Add FullStack test for simulcast screenshare mode.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 "webrtc/video/video_quality_test.h"
 
 #include <stdio.h>
 #include <algorithm>
 #include <deque>
 #include <map>
+#include <set>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/cpu_time.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/memory_usage.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/call.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
+#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/layer_filtering_transport.h"
 #include "webrtc/test/run_loop.h"
 #include "webrtc/test/statistics.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/vcm_capturer.h"
 #include "webrtc/test/video_renderer.h"
@@ skipping 87 matching lines @@
  public:
   VideoAnalyzer(test::LayerFilteringTransport* transport,
                 const std::string& test_label,
                 double avg_psnr_threshold,
                 double avg_ssim_threshold,
                 int duration_frames,
                 FILE* graph_data_output_file,
                 const std::string& graph_title,
                 uint32_t ssrc_to_analyze,
                 uint32_t rtx_ssrc_to_analyze,
-                uint32_t selected_stream_width,
-                uint32_t selected_stream_height,
+                size_t selected_stream,
                 int selected_sl,
                 int selected_tl,
                 bool is_quick_test_enabled)
       : transport_(transport),
         receiver_(nullptr),
         send_stream_(nullptr),
         receive_stream_(nullptr),
         captured_frame_forwarder_(this),
         test_label_(test_label),
         graph_data_output_file_(graph_data_output_file),
         graph_title_(graph_title),
         ssrc_to_analyze_(ssrc_to_analyze),
         rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
-        selected_stream_width_(selected_stream_width),
-        selected_stream_height_(selected_stream_height),
+        selected_stream_(selected_stream),
         selected_sl_(selected_sl),
         selected_tl_(selected_tl),
         pre_encode_proxy_(this),
         encode_timing_proxy_(this),
         frames_to_process_(duration_frames),
         frames_recorded_(0),
         frames_processed_(0),
         dropped_frames_(0),
         dropped_frames_before_first_encode_(0),
         dropped_frames_before_rendering_(0),
@@ skipping 108 matching lines @@
         RTC_CHECK(!frames_.empty());
       }
       first_encoded_timestamp_ =
           rtc::Optional<uint32_t>(video_frame.timestamp());
     }
   }
 
   void PostEncodeFrameCallback(const EncodedFrame& encoded_frame) {
     rtc::CritScope lock(&crit_);
     if (!first_sent_timestamp_ &&
-        encoded_frame.encoded_width_ == selected_stream_width_ &&
-        encoded_frame.encoded_height_ == selected_stream_height_) {
+        encoded_frame.stream_id_ == selected_stream_) {
       first_sent_timestamp_ = rtc::Optional<uint32_t>(encoded_frame.timestamp_);
     }
   }
 
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
@@ skipping 651 matching lines @@
   }
 
   VideoSendStream* send_stream_;
   VideoReceiveStream* receive_stream_;
   CapturedFrameForwarder captured_frame_forwarder_;
   const std::string test_label_;
   FILE* const graph_data_output_file_;
   const std::string graph_title_;
   const uint32_t ssrc_to_analyze_;
   const uint32_t rtx_ssrc_to_analyze_;
-  const uint32_t selected_stream_width_;
-  const uint32_t selected_stream_height_;
+  const size_t selected_stream_;
   const int selected_sl_;
   const int selected_tl_;
   PreEncodeProxy pre_encode_proxy_;
   OnEncodeTimingProxy encode_timing_proxy_;
   std::vector<Sample> samples_ GUARDED_BY(comparison_lock_);
   std::map<int64_t, int> samples_encode_time_ms_ GUARDED_BY(comparison_lock_);
   test::Statistics sender_time_ GUARDED_BY(comparison_lock_);
   test::Statistics receiver_time_ GUARDED_BY(comparison_lock_);
   test::Statistics psnr_ GUARDED_BY(comparison_lock_);
   test::Statistics ssim_ GUARDED_BY(comparison_lock_);
@@ skipping 39 matching lines @@
   bool is_quick_test_enabled_;
 
   rtc::CriticalSection comparison_lock_;
   std::vector<rtc::PlatformThread*> comparison_thread_pool_;
   rtc::PlatformThread stats_polling_thread_;
   rtc::Event comparison_available_event_;
   std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
   rtc::Event done_;
 };
 
+class Vp8EncoderFactory : public VideoEncoderFactory {
+ public:
+  Vp8EncoderFactory() = default;
+  ~Vp8EncoderFactory() override { RTC_CHECK(live_encoders_.empty()); }
+
+  VideoEncoder* Create() override {
+    VideoEncoder* encoder = VP8Encoder::Create();
+    live_encoders_.insert(encoder);
+    return encoder;
+  }
+
+  void Destroy(VideoEncoder* encoder) override {
+    auto it = live_encoders_.find(encoder);
+    RTC_CHECK(it != live_encoders_.end());
+    live_encoders_.erase(it);
+    delete encoder;
+  }
+
+  std::set<VideoEncoder*> live_encoders_;
+};
+
 VideoQualityTest::VideoQualityTest()
     : clock_(Clock::GetRealTimeClock()), receive_logs_(0), send_logs_(0) {}
 
 VideoQualityTest::Params::Params()
     : call({false, Call::Config::BitrateConfig()}),
       video({false, 640, 480, 30, 50, 800, 800, false, "VP8", 1, -1, 0, false,
              false, "", ""}),
       audio({false, false}),
       screenshare({false, 10, 0}),
       analyzer({"", 0.0, 0.0, 0, "", ""}),
@@ skipping 45 matching lines @@
   // TODO(ivica): Should max_bitrate_bps == -1 represent inf max bitrate, as it
   // does in some parts of the code?
   RTC_CHECK_GE(params_.video.max_bitrate_bps, params_.video.target_bitrate_bps);
   RTC_CHECK_GE(params_.video.target_bitrate_bps, params_.video.min_bitrate_bps);
   RTC_CHECK_LT(params_.video.selected_tl, params_.video.num_temporal_layers);
   RTC_CHECK_LT(params_.ss.selected_stream, params_.ss.streams.size());
   for (const VideoStream& stream : params_.ss.streams) {
     RTC_CHECK_GE(stream.min_bitrate_bps, 0);
     RTC_CHECK_GE(stream.target_bitrate_bps, stream.min_bitrate_bps);
     RTC_CHECK_GE(stream.max_bitrate_bps, stream.target_bitrate_bps);
-    RTC_CHECK_EQ(stream.temporal_layer_thresholds_bps.size(),
+    RTC_CHECK_LE(stream.temporal_layer_thresholds_bps.size(),
                  params_.video.num_temporal_layers - 1);
   }
   // TODO(ivica): Should we check if the sum of all streams/layers is equal to
   // the total bitrate? We anyway have to update them in the case bitrate
   // estimator changes the total bitrates.
   RTC_CHECK_GE(params_.ss.num_spatial_layers, 1);
   RTC_CHECK_LE(params_.ss.selected_sl, params_.ss.num_spatial_layers);
   RTC_CHECK(params_.ss.spatial_layers.empty() ||
             params_.ss.spatial_layers.size() ==
                 static_cast<size_t>(params_.ss.num_spatial_layers));
@@ skipping 42 matching lines @@
 // Static.
 VideoStream VideoQualityTest::DefaultVideoStream(const Params& params) {
   VideoStream stream;
   stream.width = params.video.width;
   stream.height = params.video.height;
   stream.max_framerate = params.video.fps;
   stream.min_bitrate_bps = params.video.min_bitrate_bps;
   stream.target_bitrate_bps = params.video.target_bitrate_bps;
   stream.max_bitrate_bps = params.video.max_bitrate_bps;
   stream.max_qp = 52;
-  if (params.video.num_temporal_layers == 2) {
-    stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
-  } else if (params.video.num_temporal_layers == 3) {
-    stream.temporal_layer_thresholds_bps.push_back(stream.max_bitrate_bps / 4);
-    stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
+  if (params.video.num_temporal_layers > 1) {
+    RTC_CHECK_LE(params.video.num_temporal_layers, kMaxTemporalStreams);
+    if (params.video.codec == "VP8") {
+      for (int i = 0; i < params.video.num_temporal_layers - 1; ++i) {
+        stream.temporal_layer_thresholds_bps.push_back(static_cast<int>(
+            stream.target_bitrate_bps *
+            kVp8LayerRateAlloction[params.video.num_temporal_layers][i]));
+      }
+    }

[stefan-webrtc, 2017/03/09 14:07:46]

Is this a functional change or just a rewrite?

