tg2sip/webrtc_dsp/modules/audio_processing/ns/nsx_defines.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) 2012 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_NS_MAIN_SOURCE_NSX_DEFINES_H_
#define MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
#define ANAL_BLOCKL_MAX 256 /* Max analysis block length */
#define HALF_ANAL_BLOCKL 129 /* Half max analysis block length + 1 */
#define NUM_HIGH_BANDS_MAX 2 /* Max number of high bands */
#define SIMULT 3
#define END_STARTUP_LONG 200
#define END_STARTUP_SHORT 50
#define FACTOR_Q16 2621440 /* 40 in Q16 */
#define FACTOR_Q7 5120 /* 40 in Q7 */
#define FACTOR_Q7_STARTUP 1024 /* 8 in Q7 */
#define WIDTH_Q8 3 /* 0.01 in Q8 (or 25 ) */
/* PARAMETERS FOR NEW METHOD */
#define DD_PR_SNR_Q11 2007 /* ~= Q11(0.98) DD update of prior SNR */
#define ONE_MINUS_DD_PR_SNR_Q11 41 /* DD update of prior SNR */
#define SPECT_FLAT_TAVG_Q14 \
4915 /* (0.30) tavg parameter for spectral flatness measure */
#define SPECT_DIFF_TAVG_Q8 \
77 /* (0.30) tavg parameter for spectral flatness measure */
#define PRIOR_UPDATE_Q14 1638 /* Q14(0.1) Update parameter of prior model */
#define NOISE_UPDATE_Q8 26 /* 26 ~= Q8(0.1) Update parameter for noise */
/* Probability threshold for noise state in speech/noise likelihood. */
#define ONE_MINUS_PROB_RANGE_Q8 205 /* 205 ~= Q8(0.8) */
#define HIST_PAR_EST 1000 /* Histogram size for estimation of parameters */
/* FEATURE EXTRACTION CONFIG */
/* Bin size of histogram */
#define BIN_SIZE_LRT 10
/* Scale parameters: multiply dominant peaks of the histograms by scale factor
* to obtain. */
/* Thresholds for prior model */
#define FACTOR_1_LRT_DIFF \
6 /* For LRT and spectral difference (5 times bigger) */
/* For spectral_flatness: used when noise is flatter than speech (10 times
* bigger). */
#define FACTOR_2_FLAT_Q10 922
/* Peak limit for spectral flatness (varies between 0 and 1) */
#define THRES_PEAK_FLAT 24 /* * 2 * BIN_SIZE_FLAT_FX */
/* Limit on spacing of two highest peaks in histogram: spacing determined by bin
* size. */
#define LIM_PEAK_SPACE_FLAT_DIFF 4 /* * 2 * BIN_SIZE_DIFF_FX */
/* Limit on relevance of second peak */
#define LIM_PEAK_WEIGHT_FLAT_DIFF 2
#define THRES_FLUCT_LRT \
10240 /* = 20 * inst->modelUpdate; fluctuation limit of LRT feat. */
/* Limit on the max and min values for the feature thresholds */
#define MAX_FLAT_Q10 38912 /* * 2 * BIN_SIZE_FLAT_FX */
#define MIN_FLAT_Q10 4096 /* * 2 * BIN_SIZE_FLAT_FX */
#define MAX_DIFF 100 /* * 2 * BIN_SIZE_DIFF_FX */
#define MIN_DIFF 16 /* * 2 * BIN_SIZE_DIFF_FX */
/* Criteria of weight of histogram peak to accept/reject feature */
#define THRES_WEIGHT_FLAT_DIFF \
154 /*(int)(0.3*(inst->modelUpdate)) for flatness and difference */
#define STAT_UPDATES 9 /* Update every 512 = 1 << 9 block */
#define ONE_MINUS_GAMMA_PAUSE_Q8 \
13 /* ~= Q8(0.05) Update for conservative noise estimate */
#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 \
3 /* ~= Q8(0.01) Update for transition and noise region */
#endif /* MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_ */