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Delete bandwidthsmoother.h.

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BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2367213004
Cr-Commit-Position: refs/heads/master@{#14498}
This commit is contained in:
nisse
2016-10-04 05:58:28 -07:00
committed by Commit bot
parent 7ba305111a
commit 312303340c
6 changed files with 0 additions and 273 deletions
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1
webrtc/BUILD.gn
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@ -336,7 +336,6 @@ if (rtc_include_tests) {
"base/array_view_unittest.cc",
"base/atomicops_unittest.cc",
"base/autodetectproxy_unittest.cc",
"base/bandwidthsmoother_unittest.cc",
"base/base64_unittest.cc",
"base/basictypes_unittest.cc",
"base/bind_unittest.cc",

2
webrtc/base/BUILD.gn
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@ -447,8 +447,6 @@ rtc_static_library("rtc_base") {
} else {
configs += [ ":rtc_base_warnings_config" ]
sources += [
"bandwidthsmoother.cc",
"bandwidthsmoother.h",
"callback.h",
"fileutils_mock.h",
"httpserver.cc",

86
webrtc/base/bandwidthsmoother.cc
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@ -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

60
webrtc/base/bandwidthsmoother.h
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@ -1,60 +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.
*/
#ifndef WEBRTC_BASE_BANDWIDTHSMOOTHER_H_
#define WEBRTC_BASE_BANDWIDTHSMOOTHER_H_
#include "webrtc/base/rollingaccumulator.h"
#include "webrtc/base/timeutils.h"
namespace rtc {
// The purpose of BandwidthSmoother is to smooth out bandwidth
// estimations so that 'trstate' messages can be triggered when we
// are "sure" there is sufficient bandwidth. To avoid frequent fluctuations,
// we take a slightly pessimistic view of our bandwidth. We only increase
// our estimation when we have sampled bandwidth measurements of values
// at least as large as the current estimation * percent_increase
// for at least time_between_increase time. If a sampled bandwidth
// is less than our current estimation we immediately decrease our estimation
// to that sampled value.
// We retain the initial bandwidth guess as our current bandwidth estimation
// until we have received (min_sample_count_percent * samples_count_to_average)
// number of samples. Min_sample_count_percent must be in range [0, 1].
class BandwidthSmoother {
public:
BandwidthSmoother(int initial_bandwidth_guess,
uint32_t time_between_increase,
double percent_increase,
size_t samples_count_to_average,
double min_sample_count_percent);
~BandwidthSmoother();
// Samples a new bandwidth measurement.
// bandwidth is expected to be non-negative.
// returns true if the bandwidth estimation changed
bool Sample(uint32_t sample_time, int bandwidth);
int get_bandwidth_estimation() const {
return bandwidth_estimation_;
}
private:
uint32_t time_between_increase_;
double percent_increase_;
uint32_t time_at_last_change_;
int bandwidth_estimation_;
RollingAccumulator<int> accumulator_;
double min_sample_count_percent_;
};
} // namespace rtc
#endif // WEBRTC_BASE_BANDWIDTHSMOOTHER_H_

