| /* | |
| * Copyright (c) 2017 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/render_delay_controller_metrics.h" | |
| #include <algorithm> | |
| #include "modules/audio_processing/aec3/aec3_common.h" | |
| #include "system_wrappers/include/metrics.h" | |
| namespace webrtc { | |
| namespace { | |
| enum class DelayReliabilityCategory { | |
| kNone, | |
| kPoor, | |
| kMedium, | |
| kGood, | |
| kExcellent, | |
| kNumCategories | |
| }; | |
| enum class DelayChangesCategory { | |
| kNone, | |
| kFew, | |
| kSeveral, | |
| kMany, | |
| kConstant, | |
| kNumCategories | |
| }; | |
| constexpr int kMaxSkewShiftCount = 20; | |
| } // namespace | |
| RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default; | |
| void RenderDelayControllerMetrics::Update( | |
| absl::optional<size_t> delay_samples, | |
| size_t buffer_delay_blocks, | |
| absl::optional<int> skew_shift_blocks) { | |
| ++call_counter_; | |
| if (!initial_update) { | |
| size_t delay_blocks; | |
| if (delay_samples) { | |
| ++reliable_delay_estimate_counter_; | |
| delay_blocks = (*delay_samples) / kBlockSize + 2; | |
| } else { | |
| delay_blocks = 0; | |
| } | |
| if (delay_blocks != delay_blocks_) { | |
| ++delay_change_counter_; | |
| delay_blocks_ = delay_blocks; | |
| } | |
| if (skew_shift_blocks) { | |
| skew_shift_count_ = std::min(kMaxSkewShiftCount, skew_shift_count_); | |
| } | |
| } else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) { | |
| initial_update = false; | |
| } | |
| if (call_counter_ == kMetricsReportingIntervalBlocks) { | |
| int value_to_report = static_cast<int>(delay_blocks_); | |
| value_to_report = std::min(124, value_to_report >> 1); | |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay", | |
| value_to_report, 0, 124, 125); | |
| value_to_report = static_cast<int>(buffer_delay_blocks + 2); | |
| value_to_report = std::min(124, value_to_report >> 1); | |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay", | |
| value_to_report, 0, 124, 125); | |
| DelayReliabilityCategory delay_reliability; | |
| if (reliable_delay_estimate_counter_ == 0) { | |
| delay_reliability = DelayReliabilityCategory::kNone; | |
| } else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) { | |
| delay_reliability = DelayReliabilityCategory::kExcellent; | |
| } else if (reliable_delay_estimate_counter_ > 100) { | |
| delay_reliability = DelayReliabilityCategory::kGood; | |
| } else if (reliable_delay_estimate_counter_ > 10) { | |
| delay_reliability = DelayReliabilityCategory::kMedium; | |
| } else { | |
| delay_reliability = DelayReliabilityCategory::kPoor; | |
| } | |
| RTC_HISTOGRAM_ENUMERATION( | |
| "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates", | |
| static_cast<int>(delay_reliability), | |
| static_cast<int>(DelayReliabilityCategory::kNumCategories)); | |
| DelayChangesCategory delay_changes; | |
| if (delay_change_counter_ == 0) { | |
| delay_changes = DelayChangesCategory::kNone; | |
| } else if (delay_change_counter_ > 10) { | |
| delay_changes = DelayChangesCategory::kConstant; | |
| } else if (delay_change_counter_ > 5) { | |
| delay_changes = DelayChangesCategory::kMany; | |
| } else if (delay_change_counter_ > 2) { | |
| delay_changes = DelayChangesCategory::kSeveral; | |
| } else { | |
| delay_changes = DelayChangesCategory::kFew; | |
| } | |
| RTC_HISTOGRAM_ENUMERATION( | |
| "WebRTC.Audio.EchoCanceller.DelayChanges", | |
| static_cast<int>(delay_changes), | |
| static_cast<int>(DelayChangesCategory::kNumCategories)); | |
| metrics_reported_ = true; | |
| call_counter_ = 0; | |
| ResetMetrics(); | |
| } else { | |
| metrics_reported_ = false; | |
| } | |
| if (!initial_update && ++skew_report_timer_ == 60 * kNumBlocksPerSecond) { | |
| RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxSkewShiftCount", | |
| skew_shift_count_, 0, kMaxSkewShiftCount, | |
| kMaxSkewShiftCount + 1); | |
| skew_shift_count_ = 0; | |
| skew_report_timer_ = 0; | |
| } | |
| } | |
| void RenderDelayControllerMetrics::ResetMetrics() { | |
| delay_change_counter_ = 0; | |
| reliable_delay_estimate_counter_ = 0; | |
| } | |
| } // namespace webrtc |