Add H264 bitstream rewriting to limit frame reordering marker in header

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 396 matching lines @@
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
   EXPECT_EQ(kH264SingleNalu,
             payload.type.Video.codecHeader.H264.packetization_type);
   EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestSingleNaluSpsWithResolution) {
   uint8_t packet[] = {kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50,
                       0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
-                      0x00, 0x00, 0x2A, 0xE0, 0xF1, 0x83, 0x19, 0x60};
+                      0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25};

[stefan-webrtc, 2016/05/22 23:11:50]

Why did this change?

[sprang_webrtc, 2016/05/25 09:06:03]

Parsing revealed that VUI contained:
max_num_reorder_frames	uint32_t = 2	
max_dec_frame_buffering	uint32_t = 4	

So, I added a modified version in there.

   RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(&payload, packet, sizeof(packet));
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
   EXPECT_EQ(kH264SingleNalu,
             payload.type.Video.codecHeader.H264.packetization_type);
   EXPECT_EQ(1280u, payload.type.Video.width);
@@ skipping 14 matching lines @@
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
   EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
   // NALU type for aggregated packets is the type of the first packet only.
   EXPECT_EQ(kSps, payload.type.Video.codecHeader.H264.nalu_type);
 }
 
 TEST_F(RtpDepacketizerH264Test, TestStapANaluSpsWithResolution) {
   uint8_t packet[] = {kStapA,  // F=0, NRI=0, Type=24.
                       // Length (2 bytes), nal header, payload.
-                      0,      24,   kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9,
-                      0x40,   0x50, 0x05, 0xBA, 0x10, 0x00, 0x00, 0x03,
-                      0x00,   0xC0, 0x00, 0x00, 0x2A, 0xE0, 0xF1, 0x83,
-                      0x19,   0x60, 0,    0x03, kIdr, 0xFF, 0x00, 0,
-                      0x04,   kIdr, 0xFF, 0x00, 0x11};
+                      0x00, 0x19, kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40,
+                      0x50, 0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
+                      0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25, 0x80,
+                      0x00, 0x03, kIdr, 0xFF, 0x00, 0x00, 0x04, kIdr, 0xFF,
+                      0x00, 0x11};

[stefan-webrtc, 2016/05/22 23:11:50]

... and this?

[sprang_webrtc, 2016/05/25 09:06:03]

...same here. How did you come up with these magic numbers?

[stefan-webrtc, 2016/05/26 18:08:13]

I think pbos added these.

[noahric, 2016/05/26 18:56:52]

And I think they came from my code originally, which had some comments
explaining them. See:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

+
   RtpDepacketizer::ParsedPayload payload;
 
   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
   ExpectPacket(&payload, packet, sizeof(packet));
   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
   EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
   EXPECT_TRUE(payload.type.Video.isFirstPacket);
   EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
   EXPECT_EQ(1280u, payload.type.Video.width);
   EXPECT_EQ(720u, payload.type.Video.height);
@@ skipping 94 matching lines @@
   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)));
 }
 

[stefan-webrtc, 2016/05/22 23:11:50]

Would be good with explicit test cases for the sps rewriting here.

[sprang_webrtc, 2016/05/25 09:06:03]

Yes, in the pipeline.

 }  // namespace webrtc