[ilnik, 2017/03/09 14:34:35]

It's a more correct way to do it. Before we had some random numbers I randomly
chosen. Now it's in line with other places.

   }
   return stream;
 }
 
 // Static.
 VideoStream VideoQualityTest::DefaultThumbnailStream() {
   VideoStream stream;
   stream.width = 320;
   stream.height = 180;
   stream.max_framerate = 7;
@@ skipping 72 matching lines @@
 
   size_t num_video_streams = params_.ss.streams.size();
   size_t num_flexfec_streams = params_.video.flexfec ? 1 : 0;
   CreateSendConfig(num_video_streams, 0, num_flexfec_streams, send_transport);
 
   int payload_type;
   if (params_.video.codec == "H264") {
     video_encoder_.reset(H264Encoder::Create(cricket::VideoCodec("H264")));
     payload_type = kPayloadTypeH264;
   } else if (params_.video.codec == "VP8") {
-    video_encoder_.reset(VP8Encoder::Create());
+    if (params_.screenshare.enabled && params_.ss.streams.size() > 1) {
+      // Simulcast screenshare needs a simulcast encoder adapter to work, since
+      // encoders usually can't natively do simulcast with different frame rates
+      // for the different layers.
+      video_encoder_.reset(
+          new SimulcastEncoderAdapter(new Vp8EncoderFactory()));
+    } else {
+      video_encoder_.reset(VP8Encoder::Create());
+    }
     payload_type = kPayloadTypeVP8;
   } else if (params_.video.codec == "VP9") {
     video_encoder_.reset(VP9Encoder::Create());
     payload_type = kPayloadTypeVP9;
   } else {
     RTC_NOTREACHED() << "Codec not supported!";
     return;
   }
   video_send_config_.encoder_settings.encoder = video_encoder_.get();
   video_send_config_.encoder_settings.payload_name = params_.video.codec;
@@ skipping 314 matching lines @@
 
   test::LayerFilteringTransport send_transport(
       params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params_.video.selected_tl, params_.ss.selected_sl);
   test::DirectTransport recv_transport(params_.pipe, receiver_call_.get());
 
   std::string graph_title = params_.analyzer.graph_title;
   if (graph_title.empty())
     graph_title = VideoQualityTest::GenerateGraphTitle();
 
-  VideoStream& selected_stream = params_.ss.streams[params_.ss.selected_stream];
-
   bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
   VideoAnalyzer analyzer(
       &send_transport, params_.analyzer.test_label,
       params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
       is_quick_test_enabled
           ? kFramesSentInQuickTest
           : params_.analyzer.test_durations_secs * params_.video.fps,
       graph_data_output_file, graph_title,
       kVideoSendSsrcs[params_.ss.selected_stream],
       kSendRtxSsrcs[params_.ss.selected_stream],
-      static_cast<uint32_t>(selected_stream.width),
-      static_cast<uint32_t>(selected_stream.height), params.ss.selected_sl,
+      static_cast<size_t>(params_.ss.selected_stream), params.ss.selected_sl,
       params_.video.selected_tl, is_quick_test_enabled);
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupVideo(&analyzer, &recv_transport);
   SetupThumbnails(&analyzer, &recv_transport);
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
   video_send_config_.pre_encode_callback = analyzer.pre_encode_proxy();
   RTC_DCHECK(!video_send_config_.post_encode_callback);
@@ skipping 249 matching lines @@
     std::string prefix =
         params_.video.encoded_frame_base_path + "." + str.str() + ".send.";
     stream->EnableEncodedFrameRecording(
         std::vector<rtc::PlatformFile>(
             {rtc::CreatePlatformFile(prefix + "1.ivf"),
              rtc::CreatePlatformFile(prefix + "2.ivf"),
              rtc::CreatePlatformFile(prefix + "3.ivf")}),
         10000000);
   }
 }
+
 void VideoQualityTest::StartEncodedFrameLogs(VideoReceiveStream* stream) {
   if (!params_.video.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
-
 }  // namespace webrtc