Initialize/configure video encoders asychronously.

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 <stdio.h>
@@ skipping 25 matching lines @@
 
 namespace webrtc {
 
 static const int kSendStatsPollingIntervalMs = 1000;
 static const int kPayloadTypeH264 = 122;
 static const int kPayloadTypeVP8 = 123;
 static const int kPayloadTypeVP9 = 124;
 
 class VideoAnalyzer : public PacketReceiver,
                       public Transport,
+                      public I420FrameCallback,
                       public VideoRenderer,
                       public VideoCaptureInput,
                       public EncodedFrameObserver {
  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)
       : input_(nullptr),
         transport_(transport),
         receiver_(nullptr),
         send_stream_(nullptr),
         test_label_(test_label),
         graph_data_output_file_(graph_data_output_file),
         graph_title_(graph_title),
         ssrc_to_analyze_(ssrc_to_analyze),
         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),
         last_render_time_(0),
         rtp_timestamp_delta_(0),
         avg_psnr_threshold_(avg_psnr_threshold),
         avg_ssim_threshold_(avg_ssim_threshold),
         stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
         comparison_available_event_(false, false),
         done_(true, false) {
     // Create thread pool for CPU-expensive PSNR/SSIM calculations.
 
     // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
@@ skipping 48 matching lines @@
   }
 
   void MeasuredEncodeTiming(int64_t ntp_time_ms, int encode_time_ms) {
     rtc::CritScope crit(&comparison_lock_);
     samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
   }
 
   void IncomingCapturedFrame(const VideoFrame& video_frame) override {
     VideoFrame copy = video_frame;
     copy.set_timestamp(copy.ntp_time_ms() * 90);
-
     {
       rtc::CritScope lock(&crit_);
-      if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
-        first_send_frame_ = copy;
-
       frames_.push_back(copy);
     }
 
     input_->IncomingCapturedFrame(video_frame);
   }
 
+  void FrameCallback(VideoFrame* video_frame) {
+    rtc::CritScope lock(&crit_);
+    if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0) {
+      while (frames_.front().timestamp() != video_frame->timestamp()) {
+        ++dropped_frames_before_first_encode_;
+        frames_.pop_front();
+        RTC_CHECK(!frames_.empty());
+      }
+      first_send_frame_ = *video_frame;
+    }
+  }
+
   bool SendRtp(const uint8_t* packet,
                size_t length,
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
 
     int64_t current_time =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     bool result = transport_->SendRtp(packet, length, options);
     {
       rtc::CritScope lock(&crit_);
       int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
 
-      if (rtp_timestamp_delta_ == 0) {
+      if (!first_send_frame_.IsZeroSize()) {
         rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp();
         first_send_frame_.Reset();
       }
       timestamp -= rtp_timestamp_delta_;
       send_times_[timestamp] = current_time;
       if (!transport_->DiscardedLastPacket() &&
           header.ssrc == ssrc_to_analyze_) {
         encoded_frame_sizes_[timestamp] +=
             length - (header.headerLength + header.paddingLength);
       }
@@ skipping 12 matching lines @@
   }
 
   void RenderFrame(const VideoFrame& video_frame,
                    int time_to_render_ms) override {
     int64_t render_time_ms =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     uint32_t send_timestamp = video_frame.timestamp() - rtp_timestamp_delta_;
 
     rtc::CritScope lock(&crit_);
 
-    while (frames_.front().timestamp() < send_timestamp) {
+    while (frames_.front().timestamp() != send_timestamp) {
+      if (last_rendered_frame_.IsZeroSize()) {
+        // No previous frame rendered, this one was dropped after sending but
+        // before rendering.
+        ++dropped_frames_before_rendering_;
+        frames_.pop_front();
+        RTC_CHECK(!frames_.empty());
+        continue;
+      }
       AddFrameComparison(frames_.front(), last_rendered_frame_, true,
                          render_time_ms);
       frames_.pop_front();
+      RTC_DCHECK(!frames_.empty());
     }
 
