Fix packetization logic to leave space for extensions in the last packet

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>  // max
@@ skipping 245 matching lines @@
   class VideoRotationObserver : public test::SendTest {
    public:
     VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
           kRtpExtensionVideoRotation, test::kVideoRotationExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      // Only the last packet of the frame is required to have the extension.
+      if (!header.markerBit)
+        return SEND_PACKET;
       EXPECT_TRUE(header.extension.hasVideoRotation);
       EXPECT_EQ(kVideoRotation_90, header.extension.videoRotation);
       observation_complete_.Set();
       return SEND_PACKET;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
@@ skipping 19 matching lines @@
   class VideoRotationObserver : public test::SendTest {
    public:
     VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
           kRtpExtensionVideoContentType, test::kVideoContentTypeExtensionId));
     }
 
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      // Only the last packet of the frame must have extension.
+      if (!header.markerBit)
+        return SEND_PACKET;
       EXPECT_TRUE(header.extension.hasVideoContentType);
       EXPECT_EQ(VideoContentType::SCREENSHARE,
                 header.extension.videoContentType);
       observation_complete_.Set();
       return SEND_PACKET;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
@@ skipping 495 matching lines @@
   // Normal NACKs should use the send SSRC.
   TestNackRetransmission(kVideoSendSsrcs[0], kFakeVideoSendPayloadType);
 }
 
 TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
   // NACKs over RTX should use a separate SSRC.
   TestNackRetransmission(kSendRtxSsrcs[0], kSendRtxPayloadType);
 }
 
 void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
                                                       bool with_fec) {

[sprang_webrtc, 2017/05/17 13:10:07]

Maybe would be a good idea to add optional last-packet extension header to this
test?

[ilnik, 2017/05/17 15:06:34]

No. It will complicate that test greatly, however, it's already tested in
module_unittests and video_engine_tests.

[sprang_webrtc, 2017/05/18 09:48:19]

Acknowledged.

   // Use a fake encoder to output a frame of every size in the range [90, 290],
   // for each size making sure that the exact number of payload bytes received
   // is correct and that packets are fragmented to respect max packet size.
   static const size_t kMaxPacketSize = 128;
   static const size_t start = 90;
   static const size_t stop = 290;
 
   // Observer that verifies that the expected number of packets and bytes
   // arrive for each frame size, from start_size to stop_size.
   class FrameFragmentationTest : public test::SendTest,
@@ skipping 2499 matching lines @@
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc