Implement timing frames.

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 319 matching lines @@
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
     }
   } test;
 
   test::ScopedFieldTrials override_field_trials(
       "WebRTC-VideoContentTypeExtension/Enabled/");
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, SupportsVideoTimingFrames) {
+  class VideoRotationObserver : public test::SendTest {
+   public:
+    VideoRotationObserver() : SendTest(kDefaultTimeoutMs) {
+      EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionVideoTiming, test::kVideoTimingExtensionId));
+    }
+
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      RTPHeader header;
+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
+      if (header.extension.has_video_timing) {
+        observation_complete_.Set();
+      }
+      return SEND_PACKET;
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(RtpExtension(
+          RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId));
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for timing frames.";
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
 class FakeReceiveStatistics : public NullReceiveStatistics {
  public:
   FakeReceiveStatistics(uint32_t send_ssrc,
                         uint32_t last_sequence_number,
                         uint32_t cumulative_lost,
                         uint8_t fraction_lost)
       : lossy_stats_(new LossyStatistician(last_sequence_number,
                                            cumulative_lost,
                                            fraction_lost)) {
     stats_map_[send_ssrc] = lossy_stats_.get();
@@ skipping 2984 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