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