Revert of Use sps and pps to determine decodability of H.264 frames.

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 753 matching lines @@
   payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet2, sizeof(packet2)));
   ExpectPacket(&payload, kExpected2, sizeof(kExpected2));
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_FALSE(payload.type.Video.isFirstPacket);
   {
     const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
     EXPECT_EQ(kH264FuA, h264.packetization_type);
     EXPECT_EQ(kIdr, h264.nalu_type);
-    // NALU info is only expected for the first FU-A packet.
-    EXPECT_EQ(0u, h264.nalus_length);
+    ASSERT_EQ(1u, h264.nalus_length);
+    EXPECT_EQ(static_cast<H264::NaluType>(kIdr), h264.nalus[0].type);
+    EXPECT_EQ(-1, h264.nalus[0].sps_id);
+    EXPECT_EQ(-1, h264.nalus[0].pps_id);
   }
 
   payload = RtpDepacketizer::ParsedPayload();
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet3, sizeof(packet3)));
   ExpectPacket(&payload, kExpected3, sizeof(kExpected3));
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_FALSE(payload.type.Video.isFirstPacket);
   {
     const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
     EXPECT_EQ(kH264FuA, h264.packetization_type);
     EXPECT_EQ(kIdr, h264.nalu_type);
-    // NALU info is only expected for the first FU-A packet.
-    ASSERT_EQ(0u, h264.nalus_length);
+    ASSERT_EQ(1u, h264.nalus_length);
+    EXPECT_EQ(static_cast<H264::NaluType>(kIdr), h264.nalus[0].type);
+    EXPECT_EQ(-1, h264.nalus[0].sps_id);
+    EXPECT_EQ(-1, h264.nalus[0].pps_id);
   }
 }
 
 TEST_F(RtpDepacketizerH264Test, TestEmptyPayload) {
   // Using a wild pointer to crash on accesses from inside the depacketizer.
   uint8_t* garbage_ptr = reinterpret_cast<uint8_t*>(0x4711);
   RtpDepacketizer::ParsedPayload payload;
   EXPECT_FALSE(depacketizer_->Parse(&payload, garbage_ptr, 0));
 }
 
@@ skipping 15 matching lines @@
   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)));
 }
 
 }  // namespace webrtc