webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h

/*
 *  Copyright (c) 2016 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_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_

#include <list>
#include <map>
#include <memory>
#include <utility>
#include <vector>

#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/clock.h"

namespace webrtc {
namespace test {

class TestBitrateObserver : public RemoteBitrateObserver {
 public:
  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
  virtual ~TestBitrateObserver() {}

  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
                               uint32_t bitrate) override;

  void Reset() { updated_ = false; }

  bool updated() const { return updated_; }

  uint32_t latest_bitrate() const { return latest_bitrate_; }

 private:
  bool updated_;
  uint32_t latest_bitrate_;
};

class RtpStream {
 public:
  enum { kSendSideOffsetUs = 1000000 };

  RtpStream(int fps, int bitrate_bps);

  // Generates a new frame for this stream. If called too soon after the
  // previous frame, no frame will be generated. The frame is split into
  // packets.
  int64_t GenerateFrame(int64_t time_now_us, std::vector<PacketInfo>* packets);

  // The send-side time when the next frame can be generated.
  int64_t next_rtp_time() const;

  void set_bitrate_bps(int bitrate_bps);

  int bitrate_bps() const;

  static bool Compare(const std::unique_ptr<RtpStream>& lhs,
                      const std::unique_ptr<RtpStream>& rhs);

 private:
  int fps_;
  int bitrate_bps_;
  int64_t next_rtp_time_;
  uint16_t sequence_number_;

  RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream);
};

class StreamGenerator {
 public:
  StreamGenerator(int capacity, int64_t time_now);

  // Add a new stream.
  void AddStream(RtpStream* stream);

  // Set the link capacity.
  void set_capacity_bps(int capacity_bps);

  // Divides |bitrate_bps| among all streams. The allocated bitrate per stream
  // is decided by the initial allocation ratios.
  void SetBitrateBps(int bitrate_bps);

  // Set the RTP timestamp offset for the stream identified by |ssrc|.
  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);

  // TODO(holmer): Break out the channel simulation part from this class to make
  // it possible to simulate different types of channels.
  int64_t GenerateFrame(std::vector<PacketInfo>* packets, int64_t time_now_us);

 private:
  // Capacity of the simulated channel in bits per second.
  int capacity_;
  // The time when the last packet arrived.
  int64_t prev_arrival_time_us_;
  // All streams being transmitted on this simulated channel.
  std::vector<std::unique_ptr<RtpStream>> streams_;

  RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
};
}  // namespace test

class DelayBasedBweTest : public ::testing::Test {
 public:
  DelayBasedBweTest();
  virtual ~DelayBasedBweTest();

 protected:
  void AddDefaultStream();

  // Helpers to insert a single packet into the delay-based BWE.
  void IncomingFeedback(int64_t arrival_time_ms,
                        int64_t send_time_ms,
                        uint16_t sequence_number,
                        size_t payload_size);
  void IncomingFeedback(int64_t arrival_time_ms,
                        int64_t send_time_ms,
                        uint16_t sequence_number,
                        size_t payload_size,
                        int probe_cluster_id);

  // Generates a frame of packets belonging to a stream at a given bitrate and
  // with a given ssrc. The stream is pushed through a very simple simulated
  // network, and is then given to the receive-side bandwidth estimator.
  // Returns true if an over-use was seen, false otherwise.
  // The StreamGenerator::updated() should be used to check for any changes in
  // target bitrate after the call to this function.
  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);

  // Run the bandwidth estimator with a stream of |number_of_frames| frames, or
  // until it reaches |target_bitrate|.
  // Can for instance be used to run the estimator for some time to get it
  // into a steady state.
  uint32_t SteadyStateRun(uint32_t ssrc,
                          int number_of_frames,
                          uint32_t start_bitrate,
                          uint32_t min_bitrate,
                          uint32_t max_bitrate,
                          uint32_t target_bitrate);

  void TestTimestampGroupingTestHelper();

  void TestWrappingHelper(int silence_time_s);

  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
  void CapacityDropTestHelper(int number_of_streams,
                              bool wrap_time_stamp,
                              uint32_t expected_bitrate_drop_delta,
                              int64_t receiver_clock_offset_change_ms);

  static const uint32_t kDefaultSsrc;

  SimulatedClock clock_;  // Time at the receiver.
  std::unique_ptr<test::TestBitrateObserver> bitrate_observer_;
  std::unique_ptr<RemoteBitrateEstimator> bitrate_estimator_;
  std::unique_ptr<test::StreamGenerator> stream_generator_;
  int64_t arrival_time_offset_ms_;

  RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest);
};
}  // namespace webrtc

#endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_