122
webrtc/base/bandwidthsmoother_unittest.cc
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@ -1,122 +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 <limits.h>
#include "webrtc/base/bandwidthsmoother.h"
#include "webrtc/base/gunit.h"
namespace rtc {
static const int kTimeBetweenIncrease = 10;
static const double kPercentIncrease = 1.1;
static const size_t kSamplesCountToAverage = 2;
static const double kMinSampleCountPercent = 1.0;
TEST(BandwidthSmootherTest, TestSampleIncrease) {
BandwidthSmoother mon(1000, // initial_bandwidth_guess
kTimeBetweenIncrease,
kPercentIncrease,
kSamplesCountToAverage,
kMinSampleCountPercent);
int bandwidth_sample = 1000;
EXPECT_EQ(bandwidth_sample, mon.get_bandwidth_estimation());
bandwidth_sample =
static_cast<int>(bandwidth_sample * kPercentIncrease);
EXPECT_FALSE(mon.Sample(9, bandwidth_sample));
EXPECT_TRUE(mon.Sample(10, bandwidth_sample));
EXPECT_EQ(bandwidth_sample, mon.get_bandwidth_estimation());
int next_expected_est =
static_cast<int>(bandwidth_sample * kPercentIncrease);
bandwidth_sample *= 2;
EXPECT_TRUE(mon.Sample(20, bandwidth_sample));
EXPECT_EQ(next_expected_est, mon.get_bandwidth_estimation());
}
TEST(BandwidthSmootherTest, TestSampleIncreaseFromZero) {
BandwidthSmoother mon(0, // initial_bandwidth_guess
kTimeBetweenIncrease,
kPercentIncrease,
kSamplesCountToAverage,
kMinSampleCountPercent);
const int kBandwidthSample = 1000;
EXPECT_EQ(0, mon.get_bandwidth_estimation());
EXPECT_FALSE(mon.Sample(9, kBandwidthSample));
EXPECT_TRUE(mon.Sample(10, kBandwidthSample));
EXPECT_EQ(kBandwidthSample, mon.get_bandwidth_estimation());
}
TEST(BandwidthSmootherTest, TestSampleDecrease) {
BandwidthSmoother mon(1000, // initial_bandwidth_guess
kTimeBetweenIncrease,
kPercentIncrease,
kSamplesCountToAverage,
kMinSampleCountPercent);
const int kBandwidthSample = 999;
EXPECT_EQ(1000, mon.get_bandwidth_estimation());
EXPECT_FALSE(mon.Sample(1, kBandwidthSample));
EXPECT_EQ(1000, mon.get_bandwidth_estimation());
EXPECT_TRUE(mon.Sample(2, kBandwidthSample));
EXPECT_EQ(kBandwidthSample, mon.get_bandwidth_estimation());
}
TEST(BandwidthSmootherTest, TestSampleTooFewSamples) {
BandwidthSmoother mon(1000, // initial_bandwidth_guess
kTimeBetweenIncrease,
kPercentIncrease,
10, // 10 samples.
0.5); // 5 min samples.
const int kBandwidthSample = 500;
EXPECT_EQ(1000, mon.get_bandwidth_estimation());
EXPECT_FALSE(mon.Sample(1, kBandwidthSample));
EXPECT_FALSE(mon.Sample(2, kBandwidthSample));
EXPECT_FALSE(mon.Sample(3, kBandwidthSample));
EXPECT_FALSE(mon.Sample(4, kBandwidthSample));
EXPECT_EQ(1000, mon.get_bandwidth_estimation());
EXPECT_TRUE(mon.Sample(5, kBandwidthSample));
EXPECT_EQ(kBandwidthSample, mon.get_bandwidth_estimation());
}
// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5491
#ifdef UNDEFINED_SANITIZER
#define MAYBE_TestSampleRollover DISABLED_TestSampleRollover
#else
#define MAYBE_TestSampleRollover TestSampleRollover
#endif
TEST(BandwidthSmootherTest, MAYBE_TestSampleRollover) {
const int kHugeBandwidth = 2000000000; // > INT_MAX/1.1
BandwidthSmoother mon(kHugeBandwidth,
kTimeBetweenIncrease,
kPercentIncrease,
kSamplesCountToAverage,
kMinSampleCountPercent);
EXPECT_FALSE(mon.Sample(10, INT_MAX));
EXPECT_FALSE(mon.Sample(11, INT_MAX));
EXPECT_EQ(kHugeBandwidth, mon.get_bandwidth_estimation());
}
TEST(BandwidthSmootherTest, TestSampleNegative) {
BandwidthSmoother mon(1000, // initial_bandwidth_guess
kTimeBetweenIncrease,
kPercentIncrease,
kSamplesCountToAverage,
kMinSampleCountPercent);
EXPECT_FALSE(mon.Sample(10, -1));
EXPECT_FALSE(mon.Sample(11, -1));
EXPECT_EQ(1000, mon.get_bandwidth_estimation());
}
} // namespace rtc

2
webrtc/base/base.gyp
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@ -408,8 +408,6 @@
},
}, {
'sources': [
'bandwidthsmoother.cc',
'bandwidthsmoother.h',
'callback.h',
'fileutils_mock.h',
'httpserver.cc',