Report timing frames info in GetStats.

webrtc/video/end_to_end_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>
@@ skipping 3424 matching lines @@
     std::set<uint32_t> expected_send_ssrcs_;
     std::string expected_cname_;
 
     rtc::Event check_stats_event_;
     ReceiveStreamRenderer receive_stream_renderer_;
   } test;
 
   RunBaseTest(&test);
 }
 
+TEST_F(EndToEndTest, GetTimingFrameInfoReportsTimingFrames) {
+  static const int kExtensionId = 5;
+
+  class StatsObserver : public test::EndToEndTest {
+   public:
+    StatsObserver() : EndToEndTest(kLongTimeoutMs) {}
+
+   private:
+    std::string CompoundKey(const char* name, uint32_t ssrc) {
+      std::ostringstream oss;
+      oss << name << "_" << ssrc;
+      return oss.str();
+    }
+
+    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, kExtensionId));
+      for (size_t i = 0; i < receive_configs->size(); ++i) {
+        (*receive_configs)[i].rtp.extensions.clear();
+        (*receive_configs)[i].rtp.extensions.push_back(
+            RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
+      }
+    }
+
+    void OnVideoStreamsCreated(
+        VideoSendStream* send_stream,
+        const std::vector<VideoReceiveStream*>& receive_streams) override {
+      receive_streams_ = receive_streams;
+    }
+
+    void PerformTest() override {
+      // No frames reported initially.
+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
+        EXPECT_FALSE(receive_streams_[i]->GetAndResetTimingFrameInfo());
+      }
+      // Wait for at least one timing frame to be sent with 100ms grace period.
+      SleepMs(kDefaultTimingFramesDelayMs + 100);
+      // Check that timing frames are reported for each stream.
+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
+        EXPECT_TRUE(receive_streams_[i]->GetAndResetTimingFrameInfo());
+      }
+    }
+
+    std::vector<VideoReceiveStream*> receive_streams_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
 class RtcpXrObserver : public test::EndToEndTest {
  public:
   RtcpXrObserver(bool enable_rrtr, bool enable_target_bitrate)
       : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
         enable_rrtr_(enable_rrtr),
         enable_target_bitrate_(enable_target_bitrate),
         sent_rtcp_sr_(0),
         sent_rtcp_rr_(0),
         sent_rtcp_rrtr_(0),
         sent_rtcp_target_bitrate_(false),
@@ skipping 1249 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc