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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" | 11 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" |
| 12 | 12 |
| 13 #include <assert.h> | 13 #include <assert.h> |
| 14 #include <string.h> // memset | 14 #include <string.h> // memset |
| 15 | 15 |
| 16 #include "webrtc/base/checks.h" | 16 #include "webrtc/base/checks.h" |
| 17 #include "webrtc/modules/audio_coding/neteq/decision_logic.h" | 17 #include "webrtc/modules/audio_coding/neteq/decision_logic.h" |
| 18 #include "webrtc/modules/audio_coding/neteq/delay_manager.h" | 18 #include "webrtc/modules/audio_coding/neteq/delay_manager.h" |
| 19 #include "webrtc/system_wrappers/interface/metrics.h" | 19 #include "webrtc/system_wrappers/interface/metrics.h" |
| 20 | 20 |
| 21 namespace webrtc { | 21 namespace webrtc { |
| 22 | 22 |
| 23 StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger( |
| 24 const std::string& uma_name, |
| 25 int report_interval_ms, |
| 26 int max_value) |
| 27 : uma_name_(uma_name), |
| 28 report_interval_ms_(report_interval_ms), |
| 29 max_value_(max_value), |
| 30 timer_(0) { |
| 31 } |
| 32 |
| 33 StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default; |
| 34 |
| 35 void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) { |
| 36 timer_ += step_ms; |
| 37 if (timer_ < report_interval_ms_) { |
| 38 return; |
| 39 } |
| 40 LogToUma(Metric()); |
| 41 Reset(); |
| 42 timer_ -= report_interval_ms_; |
| 43 DCHECK_GE(timer_, 0); |
| 44 } |
| 45 |
| 46 void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const { |
| 47 RTC_HISTOGRAM_COUNTS(uma_name_, value, 1, max_value_, 50); |
| 48 } |
| 49 |
| 50 StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter:: |
| 51 DelayedPacketOutagesPerMinuteCounter() |
| 52 : PeriodicUmaLogger("WebRTC.Audio.DelayedPacketOutageEventsPerMinute", |
| 53 60000, // 60 seconds report interval. |
| 54 100) { |
| 55 } |
| 56 |
| 23 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter:: | 57 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter:: |
| 24 RegisterEvent() { | 58 RegisterEvent() { |
| 25 ++counter_; | 59 ++counter_; |
| 26 } | 60 } |
| 27 | 61 |
| 28 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter::AdvanceClock( | 62 int StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter::Metric() const { |
| 29 int step_ms) { | 63 return counter_; |
| 30 timer_ += step_ms; | |
| 31 if (timer_ < kReportIntervalMs) { | |
| 32 return; | |
| 33 } | |
| 34 LogToUma(); | |
| 35 counter_ = 0; | |
| 36 timer_ -= kReportIntervalMs; | |
| 37 DCHECK_GE(timer_, 0); | |
| 38 } | 64 } |
| 39 | 65 |
| 40 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter::LogToUma() | 66 void StatisticsCalculator::DelayedPacketOutagesPerMinuteCounter::Reset() { |
| 41 const { | 67 counter_ = 0; |
| 42 RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.DelayedPacketOutageEventsPerMinute", | 68 } |
| 43 counter_); | 69 |
| 70 StatisticsCalculator::AverageExcessBufferDelayMs::AverageExcessBufferDelayMs() |
| 71 : PeriodicUmaLogger("WebRTC.Audio.AverageExcessBufferDelayMs", |
| 72 60000, // 60 seconds report interval. |
| 73 1000) { |
| 44 } | 74 } |
| 45 | 75 |
| 46 void StatisticsCalculator::AverageExcessBufferDelayMs:: | 76 void StatisticsCalculator::AverageExcessBufferDelayMs:: |
| 47 RegisterPacketWaitingTime(int time_ms) { | 77 RegisterPacketWaitingTime(int time_ms) { |
| 48 sum_ += time_ms; | 78 sum_ += time_ms; |
| 49 ++num_packets_; | 79 ++num_packets_; |
| 50 } | 80 } |
| 51 | 81 |
| 52 void StatisticsCalculator::AverageExcessBufferDelayMs::AdvanceClock( | 82 int StatisticsCalculator::AverageExcessBufferDelayMs::Metric() const { |
| 53 int step_ms) { | 83 return static_cast<int>(sum_ / num_packets_); |
| 54 timer_ += step_ms; | 84 } |
| 55 if (timer_ < kReportIntervalMs) { | 85 |
| 56 return; | 86 void StatisticsCalculator::AverageExcessBufferDelayMs::Reset() { |
| 57 } | |
| 58 LogToUma(); | |
| 59 sum_ = 0.0; | 87 sum_ = 0.0; |
| 60 num_packets_ = 0; | 88 num_packets_ = 0; |
| 61 timer_ -= kReportIntervalMs; | |
| 62 DCHECK_GE(timer_, 0); | |
| 63 } | |
| 64 | |
| 65 void StatisticsCalculator::AverageExcessBufferDelayMs::LogToUma() { | |
| 66 if (num_packets_ > 0) { | |
| 67 RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.AverageExcessBufferDelayMs", | |
| 68 static_cast<int>(sum_ / num_packets_)); | |
| 69 } | |
| 70 } | 89 } |
| 71 | 90 |
| 72 StatisticsCalculator::StatisticsCalculator() | 91 StatisticsCalculator::StatisticsCalculator() |
| 73 : preemptive_samples_(0), | 92 : preemptive_samples_(0), |
| 74 accelerate_samples_(0), | 93 accelerate_samples_(0), |
| 75 added_zero_samples_(0), | 94 added_zero_samples_(0), |
| 76 expanded_speech_samples_(0), | 95 expanded_speech_samples_(0), |
| 77 expanded_noise_samples_(0), | 96 expanded_noise_samples_(0), |
| 78 discarded_packets_(0), | 97 discarded_packets_(0), |
| 79 lost_timestamps_(0), | 98 lost_timestamps_(0), |
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| 238 // Ratio must be smaller than 1 in Q14. | 257 // Ratio must be smaller than 1 in Q14. |
| 239 assert((numerator << 14) / denominator < (1 << 14)); | 258 assert((numerator << 14) / denominator < (1 << 14)); |
| 240 return static_cast<uint16_t>((numerator << 14) / denominator); | 259 return static_cast<uint16_t>((numerator << 14) / denominator); |
| 241 } else { | 260 } else { |
| 242 // Will not produce a ratio larger than 1, since this is probably an error. | 261 // Will not produce a ratio larger than 1, since this is probably an error. |
| 243 return 1 << 14; | 262 return 1 << 14; |
| 244 } | 263 } |
| 245 } | 264 } |
| 246 | 265 |
| 247 } // namespace webrtc | 266 } // namespace webrtc |
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