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      1 /*
      2  *  Copyright 2011 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 #include "webrtc/base/bandwidthsmoother.h"
     12 
     13 #include <limits.h>
     14 #include <algorithm>
     15 
     16 namespace rtc {
     17 
     18 BandwidthSmoother::BandwidthSmoother(int initial_bandwidth_guess,
     19                                      uint32_t time_between_increase,
     20                                      double percent_increase,
     21                                      size_t samples_count_to_average,
     22                                      double min_sample_count_percent)
     23     : time_between_increase_(time_between_increase),
     24       percent_increase_(std::max(1.0, percent_increase)),
     25       time_at_last_change_(0),
     26       bandwidth_estimation_(initial_bandwidth_guess),
     27       accumulator_(samples_count_to_average),
     28       min_sample_count_percent_(
     29           std::min(1.0, std::max(0.0, min_sample_count_percent))) {
     30 }
     31 
     32 BandwidthSmoother::~BandwidthSmoother() = default;
     33 
     34 // Samples a new bandwidth measurement
     35 // returns true if the bandwidth estimation changed
     36 bool BandwidthSmoother::Sample(uint32_t sample_time, int bandwidth) {
     37   if (bandwidth < 0) {
     38     return false;
     39   }
     40 
     41   accumulator_.AddSample(bandwidth);
     42 
     43   if (accumulator_.count() < static_cast<size_t>(
     44           accumulator_.max_count() * min_sample_count_percent_)) {
     45     // We have not collected enough samples yet.
     46     return false;
     47   }
     48 
     49   // Replace bandwidth with the mean of sampled bandwidths.
     50   const int mean_bandwidth = static_cast<int>(accumulator_.ComputeMean());
     51 
     52   if (mean_bandwidth < bandwidth_estimation_) {
     53     time_at_last_change_ = sample_time;
     54     bandwidth_estimation_ = mean_bandwidth;
     55     return true;
     56   }
     57 
     58   const int old_bandwidth_estimation = bandwidth_estimation_;
     59   const double increase_threshold_d = percent_increase_ * bandwidth_estimation_;
     60   if (increase_threshold_d > INT_MAX) {
     61     // If bandwidth goes any higher we would overflow.
     62     return false;
     63   }
     64 
     65   const int increase_threshold = static_cast<int>(increase_threshold_d);
     66   if (mean_bandwidth < increase_threshold) {
     67     time_at_last_change_ = sample_time;
     68     // The value of bandwidth_estimation remains the same if we don't exceed
     69     // percent_increase_ * bandwidth_estimation_ for at least
     70     // time_between_increase_ time.
     71   } else if (sample_time >= time_at_last_change_ + time_between_increase_) {
     72     time_at_last_change_ = sample_time;
     73     if (increase_threshold == 0) {
     74       // Bandwidth_estimation_ must be zero. Assume a jump from zero to a
     75       // positive bandwidth means we have regained connectivity.
     76       bandwidth_estimation_ = mean_bandwidth;
     77     } else {
     78       bandwidth_estimation_ = increase_threshold;
     79     }
     80   }
     81   // Else don't make a change.
     82 
     83   return old_bandwidth_estimation != bandwidth_estimation_;
     84 }
     85 
     86 }  // namespace rtc
     87