| Index: webrtc/base/bandwidthsmoother.cc
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| diff --git a/webrtc/base/bandwidthsmoother.cc b/webrtc/base/bandwidthsmoother.cc
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| deleted file mode 100644
|
| index d48c12e6c65428251593339e432792a097009f67..0000000000000000000000000000000000000000
|
| --- a/webrtc/base/bandwidthsmoother.cc
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| +++ /dev/null
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| @@ -1,86 +0,0 @@
|
| -/*
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| - * Copyright 2011 The WebRTC Project Authors. All rights reserved.
|
| - *
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| - * 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
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| - * 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"
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| -
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| -#include <limits.h>
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| -#include <algorithm>
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| -
|
| -namespace rtc {
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| -
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| -BandwidthSmoother::BandwidthSmoother(int initial_bandwidth_guess,
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| - uint32_t time_between_increase,
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| - double percent_increase,
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| - size_t samples_count_to_average,
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| - double min_sample_count_percent)
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| - : time_between_increase_(time_between_increase),
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| - percent_increase_(std::max(1.0, percent_increase)),
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| - time_at_last_change_(0),
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| - bandwidth_estimation_(initial_bandwidth_guess),
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| - accumulator_(samples_count_to_average),
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| - min_sample_count_percent_(
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| - std::min(1.0, std::max(0.0, min_sample_count_percent))) {
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| -}
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| -
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| -BandwidthSmoother::~BandwidthSmoother() = default;
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| -
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| -// Samples a new bandwidth measurement
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| -// returns true if the bandwidth estimation changed
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| -bool BandwidthSmoother::Sample(uint32_t sample_time, int bandwidth) {
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| - if (bandwidth < 0) {
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| - return false;
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| - }
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| -
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| - accumulator_.AddSample(bandwidth);
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| -
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| - if (accumulator_.count() < static_cast<size_t>(
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| - accumulator_.max_count() * min_sample_count_percent_)) {
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| - // We have not collected enough samples yet.
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| - return false;
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| - }
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| -
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| - // Replace bandwidth with the mean of sampled bandwidths.
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| - const int mean_bandwidth = static_cast<int>(accumulator_.ComputeMean());
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| -
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| - if (mean_bandwidth < bandwidth_estimation_) {
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| - time_at_last_change_ = sample_time;
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| - bandwidth_estimation_ = mean_bandwidth;
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| - return true;
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| - }
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| -
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| - const int old_bandwidth_estimation = bandwidth_estimation_;
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| - const double increase_threshold_d = percent_increase_ * bandwidth_estimation_;
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| - if (increase_threshold_d > INT_MAX) {
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| - // If bandwidth goes any higher we would overflow.
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| - return false;
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| - }
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| -
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| - const int increase_threshold = static_cast<int>(increase_threshold_d);
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| - if (mean_bandwidth < increase_threshold) {
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| - time_at_last_change_ = sample_time;
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| - // The value of bandwidth_estimation remains the same if we don't exceed
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| - // percent_increase_ * bandwidth_estimation_ for at least
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| - // time_between_increase_ time.
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| - } else if (sample_time >= time_at_last_change_ + time_between_increase_) {
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| - time_at_last_change_ = sample_time;
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| - if (increase_threshold == 0) {
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| - // Bandwidth_estimation_ must be zero. Assume a jump from zero to a
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| - // positive bandwidth means we have regained connectivity.
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| - bandwidth_estimation_ = mean_bandwidth;
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| - } else {
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| - bandwidth_estimation_ = increase_threshold;
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| - }
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| - }
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| - // Else don't make a change.
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| -
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| - return old_bandwidth_estimation != bandwidth_estimation_;
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| -}
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| -
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| -} // namespace rtc
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|
|