Revert of Add content type information to encoded images and corresponding rtp extension header

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 2634 matching lines @@
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
 
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
 
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs"));
-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs"));
+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
 
   EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
@@ skipping 19 matching lines @@
 
   int num_red_samples = use_red ? 1 : 0;
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
   EXPECT_EQ(num_red_samples,
             metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
 }
 
-TEST_F(EndToEndTest, ContentTypeSwitches) {
-  class StatsObserver : public test::BaseTest,
-                        public rtc::VideoSinkInterface<VideoFrame> {
-   public:
-    StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {}
-
-    bool ShouldCreateReceivers() const override { return true; }
-
-    void OnFrame(const VideoFrame& video_frame) override {
-      // The RTT is needed to estimate |ntp_time_ms| which is used by
-      // end-to-end delay stats. Therefore, start counting received frames once
-      // |ntp_time_ms| is valid.
-      if (video_frame.ntp_time_ms() > 0 &&
-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
-              video_frame.ntp_time_ms()) {
-        rtc::CritScope lock(&crit_);
-        ++num_frames_received_;
-      }
-    }
-
-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
-      if (MinNumberOfFramesReceived())
-        observation_complete_.Set();
-      return SEND_PACKET;
-    }
-
-    bool MinNumberOfFramesReceived() const {
-      const int kMinRequiredHistogramSamples = 200;
-      rtc::CritScope lock(&crit_);
-      return num_frames_received_ > kMinRequiredHistogramSamples;
-    }
-
-    // May be called several times.
-    void PerformTest() override {
-      EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
-      // Reset frame counter so next PerformTest() call will do something.
-      {
-        rtc::CritScope lock(&crit_);
-        num_frames_received_ = 0;
-      }
-    }
-
-    rtc::CriticalSection crit_;
-    int num_frames_received_ GUARDED_BY(&crit_);
-  } test;
-
-  metrics::Reset();
-
-  Call::Config send_config(test.GetSenderCallConfig());
-  CreateSenderCall(send_config);
-  Call::Config recv_config(test.GetReceiverCallConfig());
-  CreateReceiverCall(recv_config);
-  receive_transport_.reset(test.CreateReceiveTransport());
-  send_transport_.reset(test.CreateSendTransport(sender_call_.get()));
-  send_transport_->SetReceiver(receiver_call_->Receiver());
-  receive_transport_->SetReceiver(sender_call_->Receiver());
-  receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
-  CreateSendConfig(1, 0, 0, send_transport_.get());
-  CreateMatchingReceiveConfigs(receive_transport_.get());
-
-  // Modify send and receive configs.
-  video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-  video_receive_configs_[0].renderer = &test;
-  // RTT needed for RemoteNtpTimeEstimator for the receive stream.
-  video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report = true;
-  // Start with realtime video.
-  video_encoder_config_.content_type =
-      VideoEncoderConfig::ContentType::kRealtimeVideo;
-  // Second encoder config for the second part of the test uses screenshare
-  VideoEncoderConfig encoder_config_with_screenshare_ =
-      video_encoder_config_.Copy();
-  encoder_config_with_screenshare_.content_type =
-      VideoEncoderConfig::ContentType::kScreen;
-
-  CreateVideoStreams();
-  CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
-                               kDefaultHeight);
-  Start();
-
-  test.PerformTest();
-
-  // Replace old send stream.
-  sender_call_->DestroyVideoSendStream(video_send_stream_);
-  video_send_stream_ = sender_call_->CreateVideoSendStream(
-      video_send_config_.Copy(), encoder_config_with_screenshare_.Copy());
-  video_send_stream_->SetSource(
-      frame_generator_capturer_.get(),
-      VideoSendStream::DegradationPreference::kBalanced);
-  video_send_stream_->Start();
-
-  // Continue to run test but now with screenshare.
-  test.PerformTest();
-
-  send_transport_->StopSending();
-  receive_transport_->StopSending();
-  Stop();
-  DestroyStreams();
-  DestroyCalls();
-  // Delete the call for Call stats to be reported.
-  sender_call_.reset();
-  receiver_call_.reset();
-
-  // Verify that stats have been updated for both screenshare and video.
-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
-  EXPECT_EQ(1,
-            metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
-  EXPECT_EQ(
-      1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
-}
-
 TEST_F(EndToEndTest, VerifyHistogramStatsWithRtx) {
   const bool kEnabledRtx = true;
   const bool kEnabledRed = false;
   const bool kScreenshare = false;
   VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
 }
 
 TEST_F(EndToEndTest, VerifyHistogramStatsWithRed) {
   const bool kEnabledRtx = false;
   const bool kEnabledRed = true;
@@ skipping 1645 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