RTCMediaStreamTrackStats.kind added and collected.

webrtc/api/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 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 1259 matching lines @@
   expected_remote_stream.stream_identifier = remote_stream->label();
   expected_remote_stream.track_ids = std::vector<std::string>();
   expected_remote_stream.track_ids->push_back(
       "RTCMediaStreamTrack_RemoteAudioTrackID");
   ASSERT_TRUE(report->Get(expected_remote_stream.id()));
   EXPECT_EQ(expected_remote_stream,
             report->Get(expected_remote_stream.id())->cast_to<
                 RTCMediaStreamStats>());
 
   RTCMediaStreamTrackStats expected_local_audio_track(
-      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us());
+      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us(),
+      RTCMediaStreamTrackKind::kAudio);
   expected_local_audio_track.track_identifier = local_audio_track->id();
   expected_local_audio_track.remote_source = false;
   expected_local_audio_track.ended = true;
   expected_local_audio_track.detached = false;
   expected_local_audio_track.audio_level = 1.0;
   expected_local_audio_track.echo_return_loss = 42.0;
   expected_local_audio_track.echo_return_loss_enhancement = 52.0;
   ASSERT_TRUE(report->Get(expected_local_audio_track.id()));
   EXPECT_EQ(expected_local_audio_track,
             report->Get(expected_local_audio_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 
   RTCMediaStreamTrackStats expected_remote_audio_track(
-      "RTCMediaStreamTrack_RemoteAudioTrackID", report->timestamp_us());
+      "RTCMediaStreamTrack_RemoteAudioTrackID", report->timestamp_us(),
+      RTCMediaStreamTrackKind::kAudio);
   expected_remote_audio_track.track_identifier = remote_audio_track->id();
   expected_remote_audio_track.remote_source = true;
   expected_remote_audio_track.ended = false;
   expected_remote_audio_track.detached = false;
   expected_remote_audio_track.audio_level = 0.0;
   expected_remote_audio_track.echo_return_loss = 13.0;
   expected_remote_audio_track.echo_return_loss_enhancement = 37.0;
   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())->cast_to<
@@ skipping 19 matching lines @@
       FakeAudioTrackForStats::Create(
           "LocalAudioTrackID",
           MediaStreamTrackInterface::TrackState::kEnded,
           32767,
           new FakeAudioProcessorForStats(local_audio_processor_stats));
   local_stream->AddTrack(local_audio_track);
 
   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
 
   RTCMediaStreamTrackStats expected_local_audio_track(
-      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us());
+      "RTCMediaStreamTrack_LocalAudioTrackID", report->timestamp_us(),
+      RTCMediaStreamTrackKind::kAudio);
   expected_local_audio_track.track_identifier = local_audio_track->id();
   expected_local_audio_track.remote_source = false;
   expected_local_audio_track.ended = true;
   expected_local_audio_track.detached = false;
   expected_local_audio_track.audio_level = 1.0;
   // Should be undefined: |expected_local_audio_track.echo_return_loss| and
   // |expected_local_audio_track.echo_return_loss_enhancement|.
   ASSERT_TRUE(report->Get(expected_local_audio_track.id()));
   EXPECT_EQ(expected_local_audio_track,
             report->Get(expected_local_audio_track.id())->cast_to<
@@ skipping 62 matching lines @@
   expected_remote_stream.stream_identifier = remote_stream->label();
   expected_remote_stream.track_ids = std::vector<std::string>();
   expected_remote_stream.track_ids->push_back(
       "RTCMediaStreamTrack_RemoteVideoTrackID");
   ASSERT_TRUE(report->Get(expected_remote_stream.id()));
   EXPECT_EQ(expected_remote_stream,
             report->Get(expected_remote_stream.id())->cast_to<
                 RTCMediaStreamStats>());
 
   RTCMediaStreamTrackStats expected_local_video_track(
-      "RTCMediaStreamTrack_LocalVideoTrackID", report->timestamp_us());
+      "RTCMediaStreamTrack_LocalVideoTrackID", report->timestamp_us(),
+      RTCMediaStreamTrackKind::kVideo);
   expected_local_video_track.track_identifier = local_video_track->id();
   expected_local_video_track.remote_source = false;
   expected_local_video_track.ended = false;
   expected_local_video_track.detached = false;
   expected_local_video_track.frame_width = 1234;
   expected_local_video_track.frame_height = 4321;
   ASSERT_TRUE(report->Get(expected_local_video_track.id()));
   EXPECT_EQ(expected_local_video_track,
             report->Get(expected_local_video_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
 
   RTCMediaStreamTrackStats expected_remote_video_track(
-      "RTCMediaStreamTrack_RemoteVideoTrackID", report->timestamp_us());
+      "RTCMediaStreamTrack_RemoteVideoTrackID", report->timestamp_us(),
+      RTCMediaStreamTrackKind::kVideo);
   expected_remote_video_track.track_identifier = remote_video_track->id();
   expected_remote_video_track.remote_source = true;
   expected_remote_video_track.ended = true;
   expected_remote_video_track.detached = false;
   expected_remote_video_track.frame_width = 1234;
   expected_remote_video_track.frame_height = 4321;
   ASSERT_TRUE(report->Get(expected_remote_video_track.id()));
   EXPECT_EQ(expected_remote_video_track,
             report->Get(expected_remote_video_track.id())->cast_to<
                 RTCMediaStreamTrackStats>());
@@ skipping 566 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc