mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
62238097c9
This is a safe cleanup change since top-level const applied to parameters in function declarations (that are not also definitions) are ignored by the compiler. Hence, such changes do not change the type of the declared functions and are simply no-ops. Bug: webrtc:13610 Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ali Tofigh <alito@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35802}
114 lines
4.1 KiB
C++
114 lines
4.1 KiB
C++
/*
|
|
* Copyright (c) 2016 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
|
|
|
|
#include <stddef.h>
|
|
|
|
namespace webrtc {
|
|
|
|
#ifdef _MSC_VER /* visual c++ */
|
|
#define ALIGN16_BEG __declspec(align(16))
|
|
#define ALIGN16_END
|
|
#else /* gcc or icc */
|
|
#define ALIGN16_BEG
|
|
#define ALIGN16_END __attribute__((aligned(16)))
|
|
#endif
|
|
|
|
enum class Aec3Optimization { kNone, kSse2, kAvx2, kNeon };
|
|
|
|
constexpr int kNumBlocksPerSecond = 250;
|
|
|
|
constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
|
|
constexpr int kMetricsComputationBlocks = 3;
|
|
constexpr int kMetricsCollectionBlocks =
|
|
kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
|
|
|
|
constexpr size_t kFftLengthBy2 = 64;
|
|
constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
|
|
constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
|
|
constexpr size_t kFftLength = 2 * kFftLengthBy2;
|
|
constexpr size_t kFftLengthBy2Log2 = 6;
|
|
|
|
constexpr int kRenderTransferQueueSizeFrames = 100;
|
|
|
|
constexpr size_t kMaxNumBands = 3;
|
|
constexpr size_t kFrameSize = 160;
|
|
constexpr size_t kSubFrameLength = kFrameSize / 2;
|
|
|
|
constexpr size_t kBlockSize = kFftLengthBy2;
|
|
constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
|
|
|
|
constexpr size_t kExtendedBlockSize = 2 * kFftLengthBy2;
|
|
constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
|
|
constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks =
|
|
kMatchedFilterWindowSizeSubBlocks * 3 / 4;
|
|
|
|
// TODO(peah): Integrate this with how it is done inside audio_processing_impl.
|
|
constexpr size_t NumBandsForRate(int sample_rate_hz) {
|
|
return static_cast<size_t>(sample_rate_hz / 16000);
|
|
}
|
|
|
|
constexpr bool ValidFullBandRate(int sample_rate_hz) {
|
|
return sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
|
|
sample_rate_hz == 48000;
|
|
}
|
|
|
|
constexpr int GetTimeDomainLength(int filter_length_blocks) {
|
|
return filter_length_blocks * kFftLengthBy2;
|
|
}
|
|
|
|
constexpr size_t GetDownSampledBufferSize(size_t down_sampling_factor,
|
|
size_t num_matched_filters) {
|
|
return kBlockSize / down_sampling_factor *
|
|
(kMatchedFilterAlignmentShiftSizeSubBlocks * num_matched_filters +
|
|
kMatchedFilterWindowSizeSubBlocks + 1);
|
|
}
|
|
|
|
constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
|
|
size_t num_matched_filters,
|
|
size_t filter_length_blocks) {
|
|
return GetDownSampledBufferSize(down_sampling_factor, num_matched_filters) /
|
|
(kBlockSize / down_sampling_factor) +
|
|
filter_length_blocks + 1;
|
|
}
|
|
|
|
// Detects what kind of optimizations to use for the code.
|
|
Aec3Optimization DetectOptimization();
|
|
|
|
// Computes the log2 of the input in a fast an approximate manner.
|
|
float FastApproxLog2f(float in);
|
|
|
|
// Returns dB from a power quantity expressed in log2.
|
|
float Log2TodB(float in_log2);
|
|
|
|
static_assert(1 << kBlockSizeLog2 == kBlockSize,
|
|
"Proper number of shifts for blocksize");
|
|
|
|
static_assert(1 << kFftLengthBy2Log2 == kFftLengthBy2,
|
|
"Proper number of shifts for the fft length");
|
|
|
|
static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
|
|
static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
|
|
static_assert(3 == NumBandsForRate(48000), "Number of bands for 48 kHz");
|
|
|
|
static_assert(ValidFullBandRate(16000),
|
|
"Test that 16 kHz is a valid sample rate");
|
|
static_assert(ValidFullBandRate(32000),
|
|
"Test that 32 kHz is a valid sample rate");
|
|
static_assert(ValidFullBandRate(48000),
|
|
"Test that 48 kHz is a valid sample rate");
|
|
static_assert(!ValidFullBandRate(8001),
|
|
"Test that 8001 Hz is not a valid sample rate");
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
|