| /* |
| * 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 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 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 |