tg2sip/webrtc_dsp/modules/audio_processing/aec3/aec3_common.h

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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
/*
* 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, kNeon };
constexpr int kNumBlocksPerSecond = 250;
constexpr int kMetricsReportingIntervalBlocks = 10 * kNumBlocksPerSecond;
constexpr int kMetricsComputationBlocks = 11;
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 kMaxAdaptiveFilterLength = 50;
constexpr int kRenderTransferQueueSizeFrames = 100;
constexpr size_t kMaxNumBands = 3;
constexpr size_t kSubFrameLength = 80;
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 == 8000 ? 1
: sample_rate_hz / 16000);
}
constexpr int LowestBandRate(int sample_rate_hz) {
return sample_rate_hz == 8000 ? sample_rate_hz : 16000;
}
constexpr bool ValidFullBandRate(int sample_rate_hz) {
return sample_rate_hz == 8000 || 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(const float in);
// Returns dB from a power quantity expressed in log2.
float Log2TodB(const 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(8000), "Number of bands for 8 kHz");
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(8000 == LowestBandRate(8000), "Sample rate of band 0 for 8 kHz");
static_assert(16000 == LowestBandRate(16000),
"Sample rate of band 0 for 16 kHz");
static_assert(16000 == LowestBandRate(32000),
"Sample rate of band 0 for 32 kHz");
static_assert(16000 == LowestBandRate(48000),
"Sample rate of band 0 for 48 kHz");
static_assert(ValidFullBandRate(8000),
"Test that 8 kHz is a valid sample rate");
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_