mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
130 lines
4.8 KiB
C++
130 lines
4.8 KiB
C++
/*
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* Copyright (c) 2016 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
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#include <stddef.h>
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namespace webrtc {
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#ifdef _MSC_VER /* visual c++ */
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#define ALIGN16_BEG __declspec(align(16))
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#define ALIGN16_END
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#else /* gcc or icc */
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#define ALIGN16_BEG
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#define ALIGN16_END __attribute__((aligned(16)))
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#endif
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enum class Aec3Optimization { kNone, kSse2, kNeon };
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constexpr int kNumBlocksPerSecond = 250;
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constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
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constexpr int kMetricsComputationBlocks = 11;
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constexpr int kMetricsCollectionBlocks =
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kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
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constexpr size_t kFftLengthBy2 = 64;
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constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
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constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
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constexpr size_t kFftLength = 2 * kFftLengthBy2;
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constexpr size_t kFftLengthBy2Log2 = 6;
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constexpr int kMaxAdaptiveFilterLength = 50;
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constexpr int kRenderTransferQueueSizeFrames = 100;
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constexpr size_t kMaxNumBands = 3;
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constexpr size_t kSubFrameLength = 80;
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constexpr size_t kBlockSize = kFftLengthBy2;
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constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
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constexpr size_t kExtendedBlockSize = 2 * kFftLengthBy2;
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constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
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constexpr size_t kMatchedFilterAlignmentShiftSizeSubBlocks =
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kMatchedFilterWindowSizeSubBlocks * 3 / 4;
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// TODO(peah): Integrate this with how it is done inside audio_processing_impl.
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constexpr size_t NumBandsForRate(int sample_rate_hz) {
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return static_cast<size_t>(sample_rate_hz == 8000 ? 1
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: sample_rate_hz / 16000);
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}
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constexpr int LowestBandRate(int sample_rate_hz) {
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return sample_rate_hz == 8000 ? sample_rate_hz : 16000;
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}
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constexpr bool ValidFullBandRate(int sample_rate_hz) {
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return sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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sample_rate_hz == 32000 || sample_rate_hz == 48000;
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}
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constexpr int GetTimeDomainLength(int filter_length_blocks) {
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return filter_length_blocks * kFftLengthBy2;
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}
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constexpr size_t GetDownSampledBufferSize(size_t down_sampling_factor,
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size_t num_matched_filters) {
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return kBlockSize / down_sampling_factor *
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(kMatchedFilterAlignmentShiftSizeSubBlocks * num_matched_filters +
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kMatchedFilterWindowSizeSubBlocks + 1);
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}
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constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
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size_t num_matched_filters,
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size_t filter_length_blocks) {
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return GetDownSampledBufferSize(down_sampling_factor, num_matched_filters) /
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(kBlockSize / down_sampling_factor) +
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filter_length_blocks + 1;
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}
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// Detects what kind of optimizations to use for the code.
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Aec3Optimization DetectOptimization();
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// Computes the log2 of the input in a fast an approximate manner.
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float FastApproxLog2f(const float in);
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// Returns dB from a power quantity expressed in log2.
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float Log2TodB(const float in_log2);
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static_assert(1 << kBlockSizeLog2 == kBlockSize,
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"Proper number of shifts for blocksize");
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static_assert(1 << kFftLengthBy2Log2 == kFftLengthBy2,
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"Proper number of shifts for the fft length");
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static_assert(1 == NumBandsForRate(8000), "Number of bands for 8 kHz");
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static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
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static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
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static_assert(3 == NumBandsForRate(48000), "Number of bands for 48 kHz");
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static_assert(8000 == LowestBandRate(8000), "Sample rate of band 0 for 8 kHz");
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static_assert(16000 == LowestBandRate(16000),
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"Sample rate of band 0 for 16 kHz");
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static_assert(16000 == LowestBandRate(32000),
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"Sample rate of band 0 for 32 kHz");
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static_assert(16000 == LowestBandRate(48000),
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"Sample rate of band 0 for 48 kHz");
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static_assert(ValidFullBandRate(8000),
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"Test that 8 kHz is a valid sample rate");
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static_assert(ValidFullBandRate(16000),
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"Test that 16 kHz is a valid sample rate");
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static_assert(ValidFullBandRate(32000),
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"Test that 32 kHz is a valid sample rate");
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static_assert(ValidFullBandRate(48000),
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"Test that 48 kHz is a valid sample rate");
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static_assert(!ValidFullBandRate(8001),
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"Test that 8001 Hz is not a valid sample rate");
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
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