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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
117 lines
3.7 KiB
C++
117 lines
3.7 KiB
C++
/*
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* Copyright (c) 2017 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 "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
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#include <algorithm>
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#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
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#include "webrtc/system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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enum class DelayReliabilityCategory {
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kNone,
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kPoor,
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kMedium,
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kGood,
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kExcellent,
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kNumCategories
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};
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enum class DelayChangesCategory {
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kNone,
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kFew,
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kSeveral,
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kMany,
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kConstant,
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kNumCategories
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};
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} // namespace
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void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
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size_t buffer_delay_blocks) {
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++call_counter_;
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if (!initial_update) {
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if (delay_samples) {
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++reliable_delay_estimate_counter_;
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size_t delay_blocks = (*delay_samples) / kBlockSize;
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if (delay_blocks != delay_blocks_) {
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++delay_change_counter_;
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delay_blocks_ = delay_blocks;
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}
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}
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} else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
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initial_update = false;
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}
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if (call_counter_ == kMetricsReportingIntervalBlocks) {
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int value_to_report = static_cast<int>(delay_blocks_);
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value_to_report = std::min(124, value_to_report);
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RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
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value_to_report, 0, 124, 125);
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value_to_report = static_cast<int>(buffer_delay_blocks);
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value_to_report = std::min(124, value_to_report);
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RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
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value_to_report, 0, 124, 125);
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DelayReliabilityCategory delay_reliability;
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if (reliable_delay_estimate_counter_ == 0) {
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delay_reliability = DelayReliabilityCategory::kNone;
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} else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
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delay_reliability = DelayReliabilityCategory::kExcellent;
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} else if (reliable_delay_estimate_counter_ > 100) {
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delay_reliability = DelayReliabilityCategory::kGood;
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} else if (reliable_delay_estimate_counter_ > 10) {
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delay_reliability = DelayReliabilityCategory::kMedium;
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} else {
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delay_reliability = DelayReliabilityCategory::kPoor;
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}
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RTC_HISTOGRAM_ENUMERATION(
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"WebRTC.Audio.EchoCanceller.ReliableDelayEstimates",
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static_cast<int>(delay_reliability),
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static_cast<int>(DelayReliabilityCategory::kNumCategories));
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DelayChangesCategory delay_changes;
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if (delay_change_counter_ == 0) {
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delay_changes = DelayChangesCategory::kNone;
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} else if (delay_change_counter_ > 10) {
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delay_changes = DelayChangesCategory::kConstant;
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} else if (delay_change_counter_ > 5) {
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delay_changes = DelayChangesCategory::kMany;
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} else if (delay_change_counter_ > 2) {
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delay_changes = DelayChangesCategory::kSeveral;
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} else {
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delay_changes = DelayChangesCategory::kFew;
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}
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RTC_HISTOGRAM_ENUMERATION(
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"WebRTC.Audio.EchoCanceller.DelayChanges",
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static_cast<int>(delay_changes),
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static_cast<int>(DelayChangesCategory::kNumCategories));
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metrics_reported_ = true;
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call_counter_ = 0;
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ResetMetrics();
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} else {
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metrics_reported_ = false;
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}
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}
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void RenderDelayControllerMetrics::ResetMetrics() {
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delay_change_counter_ = 0;
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reliable_delay_estimate_counter_ = 0;
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}
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} // namespace webrtc
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