summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
blob: 634e6a4d82ff646cc4a92b9b31a681ed228173a0 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
/*
 *  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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_

#include <stddef.h>
#include <stdint.h>

#include <memory>
#include <vector>

#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_types.h"
#include "api/units/timestamp.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"

namespace webrtc {
namespace test {

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

  void OnReceiveBitrateChanged(uint32_t bitrate);

  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);

  RtpStream(const RtpStream&) = delete;
  RtpStream& operator=(const RtpStream&) = delete;

  // 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,
                        int64_t* next_sequence_number,
                        std::vector<PacketResult>* 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_;
};

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

  StreamGenerator(const StreamGenerator&) = delete;
  StreamGenerator& operator=(const StreamGenerator&) = delete;

  // 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(int64_t time_now_us,
                        int64_t* next_sequence_number,
                        std::vector<PacketResult>* packets);

 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_;
};
}  // namespace test

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

 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,
                        size_t payload_size);
  void IncomingFeedback(int64_t arrival_time_ms,
                        int64_t send_time_ms,
                        size_t payload_size,
                        const PacedPacketInfo& pacing_info);
  void IncomingFeedback(Timestamp receive_time,
                        Timestamp send_time,
                        size_t payload_size,
                        const PacedPacketInfo& pacing_info);

  // 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;
  FieldTrialBasedConfig field_trial_config_;

  std::unique_ptr<test::ScopedFieldTrials>
      field_trial;        // Must be initialized first.
  SimulatedClock clock_;  // Time at the receiver.
  test::TestBitrateObserver bitrate_observer_;
  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
      acknowledged_bitrate_estimator_;
  const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
  std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
  std::unique_ptr<test::StreamGenerator> stream_generator_;
  int64_t arrival_time_offset_ms_;
  int64_t next_sequence_number_;
  bool first_update_;
};

}  // namespace webrtc

#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_