Delete VideoReceiveStream::Config::pre_render_callback.

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 177 matching lines @@
 
   DestroyStreams();
 }
 
 TEST_P(EndToEndTest, RendersSingleDelayedFrame) {
   static const int kWidth = 320;
   static const int kHeight = 240;
   // This constant is chosen to be higher than the timeout in the video_render
   // module. This makes sure that frames aren't dropped if there are no other
   // frames in the queue.
-  static const int kDelayRenderCallbackMs = 1000;
+  static const int kRenderDelayMs = 1000;
 
   class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
    public:
     Renderer() : event_(false, false) {}
 
-    void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
+    void OnFrame(const VideoFrame& video_frame) override {
+      SleepMs(kRenderDelayMs);
+      event_.Set();
+    }
 
     bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
 
     rtc::Event event_;
   } renderer;
 
-  class TestFrameCallback : public I420FrameCallback {
-   public:
-    TestFrameCallback() : event_(false, false) {}
-
-    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
-
-   private:
-    void FrameCallback(VideoFrame* frame) override {
-      SleepMs(kDelayRenderCallbackMs);
-      event_.Set();
-    }
-
-    rtc::Event event_;
-  };
-
   CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
 
   test::DirectTransport sender_transport(sender_call_.get());
   test::DirectTransport receiver_transport(receiver_call_.get());
   sender_transport.SetReceiver(receiver_call_->Receiver());
   receiver_transport.SetReceiver(sender_call_->Receiver());
 
   CreateSendConfig(1, 0, 0, &sender_transport);
   CreateMatchingReceiveConfigs(&receiver_transport);
 
-  TestFrameCallback pre_render_callback;
-  video_receive_configs_[0].pre_render_callback = &pre_render_callback;
   video_receive_configs_[0].renderer = &renderer;
 
   CreateVideoStreams();
   Start();
 
   // Create frames that are smaller than the send width/height, this is done to
   // check that the callbacks are done after processing video.
   std::unique_ptr<test::FrameGenerator> frame_generator(
       test::FrameGenerator::CreateChromaGenerator(kWidth, kHeight));
   test::FrameForwarder frame_forwarder;
   video_send_stream_->SetSource(
       &frame_forwarder, VideoSendStream::DegradationPreference::kBalanced);
 
   frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
-  EXPECT_TRUE(pre_render_callback.Wait())
-      << "Timed out while waiting for pre-render callback.";
   EXPECT_TRUE(renderer.Wait())
       << "Timed out while waiting for the frame to render.";
 
   Stop();
 
   sender_transport.StopSending();
   receiver_transport.StopSending();
 
   DestroyStreams();
 }
@@ skipping 779 matching lines @@
   } test;
 
   RunBaseTest(&test);
 }
 
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
 void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) {
   static const int kDroppedFrameNumber = 10;
   class RetransmissionObserver : public test::EndToEndTest,
-                                 public I420FrameCallback {
+                                 public rtc::VideoSinkInterface<VideoFrame> {
    public:
     RetransmissionObserver(bool enable_rtx, bool enable_red)
         : EndToEndTest(kDefaultTimeoutMs),
           payload_type_(GetPayloadType(false, enable_red)),
           retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
                                           : kVideoSendSsrcs[0]),
           retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
           encoder_(VP8Encoder::Create()),
           marker_bits_observed_(0),
           retransmitted_timestamp_(0) {}
@@ skipping 35 matching lines @@
           // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
           // details.
           observation_complete_.Set();
         }
         return DROP_PACKET;
       }
 
       return SEND_PACKET;
     }
 
-    void FrameCallback(VideoFrame* frame) override {
+    void OnFrame(const VideoFrame& frame) override {
       rtc::CritScope lock(&crit_);
-      if (frame->timestamp() == retransmitted_timestamp_)
+      if (frame.timestamp() == retransmitted_timestamp_)
         observation_complete_.Set();
-      rendered_timestamps_.push_back(frame->timestamp());
+      rendered_timestamps_.push_back(frame.timestamp());
+      orig_renderer_->OnFrame(frame);
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
-      (*receive_configs)[0].pre_render_callback = this;
+
+      // Insert ourselves into the rendering pipeline.
+      orig_renderer_ = (*receive_configs)[0].renderer;

[tommi, 2017/02/07 11:58:20]

first DCHECK_EQ(nullptr, orig_renderer_);

after assignment, it would also be nice to:

DCHECK_NE(nullptr, orig_renderer_);

[nisse-webrtc, 2017/02/07 13:30:59]

Added DCHECK. DCHECK_NE and DCHECK_EQ doesn't work with nullptr, the compiler
complains "use of overloaded operator '<<' is ambiguous".

+      (*receive_configs)[0].renderer = this;
+
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
 
       if (payload_type_ == kRedPayloadType) {
         send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
         send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
         if (retransmission_ssrc_ == kSendRtxSsrcs[0])
           send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
         (*receive_configs)[0].rtp.ulpfec.ulpfec_payload_type =
             send_config->rtp.ulpfec.ulpfec_payload_type;
         (*receive_configs)[0].rtp.ulpfec.red_payload_type =
@@ skipping 27 matching lines @@
         if (use_rtx)
           return kRtxRedPayloadType;
         return kRedPayloadType;
       }
       if (use_rtx)
         return kSendRtxPayloadType;
       return kFakeVideoSendPayloadType;
     }
 
     rtc::CriticalSection crit_;
+    rtc::VideoSinkInterface<VideoFrame>* orig_renderer_;

[tommi, 2017/02/07 11:58:20]

initialize to nullptr

[nisse-webrtc, 2017/02/07 13:30:58]

Done.

     const int payload_type_;
     const uint32_t retransmission_ssrc_;
     const int retransmission_payload_type_;
     std::unique_ptr<VideoEncoder> encoder_;
     const std::string payload_name_;
     int marker_bits_observed_;
     uint32_t retransmitted_timestamp_ GUARDED_BY(&crit_);
     std::vector<uint32_t> rendered_timestamps_ GUARDED_BY(&crit_);
   } test(enable_rtx, enable_red);
 
@@ skipping 2993 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