Resurrected test_api_audio.cc

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
index ab666cec2c4c616ab9382c69c154a7f793faf072..437499ac4b919e2239585e7630b5950d0ef5b393 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -44,7 +44,7 @@ bool LoopBackTransport::SendRtp(const uint8_t* data,
   }
   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;
@@ -52,9 +52,11 @@ bool LoopBackTransport::SendRtp(const uint8_t* data,
                                                   &payload_specific)) {
     return false;
   }
+  const uint8_t* payload = data + header.headerLength;
+  RTC_DCHECK_GE(len, header.headerLength);

[hlundin-webrtc, 2016/10/04 13:45:02]

This is test code, so you might just as well RTC_CHECK_GE.

[ossu, 2016/10/04 14:55:29]

Alright.

+  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;
   }