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https://github.com/mstorsjo/fdk-aac.git
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* AAC-Encoder - AAC-ELD core encoder audio quality tuning. Update tuning tables, configure bitreservoir size and adapt afterburner iteration value. Modified file(s): libAACenc/src/aacenc.h libAACenc/src/aacenc_lib.cpp libAACenc/src/adj_thr.cpp libAACenc/src/adj_thr.h libAACenc/src/adj_thr_data.h libAACenc/src/bandwidth.cpp libAACenc/src/pnsparam.cpp libAACenc/src/qc_main.cpp - Introduze dead zone quantizer for ELD to improve audio quality at certain configurations. Modified file(s): libAACenc/src/aacenc_lib.cpp libAACenc/src/adj_thr.cpp libAACenc/src/adj_thr.h libAACenc/src/qc_data.h libAACenc/src/qc_main.cpp libAACenc/src/quantize.cpp libAACenc/src/quantize.h libAACenc/src/sf_estim.cpp libAACenc/src/sf_estim.h - Revise TNS module to improve ELD audio quality. - Use new window function and separate prediction gain according TNS filters. - Add missing memory initilization to TNS configuration. Modified file(s): libAACenc/src/aacenc_lib.cpp libAACenc/src/aacenc_tns.cpp libAACenc/src/aacenc_tns.h libAACenc/src/psy_main.cpp libAACenc/src/tns_func.h * SBR-Encoder - Revise frequency resolution calculation and handle differently depending on number of envelopes and split frames decision. - Add and adjust ELD SBR tuning tables. Modified file(s): libSBRenc/include/sbr_encoder.h libSBRenc/src/bit_sbr.h libSBRenc/src/env_est.cpp libSBRenc/src/fram_gen.cpp libSBRenc/src/fram_gen.h libSBRenc/src/mh_det.cpp libSBRenc/src/sbr_def.h libSBRenc/src/sbr_encoder.cpp libSBRenc/src/sbr_rom.cpp libSBRenc/src/tran_det.cpp - Replace ELD transient detector with fast implementation. Modified file(s): libSBRenc/src/env_est.cpp libSBRenc/src/env_est.h libSBRenc/src/fram_gen.cpp libSBRenc/src/sbr_def.h libSBRenc/src/sbr_encoder.cpp libSBRenc/src/tran_det.cpp libSBRenc/src/tran_det.h * FDK-Library - Introduce generic compare function in tools library. Modified file(s): libFDK/include/fixpoint_math.h libFDK/src/FDK_core.cpp * SBR-Encoder - Revise ELD frame splitter to improve bit distribution. Modified file(s): libSBRenc/include/sbr_encoder.h libSBRenc/src/bit_sbr.h libSBRenc/src/env_est.cpp libSBRenc/src/fram_gen.cpp libSBRenc/src/fram_gen.h libSBRenc/src/sbr_encoder.cpp libSBRenc/src/tran_det.cpp libSBRenc/src/tran_det.h - Configure amplitude resolution according the tonality of the audio signal. Modified file(s): libSBRenc/include/sbr_encoder.h libSBRenc/src/bit_sbr.h libSBRenc/src/env_est.cpp libSBRenc/src/nf_est.cpp libSBRenc/src/nf_est.h libSBRenc/src/sbr_def.h libSBRenc/src/sbr_encoder.cpp libSBRenc/src/ton_corr.cpp libSBRenc/src/ton_corr.h libSBRenc/src/tran_det.cpp libSBRenc/src/tran_det.h Change-Id: Ie0672b989a06ee63b50240616b8d1d4b790b6cb2
585 lines
22 KiB
C++
585 lines
22 KiB
C++
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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#include "nf_est.h"
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#include "sbr_misc.h"
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#include "genericStds.h"
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/* smoothFilter[4] = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */
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static const FIXP_DBL smoothFilter[4] = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 };
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/* static const INT smoothFilterLength = 4; */
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static const FIXP_DBL QuantOffset = (INT)0xfc000000; /* ld64(0.25) */
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#ifndef min
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#define min(a,b) ( a < b ? a:b)
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#endif
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#ifndef max
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#define max(a,b) ( a > b ? a:b)
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#endif
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#define NOISE_FLOOR_OFFSET_SCALING (4)
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/**************************************************************************/
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/*!
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\brief The function applies smoothing to the noise levels.
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\return none
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*/
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/**************************************************************************/
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static void
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smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels, /*!< pointer to noise-floor levels.*/
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INT nEnvelopes, /*!< Number of noise floor envelopes.*/
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INT noNoiseBands, /*!< Number of noise bands for every noise floor envelope. */
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FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */
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const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */
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INT transientFlag) /*!< flag indicating if a transient is present*/
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{
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INT i,band,env;
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FIXP_DBL accu;
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for(env = 0; env < nEnvelopes; env++){
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if(transientFlag){
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for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
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FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
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}
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}
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else {
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for (i = 1; i < NF_SMOOTHING_LENGTH; i++){
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FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL));
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}
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FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
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}
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for (band = 0; band < noNoiseBands; band++){
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accu = FL2FXCONST_DBL(0.0f);
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for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
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accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]);
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}
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FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
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NoiseLevels[band+ env*noNoiseBands] = accu<<1;
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}
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}
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}
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/**************************************************************************/
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/*!
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\brief Does the noise floor level estiamtion.
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The noiseLevel samples are scaled by the factor 0.25
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\return none
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*/
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/**************************************************************************/
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static void
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qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel, /*!< Pointer to vector to store the noise levels in.*/
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FIXP_DBL ** quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */
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SCHAR *indexVector, /*!< Index vector to obtain the patched data. */
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INT startIndex, /*!< Start index. */
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INT stopIndex, /*!< Stop index. */
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INT startChannel, /*!< Start channel of the current noise floor band.*/
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INT stopChannel, /*!< Stop channel of the current noise floor band. */
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FIXP_DBL ana_max_level, /*!< Maximum level of the adaptive noise.*/
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FIXP_DBL noiseFloorOffset, /*!< Noise floor offset. */
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INT missingHarmonicFlag, /*!< Flag indicating if a strong tonal component is missing.*/
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FIXP_DBL weightFac, /*!< Weightening factor for the difference between orig and sbr. */
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INVF_MODE diffThres, /*!< Threshold value to control the inverse filtering decision.*/
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INVF_MODE inverseFilteringLevel) /*!< Inverse filtering level of the current band.*/
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{
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INT scale, l, k;
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FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff;
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FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex);
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FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel);
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FIXP_DBL accu;
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/*
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Calculate the mean value, over the current time segment, for the original, the HFR
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and the difference, over all channels in the current frequency range.
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*/
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if(missingHarmonicFlag == 1){
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for(l = startChannel; l < stopChannel;l++){
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/* tonalityOrig */
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accu = FL2FXCONST_DBL(0.0f);
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for(k = startIndex ; k < stopIndex; k++){
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accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
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}
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meanOrig = fixMax(meanOrig,(accu<<1));
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/* tonalitySbr */
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accu = FL2FXCONST_DBL(0.0f);
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for(k = startIndex ; k < stopIndex; k++){
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accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
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}
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meanSbr = fixMax(meanSbr,(accu<<1));
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}
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}
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else{
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for(l = startChannel; l < stopChannel;l++){
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/* tonalityOrig */
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accu = FL2FXCONST_DBL(0.0f);
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for(k = startIndex ; k < stopIndex; k++){
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accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
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}
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meanOrig += fMult((accu<<1), invChannel);
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/* tonalitySbr */
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accu = FL2FXCONST_DBL(0.0f);
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for(k = startIndex ; k < stopIndex; k++){
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accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
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}
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meanSbr += fMult((accu<<1), invChannel);
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}
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}
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/* Small fix to avoid noise during silent passages.*/
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if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) &&
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meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) )
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{
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meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
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meanSbr = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
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}
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meanOrig = fixMax(meanOrig,RELAXATION);
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meanSbr = fixMax(meanSbr,RELAXATION);
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if (missingHarmonicFlag == 1 ||
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inverseFilteringLevel == INVF_MID_LEVEL ||
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inverseFilteringLevel == INVF_LOW_LEVEL ||
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inverseFilteringLevel == INVF_OFF ||
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inverseFilteringLevel <= diffThres)
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{
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diff = RELAXATION;
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}
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else {
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accu = fDivNorm(meanSbr, meanOrig, &scale);
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diff = fixMax( RELAXATION,
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fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ;
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}
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/*
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* noise Level is now a positive value, i.e.
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* the more harmonic the signal is the higher noise level,
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* this makes no sense so we change the sign.
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*********************************************************/
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accu = fDivNorm(diff, meanOrig, &scale);
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scale -= 2;
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if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) {
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*noiseLevel = (FIXP_DBL)MAXVAL_DBL;
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}
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else {
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*noiseLevel = scaleValue(accu, scale);
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}
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/*
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* Add a noise floor offset to compensate for bias in the detector
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*****************************************************************/
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if(!missingHarmonicFlag) {
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*noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING;
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}
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/*
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* check to see that we don't exceed the maximum allowed level
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**************************************************************/
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*noiseLevel = fixMin(*noiseLevel, ana_max_level); /* ana_max_level is scaled with factor 0.25 */
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}
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/**************************************************************************/
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/*!
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\brief Does the noise floor level estiamtion.
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The function calls the Noisefloor estimation function
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for the time segments decided based upon the transient
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information. The block is always divided into one or two segments.
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\return none
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*/
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/**************************************************************************/
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void
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FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
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const SBR_FRAME_INFO *frame_info, /*!< Time frequency grid of the current frame. */
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FIXP_DBL *noiseLevels, /*!< Pointer to vector to store the noise levels in.*/
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FIXP_DBL **quotaMatrixOrig, /*!< Matrix holding the quota values of the original. */
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SCHAR *indexVector, /*!< Index vector to obtain the patched data. */
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INT missingHarmonicsFlag, /*!< Flag indicating if a strong tonal component will be missing. */
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INT startIndex, /*!< Start index. */
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UINT numberOfEstimatesPerFrame, /*!< The number of tonality estimates per frame. */
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int transientFrame, /*!< A flag indicating if a transient is present. */
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INVF_MODE* pInvFiltLevels, /*!< Pointer to the vector holding the inverse filtering levels. */
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UINT sbrSyntaxFlags
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)
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{
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INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band;
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INT noNoiseBands = h_sbrNoiseFloorEstimate->noNoiseBands;
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INT *freqBandTable = h_sbrNoiseFloorEstimate->freqBandTableQmf;
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nNoiseEnvelopes = frame_info->nNoiseEnvelopes;
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if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
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nNoiseEnvelopes = 1;
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startPos[0] = startIndex;
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stopPos[0] = startIndex + min(numberOfEstimatesPerFrame,2);
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} else
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if(nNoiseEnvelopes == 1){
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startPos[0] = startIndex;
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stopPos[0] = startIndex + 2;
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}
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else{
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startPos[0] = startIndex;
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stopPos[0] = startIndex + 1;
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startPos[1] = startIndex + 1;
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stopPos[1] = startIndex + 2;
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}
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/*
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* Estimate the noise floor.
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**************************************/
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for(env = 0; env < nNoiseEnvelopes; env++){
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for(band = 0; band < noNoiseBands; band++){
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FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
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qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands],
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quotaMatrixOrig,
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indexVector,
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startPos[env],
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stopPos[env],
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freqBandTable[band],
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freqBandTable[band+1],
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h_sbrNoiseFloorEstimate->ana_max_level,
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h_sbrNoiseFloorEstimate->noiseFloorOffset[band],
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missingHarmonicsFlag,
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h_sbrNoiseFloorEstimate->weightFac,
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h_sbrNoiseFloorEstimate->diffThres,
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pInvFiltLevels[band]);
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}
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}
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/*
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* Smoothing of the values.
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**************************/
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smoothingOfNoiseLevels(noiseLevels,
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nNoiseEnvelopes,
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h_sbrNoiseFloorEstimate->noNoiseBands,
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h_sbrNoiseFloorEstimate->prevNoiseLevels,
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h_sbrNoiseFloorEstimate->smoothFilter,
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transientFrame);
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/* quantisation*/
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for(env = 0; env < nNoiseEnvelopes; env++){
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for(band = 0; band < noNoiseBands; band++){
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FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
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noiseLevels[band + env*noNoiseBands] =
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(FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset;
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}
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}
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}
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/**************************************************************************/
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/*!
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\brief
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\return errorCode, noError if successful
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*/
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/**************************************************************************/
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static INT
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downSampleLoRes(INT *v_result, /*!< */
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INT num_result, /*!< */
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const UCHAR *freqBandTableRef,/*!< */
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INT num_Ref) /*!< */
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{
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INT step;
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INT i,j;
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INT org_length,result_length;
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INT v_index[MAX_FREQ_COEFFS/2];
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/* init */
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org_length=num_Ref;
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result_length=num_result;
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v_index[0]=0; /* Always use left border */
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i=0;
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while(org_length > 0) /* Create downsample vector */
|
|
{
|
|
i++;
|
|
step=org_length/result_length; /* floor; */
|
|
org_length=org_length - step;
|
|
result_length--;
|
|
v_index[i]=v_index[i-1]+step;
|
|
}
|
|
|
|
if(i != num_result ) /* Should never happen */
|
|
return (1);/* error downsampling */
|
|
|
|
for(j=0;j<=i;j++) /* Use downsample vector to index LoResolution vector. */
|
|
{
|
|
v_result[j]=freqBandTableRef[v_index[j]];
|
|
}
|
|
|
|
return (0);
|
|
}
|
|
|
|
/**************************************************************************/
|
|
/*!
|
|
\brief Initialize an instance of the noise floor level estimation module.
