| /* |
| * 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 |