/* * 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_DELAY_BASED_BWE_UNITTEST_HELPER_H_ #define MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_ #include #include #include #include #include #include "modules/congestion_controller/acknowledged_bitrate_estimator.h" #include "modules/congestion_controller/delay_based_bwe.h" #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "rtc_base/constructormagic.h" #include "system_wrappers/include/clock.h" #include "test/gtest.h" namespace webrtc { namespace test { class TestBitrateObserver : public RemoteBitrateObserver { public: TestBitrateObserver() : updated_(false), latest_bitrate_(0) {} virtual ~TestBitrateObserver() {} void OnReceiveBitrateChanged(const std::vector& 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* 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& lhs, const std::unique_ptr& 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* 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> 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, 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; SimulatedClock clock_; // Time at the receiver. test::TestBitrateObserver bitrate_observer_; std::unique_ptr acknowledged_bitrate_estimator_; std::unique_ptr bitrate_estimator_; std::unique_ptr stream_generator_; int64_t arrival_time_offset_ms_; bool first_update_; RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest); }; } // namespace webrtc #endif // MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_