1 /* 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_ 12 #define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 17 #include <memory> 18 #include <string> 19 #include <vector> 20 21 #include "api/transport/field_trial_based_config.h" 22 #include "api/transport/network_types.h" 23 #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h" 24 #include "modules/congestion_controller/goog_cc/delay_based_bwe.h" 25 #include "rtc_base/constructor_magic.h" 26 #include "system_wrappers/include/clock.h" 27 #include "test/field_trial.h" 28 #include "test/gtest.h" 29 30 namespace webrtc { 31 namespace test { 32 33 class TestBitrateObserver { 34 public: TestBitrateObserver()35 TestBitrateObserver() : updated_(false), latest_bitrate_(0) {} ~TestBitrateObserver()36 ~TestBitrateObserver() {} 37 38 void OnReceiveBitrateChanged(uint32_t bitrate); 39 Reset()40 void Reset() { updated_ = false; } 41 updated()42 bool updated() const { return updated_; } 43 latest_bitrate()44 uint32_t latest_bitrate() const { return latest_bitrate_; } 45 46 private: 47 bool updated_; 48 uint32_t latest_bitrate_; 49 }; 50 51 class RtpStream { 52 public: 53 enum { kSendSideOffsetUs = 1000000 }; 54 55 RtpStream(int fps, int bitrate_bps); 56 57 // Generates a new frame for this stream. If called too soon after the 58 // previous frame, no frame will be generated. The frame is split into 59 // packets. 60 int64_t GenerateFrame(int64_t time_now_us, 61 std::vector<PacketResult>* packets); 62 63 // The send-side time when the next frame can be generated. 64 int64_t next_rtp_time() const; 65 66 void set_bitrate_bps(int bitrate_bps); 67 68 int bitrate_bps() const; 69 70 static bool Compare(const std::unique_ptr<RtpStream>& lhs, 71 const std::unique_ptr<RtpStream>& rhs); 72 73 private: 74 int fps_; 75 int bitrate_bps_; 76 int64_t next_rtp_time_; 77 78 RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream); 79 }; 80 81 class StreamGenerator { 82 public: 83 StreamGenerator(int capacity, int64_t time_now); 84 ~StreamGenerator(); 85 86 // Add a new stream. 87 void AddStream(RtpStream* stream); 88 89 // Set the link capacity. 90 void set_capacity_bps(int capacity_bps); 91 92 // Divides |bitrate_bps| among all streams. The allocated bitrate per stream 93 // is decided by the initial allocation ratios. 94 void SetBitrateBps(int bitrate_bps); 95 96 // Set the RTP timestamp offset for the stream identified by |ssrc|. 97 void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset); 98 99 // TODO(holmer): Break out the channel simulation part from this class to make 100 // it possible to simulate different types of channels. 101 int64_t GenerateFrame(std::vector<PacketResult>* packets, 102 int64_t time_now_us); 103 104 private: 105 // Capacity of the simulated channel in bits per second. 106 int capacity_; 107 // The time when the last packet arrived. 108 int64_t prev_arrival_time_us_; 109 // All streams being transmitted on this simulated channel. 110 std::vector<std::unique_ptr<RtpStream>> streams_; 111 112 RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator); 113 }; 114 } // namespace test 115 116 class DelayBasedBweTest : public ::testing::Test { 117 public: 118 DelayBasedBweTest(); 119 explicit DelayBasedBweTest(const std::string& field_trial_string); 120 ~DelayBasedBweTest() override; 121 122 protected: 123 void AddDefaultStream(); 124 125 // Helpers to insert a single packet into the delay-based BWE. 126 void IncomingFeedback(int64_t arrival_time_ms, 127 int64_t send_time_ms, 128 size_t payload_size); 129 void IncomingFeedback(int64_t arrival_time_ms, 130 int64_t send_time_ms, 131 size_t payload_size, 132 const PacedPacketInfo& pacing_info); 133 134 // Generates a frame of packets belonging to a stream at a given bitrate and 135 // with a given ssrc. The stream is pushed through a very simple simulated 136 // network, and is then given to the receive-side bandwidth estimator. 137 // Returns true if an over-use was seen, false otherwise. 138 // The StreamGenerator::updated() should be used to check for any changes in 139 // target bitrate after the call to this function. 140 bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps); 141 142 // Run the bandwidth estimator with a stream of |number_of_frames| frames, or 143 // until it reaches |target_bitrate|. 144 // Can for instance be used to run the estimator for some time to get it 145 // into a steady state. 146 uint32_t SteadyStateRun(uint32_t ssrc, 147 int number_of_frames, 148 uint32_t start_bitrate, 149 uint32_t min_bitrate, 150 uint32_t max_bitrate, 151 uint32_t target_bitrate); 152 153 void TestTimestampGroupingTestHelper(); 154 155 void TestWrappingHelper(int silence_time_s); 156 157 void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate); 158 void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate); 159 void RateIncreaseRtpTimestampsTestHelper(int expected_iterations); 160 void CapacityDropTestHelper(int number_of_streams, 161 bool wrap_time_stamp, 162 uint32_t expected_bitrate_drop_delta, 163 int64_t receiver_clock_offset_change_ms); 164 165 static const uint32_t kDefaultSsrc; 166 FieldTrialBasedConfig field_trial_config_; 167 168 std::unique_ptr<test::ScopedFieldTrials> 169 field_trial; // Must be initialized first. 170 SimulatedClock clock_; // Time at the receiver. 171 test::TestBitrateObserver bitrate_observer_; 172 std::unique_ptr<AcknowledgedBitrateEstimatorInterface> 173 acknowledged_bitrate_estimator_; 174 const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_; 175 std::unique_ptr<DelayBasedBwe> bitrate_estimator_; 176 std::unique_ptr<test::StreamGenerator> stream_generator_; 177 int64_t arrival_time_offset_ms_; 178 bool first_update_; 179 180 RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest); 181 }; 182 } // namespace webrtc 183 184 #endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_ 185