Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d69654fb0f2451f1a5e49564f63ef90a32056aff..16f752cb273bbc33d368ab50a0d26d96f81217e7 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -32,6 +32,7 @@
 #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"
@@ -1551,31 +1552,49 @@ int32_t Channel::DeRegisterExternalTransport() {
   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,