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diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
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+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_CALL_RAMPUP_TESTS_H_
+#define WEBRTC_CALL_RAMPUP_TESTS_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/event.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call.h"
+#include "webrtc/test/call_test.h"
+
+namespace webrtc {
+
+static const int kTransmissionTimeOffsetExtensionId = 6;
+static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
+static const unsigned int kSingleStreamTargetBps = 1000000;
+
+class Clock;
+
+class RampUpTester : public test::EndToEndTest {
+ public:
+  RampUpTester(size_t num_video_streams,
+               size_t num_audio_streams,
+               unsigned int start_bitrate_bps,
+               const std::string& extension_type,
+               bool rtx,
+               bool red);
+  ~RampUpTester() override;
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+
+  void PerformTest() override;
+
+ protected:
+  virtual bool PollStats();
+
+  void AccumulateStats(const VideoSendStream::StreamStats& stream,
+                       size_t* total_packets_sent,
+                       size_t* total_sent,
+                       size_t* padding_sent,
+                       size_t* media_sent) const;
+
+  void ReportResult(const std::string& measurement,
+                    size_t value,
+                    const std::string& units) const;
+  void TriggerTestDone();
+
+  rtc::Event event_;
+  Clock* const clock_;
+  FakeNetworkPipe::Config forward_transport_config_;
+  const size_t num_video_streams_;
+  const size_t num_audio_streams_;
+  const bool rtx_;
+  const bool red_;
+  VideoSendStream* send_stream_;
+  test::PacketTransport* send_transport_;
+
+ private:
+  typedef std::map<uint32_t, uint32_t> SsrcMap;
+
+  Call::Config GetSenderCallConfig() override;
+  void OnVideoStreamsCreated(
+      VideoSendStream* send_stream,
+      const std::vector<VideoReceiveStream*>& receive_streams) override;
+  test::PacketTransport* CreateSendTransport(Call* sender_call) override;
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override;
+  void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStream::Config>* receive_configs) override;
+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
+
+  static bool BitrateStatsPollingThread(void* obj);
+
+  const int start_bitrate_bps_;
+  bool start_bitrate_verified_;
+  int expected_bitrate_bps_;
+  int64_t test_start_ms_;
+  int64_t ramp_up_finished_ms_;
+
+  const std::string extension_type_;
+  std::vector<uint32_t> video_ssrcs_;
+  std::vector<uint32_t> video_rtx_ssrcs_;
+  std::vector<uint32_t> audio_ssrcs_;
+  SsrcMap rtx_ssrc_map_;
+
+  rtc::PlatformThread poller_thread_;
+  Call* sender_call_;
+};
+
+class RampUpDownUpTester : public RampUpTester {
+ public:
+  RampUpDownUpTester(size_t num_video_streams,
+                     size_t num_audio_streams,
+                     unsigned int start_bitrate_bps,
+                     const std::string& extension_type,
+                     bool rtx,
+                     bool red);
+  ~RampUpDownUpTester() override;
+
+ protected:
+  bool PollStats() override;
+
+ private:
+  static const int kHighBandwidthLimitBps = 80000;
+  static const int kExpectedHighBitrateBps = 60000;
+  static const int kLowBandwidthLimitBps = 20000;
+  static const int kExpectedLowBitrateBps = 20000;
+  enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
+
+  Call::Config GetReceiverCallConfig() override;
+
+  std::string GetModifierString() const;
+  void EvolveTestState(int bitrate_bps, bool suspended);
+
+  TestStates test_state_;
+  int64_t state_start_ms_;
+  int64_t interval_start_ms_;
+  int sent_bytes_;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_CALL_RAMPUP_TESTS_H_