     VideoFrame reference_frame = frames_.front();
     frames_.pop_front();
     assert(!reference_frame.IsZeroSize());
     if (send_timestamp == reference_frame.timestamp() - 1) {
       // TODO(ivica): Make this work for > 2 streams.
       // Look at RTPSender::BuildRTPHeader.
       ++send_timestamp;
     }
@@ skipping 130 matching lines @@
 
    private:
     VideoAnalyzer* const parent_;
   };
 
   void AddFrameComparison(const VideoFrame& reference,
                           const VideoFrame& render,
                           bool dropped,
                           int64_t render_time_ms)
       EXCLUSIVE_LOCKS_REQUIRED(crit_) {
+    RTC_DCHECK(!render.IsZeroSize());
     int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
     int64_t send_time_ms = send_times_[reference_timestamp];
     send_times_.erase(reference_timestamp);
     int64_t recv_time_ms = recv_times_[reference_timestamp];
     recv_times_.erase(reference_timestamp);
 
     // TODO(ivica): Make this work for > 2 streams.
     auto it = encoded_frame_sizes_.find(reference_timestamp);
     if (it == encoded_frame_sizes_.end())
       it = encoded_frame_sizes_.find(reference_timestamp - 1);
@@ skipping 114 matching lines @@
     return frames_processed_ == frames_to_process_;
   }
 
   void PrintResults() {
     rtc::CritScope crit(&comparison_lock_);
     PrintResult("psnr", psnr_, " dB");
     PrintResult("ssim", ssim_, " score");
     PrintResult("sender_time", sender_time_, " ms");
     printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
            dropped_frames_);
+    printf("RESULT dropped_frames_before_first_encode: %s = %d frames\n",
+           test_label_.c_str(), dropped_frames_before_first_encode_);
+    printf("RESULT dropped_frames_before_rendering: %s = %d frames\n",
+           test_label_.c_str(), dropped_frames_before_rendering_);

[stefan-webrtc, 2016/03/14 13:14:03]

It's a bit weird that we mix PrintResult with printfs. Could you please align
(either way)?

[pbos-webrtc, 2016/03/14 15:12:58]

Done.

     PrintResult("receiver_time", receiver_time_, " ms");
     PrintResult("total_delay_incl_network", end_to_end_, " ms");
     PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
     PrintResult("encoded_frame_size", encoded_frame_size_, " bytes");
     PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
     PrintResult("encode_time", encode_time_ms, " ms");
     PrintResult("encode_usage_percent", encode_usage_percent, " percent");
     PrintResult("media_bitrate", media_bitrate_bps, " bps");
 
     EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
@@ skipping 103 matching lines @@
   test::Statistics encoded_frame_size_ GUARDED_BY(comparison_lock_);
   test::Statistics encode_frame_rate_ GUARDED_BY(comparison_lock_);
   test::Statistics encode_time_ms GUARDED_BY(comparison_lock_);
   test::Statistics encode_usage_percent GUARDED_BY(comparison_lock_);
   test::Statistics media_bitrate_bps GUARDED_BY(comparison_lock_);
 
   const int frames_to_process_;
   int frames_recorded_;
   int frames_processed_;
   int dropped_frames_;
+  int dropped_frames_before_first_encode_;
+  int dropped_frames_before_rendering_;
   int64_t last_render_time_;
   uint32_t rtp_timestamp_delta_;
 
   rtc::CriticalSection crit_;
   std::deque<VideoFrame> frames_ GUARDED_BY(crit_);
   VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
   rtc::TimestampWrapAroundHandler wrap_handler_ GUARDED_BY(crit_);
   std::map<int64_t, int64_t> send_times_ GUARDED_BY(crit_);
   std::map<int64_t, int64_t> recv_times_ GUARDED_BY(crit_);
   std::map<int64_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
@@ skipping 373 matching lines @@
       params_.analyzer.test_durations_secs * params_.common.fps,
       graph_data_output_file, graph_title,
       kVideoSendSsrcs[params_.ss.selected_stream]);
 
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupCommon(&analyzer, &recv_transport);
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
+  video_send_config_.pre_encode_callback = &analyzer;
   for (auto& config : video_receive_configs_)
     config.pre_decode_callback = &analyzer;
   RTC_DCHECK(!video_send_config_.post_encode_callback);
   video_send_config_.post_encode_callback = analyzer.encode_timing_proxy();
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
   CreateVideoStreams();
   analyzer.input_ = video_send_stream_->Input();
@@ skipping 80 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc