122 lines
4.0 KiB
C++
122 lines
4.0 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* 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
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
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#include <algorithm>
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#include <limits>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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bool TimeToReportMetrics(int frames_since_last_report) {
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constexpr int kNumFramesPerSecond = 100;
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constexpr int kReportingIntervalFrames = 10 * kNumFramesPerSecond;
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return frames_since_last_report == kReportingIntervalFrames;
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}
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} // namespace
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ApiCallJitterMetrics::Jitter::Jitter()
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: max_(0), min_(std::numeric_limits<int>::max()) {}
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void ApiCallJitterMetrics::Jitter::Update(int num_api_calls_in_a_row) {
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min_ = std::min(min_, num_api_calls_in_a_row);
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max_ = std::max(max_, num_api_calls_in_a_row);
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}
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void ApiCallJitterMetrics::Jitter::Reset() {
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min_ = std::numeric_limits<int>::max();
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max_ = 0;
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}
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void ApiCallJitterMetrics::Reset() {
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render_jitter_.Reset();
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capture_jitter_.Reset();
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num_api_calls_in_a_row_ = 0;
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frames_since_last_report_ = 0;
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last_call_was_render_ = false;
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proper_call_observed_ = false;
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}
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void ApiCallJitterMetrics::ReportRenderCall() {
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if (!last_call_was_render_) {
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// If the previous call was a capture and a proper call has been observed
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// (containing both render and capture data), storing the last number of
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// capture calls into the metrics.
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if (proper_call_observed_) {
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capture_jitter_.Update(num_api_calls_in_a_row_);
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}
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// Reset the call counter to start counting render calls.
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num_api_calls_in_a_row_ = 0;
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}
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++num_api_calls_in_a_row_;
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last_call_was_render_ = true;
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}
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void ApiCallJitterMetrics::ReportCaptureCall() {
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if (last_call_was_render_) {
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// If the previous call was a render and a proper call has been observed
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// (containing both render and capture data), storing the last number of
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// render calls into the metrics.
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if (proper_call_observed_) {
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render_jitter_.Update(num_api_calls_in_a_row_);
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}
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// Reset the call counter to start counting capture calls.
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num_api_calls_in_a_row_ = 0;
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// If this statement is reached, at least one render and one capture call
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// have been observed.
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proper_call_observed_ = true;
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}
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++num_api_calls_in_a_row_;
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last_call_was_render_ = false;
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// Only report and update jitter metrics for when a proper call, containing
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// both render and capture data, has been observed.
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if (proper_call_observed_ &&
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TimeToReportMetrics(++frames_since_last_report_)) {
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// Report jitter, where the base basic unit is frames.
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constexpr int kMaxJitterToReport = 50;
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// Report max and min jitter for render and capture, in units of 20 ms.
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RTC_HISTOGRAM_COUNTS_LINEAR(
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"WebRTC.Audio.EchoCanceller.MaxRenderJitter",
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std::min(kMaxJitterToReport, render_jitter().max()), 1,
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kMaxJitterToReport, kMaxJitterToReport);
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RTC_HISTOGRAM_COUNTS_LINEAR(
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"WebRTC.Audio.EchoCanceller.MinRenderJitter",
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std::min(kMaxJitterToReport, render_jitter().min()), 1,
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kMaxJitterToReport, kMaxJitterToReport);
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RTC_HISTOGRAM_COUNTS_LINEAR(
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"WebRTC.Audio.EchoCanceller.MaxCaptureJitter",
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std::min(kMaxJitterToReport, capture_jitter().max()), 1,
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kMaxJitterToReport, kMaxJitterToReport);
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RTC_HISTOGRAM_COUNTS_LINEAR(
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"WebRTC.Audio.EchoCanceller.MinCaptureJitter",
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std::min(kMaxJitterToReport, capture_jitter().min()), 1,
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kMaxJitterToReport, kMaxJitterToReport);
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frames_since_last_report_ = 0;
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Reset();
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}
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}
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bool ApiCallJitterMetrics::WillReportMetricsAtNextCapture() const {
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return TimeToReportMetrics(frames_since_last_report_ + 1);
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}
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} // namespace webrtc
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