Resurrected test_api_audio.cc

webrtc/modules/rtp_rtcp/test/testAPI/test_api.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.
  */
 
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/test/null_transport.h"
 
 namespace webrtc {
 
 void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
                                       RTPPayloadRegistry* payload_registry,
                                       RtpReceiver* receiver,
                                       ReceiveStatistics* receive_statistics) {
   rtp_rtcp_module_ = rtp_rtcp_module;
@@ skipping 10 matching lines @@
                                 size_t len,
                                 const PacketOptions& options) {
   count_++;
   if (packet_loss_ > 0) {
     if ((count_ % packet_loss_) == 0) {
       return true;
     }
   }
   RTPHeader header;
   std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-  if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
+  if (!parser->Parse(data, len, &header)) {
     return false;
   }
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
     return false;
   }
+  const uint8_t* payload = data + header.headerLength;
+  RTC_CHECK_GE(len, header.headerLength);
+  const size_t payload_length = len - header.headerLength;
   receive_statistics_->IncomingPacket(header, len, false);
-  if (!rtp_receiver_->IncomingRtpPacket(header,
-                                        static_cast<const uint8_t*>(data), len,
+  if (!rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
                                         payload_specific, true)) {
     return false;
   }
   return true;
 }
 
 bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
   if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
     return false;
   }
@@ skipping 114 matching lines @@
   rtx_header.payloadType = kRtxPayloadType;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = 0;
   EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }
 
 }  // namespace webrtc