header);
 }
 
 bool Channel::ReceivePacket(const uint8_t* packet,
                             size_t packet_length,
                             const RTPHeader& header,
                             bool in_order) {
   if (rtp_payload_registry_->IsRtx(header)) {
     return HandleRtxPacket(packet, packet_length, header);
   }
   const uint8_t* payload = packet + header.headerLength;
@@ skipping 1460 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc