| /* | |
| * Copyright (c) 2021 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/agc/clipping_predictor_level_buffer.h" | |
| #include <algorithm> | |
| #include <cmath> | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/logging.h" | |
| namespace webrtc { | |
| bool ClippingPredictorLevelBuffer::Level::operator==(const Level& level) const { | |
| constexpr float kEpsilon = 1e-6f; | |
| return std::fabs(average - level.average) < kEpsilon && | |
| std::fabs(max - level.max) < kEpsilon; | |
| } | |
| ClippingPredictorLevelBuffer::ClippingPredictorLevelBuffer(int capacity) | |
| : tail_(-1), size_(0), data_(std::max(1, capacity)) { | |
| if (capacity > kMaxCapacity) { | |
| RTC_LOG(LS_WARNING) << "[agc]: ClippingPredictorLevelBuffer exceeds the " | |
| << "maximum allowed capacity. Capacity: " << capacity; | |
| } | |
| RTC_DCHECK(!data_.empty()); | |
| } | |
| void ClippingPredictorLevelBuffer::Reset() { | |
| tail_ = -1; | |
| size_ = 0; | |
| } | |
| void ClippingPredictorLevelBuffer::Push(Level level) { | |
| ++tail_; | |
| if (tail_ == Capacity()) { | |
| tail_ = 0; | |
| } | |
| if (size_ < Capacity()) { | |
| size_++; | |
| } | |
| data_[tail_] = level; | |
| } | |
| // TODO(bugs.webrtc.org/12774): Optimize partial computation for long buffers. | |
| absl::optional<ClippingPredictorLevelBuffer::Level> | |
| ClippingPredictorLevelBuffer::ComputePartialMetrics(int delay, | |
| int num_items) const { | |
| RTC_DCHECK_GE(delay, 0); | |
| RTC_DCHECK_LT(delay, Capacity()); | |
| RTC_DCHECK_GT(num_items, 0); | |
| RTC_DCHECK_LE(num_items, Capacity()); | |
| RTC_DCHECK_LE(delay + num_items, Capacity()); | |
| if (delay + num_items > Size()) { | |
| return absl::nullopt; | |
| } | |
| float sum = 0.0f; | |
| float max = 0.0f; | |
| for (int i = 0; i < num_items && i < Size(); ++i) { | |
| int idx = tail_ - delay - i; | |
| if (idx < 0) { | |
| idx += Capacity(); | |
| } | |
| sum += data_[idx].average; | |
| max = std::fmax(data_[idx].max, max); | |
| } | |
| return absl::optional<Level>({sum / static_cast<float>(num_items), max}); | |
| } | |
| } // namespace webrtc |