Index: webrtc/base/bandwidthsmoother.cc |
diff --git a/webrtc/base/bandwidthsmoother.cc b/webrtc/base/bandwidthsmoother.cc |
deleted file mode 100644 |
index d48c12e6c65428251593339e432792a097009f67..0000000000000000000000000000000000000000 |
--- a/webrtc/base/bandwidthsmoother.cc |
+++ /dev/null |
@@ -1,86 +0,0 @@ |
-/* |
- * 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 |