|
|
|
|
|
|
\return errorCode, noError if successful
|
|
|
|
*/
|
|
/**************************************************************************/
|
|
INT
|
|
FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
|
|
INT ana_max_level, /*!< Maximum level of the adaptive noise. */
|
|
const UCHAR *freqBandTable, /*!< Frequany band table. */
|
|
INT nSfb, /*!< Number of frequency bands. */
|
|
INT noiseBands, /*!< Number of noise bands per octave. */
|
|
INT noiseFloorOffset, /*!< Noise floor offset. */
|
|
INT timeSlots, /*!< Number of time slots in a frame. */
|
|
UINT useSpeechConfig /*!< Flag: adapt tuning parameters according to speech */
|
|
)
|
|
{
|
|
INT i, qexp, qtmp;
|
|
FIXP_DBL tmp, exp;
|
|
|
|
FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE));
|
|
|
|
h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter;
|
|
if (useSpeechConfig) {
|
|
h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL;
|
|
h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL;
|
|
}
|
|
else {
|
|
h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f);
|
|
h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL;
|
|
}
|
|
|
|
h_sbrNoiseFloorEstimate->timeSlots = timeSlots;
|
|
h_sbrNoiseFloorEstimate->noiseBands = noiseBands;
|
|
|
|
/* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25 */
|
|
switch(ana_max_level)
|
|
{
|
|
case 6:
|
|
h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
|
|
break;
|
|
case 3:
|
|
h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5);
|
|
break;
|
|
case -3:
|
|
h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125);
|
|
break;
|
|
default:
|
|
/* Should not enter here */
|
|
h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
|
|
break;
|
|
}
|
|
|
|
/*
|
|
calculate number of noise bands and allocate
|
|
*/
|
|
if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb))
|
|
return(1);
|
|
|
|
if(noiseFloorOffset == 0) {
|
|
tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING;
|
|
}
|
|
else {
|
|
/* noiseFloorOffset has to be smaller than 12, because
|
|
the result of the calculation below must be smaller than 1:
|
|
(2^(noiseFloorOffset/3))*2^4<1 */
|
|
FDK_ASSERT(noiseFloorOffset<12);
|
|
|
|
/* Assumes the noise floor offset in tuning table are in q31 */
|
|
/* Change the qformat here when non-zero values would be filled */
|
|
exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
|
|
tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
|
|
tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING);
|
|
}
|
|
|
|
for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) {
|
|
h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp;
|
|
}
|
|
|
|
return (0);
|
|
}
|
|
|
|
/**************************************************************************/
|
|
/*!
|
|
\brief Resets the current instance of the noise floor estiamtion
|
|
module.
|
|
|
|
|
|
\return errorCode, noError if successful
|
|
|
|
*/
|
|
/**************************************************************************/
|
|
INT
|
|
FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
|
|
const UCHAR *freqBandTable, /*!< Frequany band table. */
|
|
INT nSfb) /*!< Number of bands in the frequency band table. */
|
|
{
|
|
INT k2,kx;
|
|
|
|
/*
|
|
* Calculate number of noise bands
|
|
***********************************/
|
|
k2=freqBandTable[nSfb];
|
|
kx=freqBandTable[0];
|
|
if(h_sbrNoiseFloorEstimate->noiseBands == 0){
|
|
h_sbrNoiseFloorEstimate->noNoiseBands = 1;
|
|
}
|
|
else{
|
|
/*
|
|
* Calculate number of noise bands 1,2 or 3 bands/octave
|
|
********************************************************/
|
|
FIXP_DBL tmp, ratio, lg2;
|
|
INT ratio_e, qlg2, nNoiseBands;
|
|
|
|
ratio = fDivNorm(k2, kx, &ratio_e);
|
|
lg2 = fLog2(ratio, ratio_e, &qlg2);
|
|
tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2);
|
|
tmp = scaleValue(tmp, qlg2-23);
|
|
|
|
nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
|
|
|
|
|
|
if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
|
|
nNoiseBands = MAX_NUM_NOISE_COEFFS;
|
|
}
|
|
|
|
if( nNoiseBands == 0 ) {
|
|
nNoiseBands = 1;
|
|
}
|
|
|
|
h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
|
|
|
|
}
|
|
|
|
|
|
return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf,
|
|
h_sbrNoiseFloorEstimate->noNoiseBands,
|
|
freqBandTable,nSfb));
|
|
}
|
|
|
|
/**************************************************************************/
|
|
/*!
|
|
\brief Deletes the current instancce of the noise floor level
|
|
estimation module.
|
|
|
|
|
|
\return none
|
|
|
|
*/
|
|
/**************************************************************************/
|
|
void
|
|
FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate) /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
|
|
{
|
|
|
|
if (h_sbrNoiseFloorEstimate) {
|
|
/*
|
|
nothing to do
|
|
*/
|
|
}
|
|
}
|