Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

 /*
  *  Copyright (c) 2014 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 849 matching lines @@
   RtpDepacketizer::ParsedPayload payload;
   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 TEST_F(RtpDepacketizerH264Test, TestTruncatedSingleStapANalu) {
   const uint8_t kPayload[] = {0xd8, 0x27};
   RtpDepacketizer::ParsedPayload payload;
   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;
   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 TEST_F(RtpDepacketizerH264Test, TestShortSpsPacket) {
   const uint8_t kPayload[] = {0x27, 0x80, 0x00};
   RtpDepacketizer::ParsedPayload payload;
   EXPECT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 TEST_F(RtpDepacketizerH264Test, TestSeiPacket) {
   const uint8_t kPayload[] = {
       kSei,  // F=0, NRI=0, Type=6.
       0x03, 0x03, 0x03, 0x03  // Payload.
   };
   RtpDepacketizer::ParsedPayload payload;
   ASSERT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
   const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
   EXPECT_EQ(kH264SingleNalu, h264.packetization_type);
   EXPECT_EQ(kSei, h264.nalu_type);
   ASSERT_EQ(1u, h264.nalus_length);
   EXPECT_EQ(static_cast<H264::NaluType>(kSei), h264.nalus[0].type);
   EXPECT_EQ(-1, h264.nalus[0].sps_id);
   EXPECT_EQ(-1, h264.nalus[0].pps_id);
 }
 
 }  // namespace webrtc