BitrateProber: Support higher probing bitrates

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 1026 matching lines @@
     Clock* const clock_;
     std::unique_ptr<internal::TransportAdapter> transport_adapter_;
     rtc::CriticalSection crit_;
     int64_t last_packet_time_ms_ GUARDED_BY(crit_);
     test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
+TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
+  class PaddingIsPrimarilyRetransmissions : public test::EndToEndTest {
+   public:
+    PaddingIsPrimarilyRetransmissions()
+        : EndToEndTest(kDefaultTimeoutMs),
+          clock_(Clock::GetRealTimeClock()),
+          padding_length_(0),
+          total_length_(0) {}
+
+   private:
+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
+      rtc::CritScope lock(&crit_);
+
+      RTPHeader header;
+      parser_->Parse(packet, length, &header);
+      padding_length_ += header.paddingLength;
+      total_length_ += length;
+
+      return SEND_PACKET;
+    }
+
+    test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+      const int kNetworkDelayMs = 50;
+      FakeNetworkPipe::Config config;
+      config.loss_percent = 10;
+      config.link_capacity_kbps = 10000;  // 10 Mbps
+      config.queue_delay_ms = kNetworkDelayMs;
+      return new test::PacketTransport(sender_call, this,
+                                       test::PacketTransport::kSender, config);
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStream::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                       test::kTransportSequenceNumberExtensionId));
+      (*receive_configs)[0].rtp.extensions.clear();
+      (*receive_configs)[0].rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                       test::kTransportSequenceNumberExtensionId));
+      (*receive_configs)[0].rtp.transport_cc = true;
+    }
+
+    void PerformTest() override {
+      SleepMs(5000);

[stefan-webrtc, 2016/09/27 14:18:33]

Should you instead run the test until Call::Stats::send_bandwidth_bps > 8000
kbps or something?

+      // Expect padding to not dominate the data sent while probing for BW.
+      EXPECT_LT(padding_length_, .5 * total_length_);
+    }
+
+    rtc::CriticalSection crit_;
+    Clock* const clock_;
+    size_t padding_length_;
+    size_t total_length_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
 // This test first observes "high" bitrate use at which point it sends a REMB to
 // indicate that it should be lowered significantly. The test then observes that
 // the bitrate observed is sinking well below the min-transmit-bitrate threshold
 // to verify that the min-transmit bitrate respects incoming REMB.
 //
 // Note that the test starts at "high" bitrate and does not ramp up to "higher"
 // bitrate since no receiver block or remb is sent in the initial phase.
 TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
   static const int kMinTransmitBitrateBps = 400000;
   static const int kHighBitrateBps = 150000;
@@ skipping 1513 matching lines @@
         observation_complete_.Set();
       }
     }
   } test;
 
   RunBaseTest(&test);
 }
 #endif  // !defined(RTC_DISABLE_VP9)
 
 }  // namespace webrtc