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.
  */
 
@@ skipping 34 matching lines @@
   RTPHeader header;
   std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
   if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
     return false;
   }
   PayloadUnion payload_specific;
   if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
                                                   &payload_specific)) {
     return false;
   }
+  const uint8_t* payload =
+      static_cast<const uint8_t*>(data) + header.headerLength;

[danilchap, 2016/10/03 13:51:19]

static_cast is not needed here (it casts to same type data already is).

[ossu, 2016/10/03 14:12:03]

Acknowledged.

+  assert(len >= header.headerLength);

[danilchap, 2016/10/03 13:51:19]

RTC_DCHECK_GE(len, header.headerLength)

[ossu, 2016/10/03 14:12:03]

You are right on both counts! I just borrowed this code and didn't clean it up.
I'll get on it!

+  const size_t payload_length = len - header.headerLength;

[danilchap, 2016/10/03 13:51:19]

may be just "size_t payload_length":
function tail is short, payload_length used just once there, probably no point
to stress it is not changed next line.

[ossu, 2016/10/03 14:12:03]

Granted, it's not strictly necessary. However, I'm trying to put const wherever
a variable doesn't change, for the few benefits that can be gotten in the
general case (mostly getting slapped if accidentally changing the value).

   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