Periodically update codec bit/frame rate settings.

webrtc/video/video_send_stream_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>  // max
@@ skipping 913 matching lines @@
             ++current_size_frame_;
           }
         }
       }
 
       return SEND_PACKET;
     }
 
     void TriggerLossReport(const RTPHeader& header) {
       // Send lossy receive reports to trigger FEC enabling.
-      if (packet_count_++ % 2 != 0) {
-        // Receive statistics reporting having lost 50% of the packets.
+      const int kLossPercent = 5;
+      if (packet_count_++ % (100 / kLossPercent) != 0) {
         FakeReceiveStatistics lossy_receive_stats(
-            kVideoSendSsrcs[0], header.sequenceNumber, packet_count_ / 2, 127);
+            kVideoSendSsrcs[0], header.sequenceNumber,
+            (packet_count_ * (100 - kLossPercent)) / 100,  // Cumulative lost.
+            static_cast<uint8_t>((255 * kLossPercent) / 100));  // Loss percent.
         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
                                &lossy_receive_stats, nullptr, nullptr,
                                transport_adapter_.get());
 
         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
 
         RTCPSender::FeedbackState feedback_state;
 
         EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
@@ skipping 32 matching lines @@
       if (!test_generic_packetization_)
         send_config->encoder_settings.payload_name = "VP8";
 
       send_config->encoder_settings.encoder = &encoder_;
       send_config->rtp.max_packet_size = kMaxPacketSize;
       send_config->post_encode_callback = this;
 
       // Make sure there is at least one extension header, to make the RTP
       // header larger than the base length of 12 bytes.
       EXPECT_FALSE(send_config->rtp.extensions.empty());
+
+      // Setup screen content disables frame dropping which makes this easier.

[stefan-webrtc, 2017/05/23 17:03:20]

This seems a bit unfortunate. What if we want to reenable frame dropping for
screen content?

[sprang_webrtc, 2017/05/24 09:07:07]

Then we should probably expose some API surface to enable disabling of frame
dropping, but that touches a lot of layers....

+      class VideoStreamFactory
+          : public VideoEncoderConfig::VideoStreamFactoryInterface {
+       public:
+        explicit VideoStreamFactory(size_t num_temporal_layers)
+            : num_temporal_layers_(num_temporal_layers) {
+          EXPECT_GT(num_temporal_layers, 0u);
+        }
+
+       private:
+        std::vector<VideoStream> CreateEncoderStreams(
+            int width,
+            int height,
+            const VideoEncoderConfig& encoder_config) override {
+          std::vector<VideoStream> streams =
+              test::CreateVideoStreams(width, height, encoder_config);
+          for (VideoStream& stream : streams) {
+            stream.temporal_layer_thresholds_bps.resize(num_temporal_layers_ -
+                                                        1);
+          }
+          return streams;
+        }
+        const size_t num_temporal_layers_;
+      };
+
+      encoder_config->video_stream_factory =
+          new rtc::RefCountedObject<VideoStreamFactory>(2);
+      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
     }
 
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
     }
 
     std::unique_ptr<internal::TransportAdapter> transport_adapter_;
     test::ConfigurableFrameSizeEncoder encoder_;
 
     const size_t max_packet_size_;
@@ skipping 2331 matching lines @@
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc