Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index f22f1082621d5daf60fd1c0d5bcf40a4c95deccc..22f050b8d080b71079f82bf423905381d9f05c52 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -867,6 +867,12 @@ TEST_F(RtpDepacketizerH264Test, TestTruncatedSingleStapANalu) {
   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
+TEST_F(RtpDepacketizerH264Test, TestStapAPacketWithTruncatedNalUnits) {
+  const uint8_t kPayload[] = { 0x58, 0xCB, 0xED, 0xDF};

[nisse-webrtc, 2017/02/13 12:44:57]

This is the same payload as in the deleted testcase, but without 12 bytes of
headers.

+  RtpDepacketizer::ParsedPayload payload;
+  EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
+}
+
 TEST_F(RtpDepacketizerH264Test, TestTruncationJustAfterSingleStapANalu) {
   const uint8_t kPayload[] = {0x38, 0x27, 0x27};
   RtpDepacketizer::ParsedPayload payload;