/* | |

* Copyright 2011 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. | |

*/ | |

#include "webrtc/base/bandwidthsmoother.h" | |

#include <limits.h> | |

#include <algorithm> | |

namespace rtc { | |

BandwidthSmoother::BandwidthSmoother(int initial_bandwidth_guess, | |

uint32_t time_between_increase, | |

double percent_increase, | |

size_t samples_count_to_average, | |

double min_sample_count_percent) | |

: time_between_increase_(time_between_increase), | |

percent_increase_(std::max(1.0, percent_increase)), | |

time_at_last_change_(0), | |

bandwidth_estimation_(initial_bandwidth_guess), | |

accumulator_(samples_count_to_average), | |

min_sample_count_percent_( | |

std::min(1.0, std::max(0.0, min_sample_count_percent))) { | |

} | |

BandwidthSmoother::~BandwidthSmoother() = default; | |

// Samples a new bandwidth measurement | |

// returns true if the bandwidth estimation changed | |

bool BandwidthSmoother::Sample(uint32_t sample_time, int bandwidth) { | |

if (bandwidth < 0) { | |

return false; | |

} | |

accumulator_.AddSample(bandwidth); | |

if (accumulator_.count() < static_cast<size_t>( | |

accumulator_.max_count() * min_sample_count_percent_)) { | |

// We have not collected enough samples yet. | |

return false; | |

} | |

// Replace bandwidth with the mean of sampled bandwidths. | |

const int mean_bandwidth = static_cast<int>(accumulator_.ComputeMean()); | |

if (mean_bandwidth < bandwidth_estimation_) { | |

time_at_last_change_ = sample_time; | |

bandwidth_estimation_ = mean_bandwidth; | |

return true; | |

} | |

const int old_bandwidth_estimation = bandwidth_estimation_; | |

const double increase_threshold_d = percent_increase_ * bandwidth_estimation_; | |

if (increase_threshold_d > INT_MAX) { | |

// If bandwidth goes any higher we would overflow. | |

return false; | |

} | |

const int increase_threshold = static_cast<int>(increase_threshold_d); | |

if (mean_bandwidth < increase_threshold) { | |

time_at_last_change_ = sample_time; | |

// The value of bandwidth_estimation remains the same if we don't exceed | |

// percent_increase_ * bandwidth_estimation_ for at least | |

// time_between_increase_ time. | |

} else if (sample_time >= time_at_last_change_ + time_between_increase_) { | |

time_at_last_change_ = sample_time; | |

if (increase_threshold == 0) { | |

// Bandwidth_estimation_ must be zero. Assume a jump from zero to a | |

// positive bandwidth means we have regained connectivity. | |

bandwidth_estimation_ = mean_bandwidth; | |

} else { | |

bandwidth_estimation_ = increase_threshold; | |

} | |

} | |

// Else don't make a change. | |

return old_bandwidth_estimation != bandwidth_estimation_; | |

} | |

} // namespace rtc |