2018-02-26 20:17:00 +01:00
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/* -----------------------------------------------------------------------------
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2013-12-31 01:01:08 +01:00
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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2018-02-26 20:17:00 +01:00
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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2013-12-31 01:01:08 +01:00
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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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
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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information and documentation.
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2. COPYRIGHT LICENSE
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2018-02-26 20:17:00 +01:00
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Redistribution and use in source and binary forms, with or without modification,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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2013-12-31 01:01:08 +01:00
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2018-02-26 20:17:00 +01:00
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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2018-02-26 20:17:00 +01:00
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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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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2018-02-26 20:17:00 +01:00
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The name of Fraunhofer may not be used to endorse or promote products derived
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from this library without prior written permission.
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2018-02-26 20:17:00 +01:00
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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2018-02-26 20:17:00 +01:00
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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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
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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2018-02-26 20:17:00 +01:00
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized by appropriate patent licenses.
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4. DISCLAIMER
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2018-02-26 20:17:00 +01:00
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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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,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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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
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this software, even if 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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2018-02-26 20:17:00 +01:00
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/**************************** PCM utility library ******************************
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Author(s): Matthias Neusinger
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2018-02-26 20:17:00 +01:00
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2013-12-31 01:01:08 +01:00
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Description: Hard limiter for clipping prevention
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*******************************************************************************/
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2018-02-26 20:17:00 +01:00
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#ifndef LIMITER_H
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#define LIMITER_H
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#include "common_fix.h"
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#include "FDK_audio.h"
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#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */
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#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */
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#define TDL_GAIN_SCALING (15) /* scaling of gain value. */
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#ifdef __cplusplus
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extern "C" {
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#endif
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2018-02-26 20:17:00 +01:00
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struct TDLimiter {
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unsigned int attack;
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FIXP_DBL attackConst, releaseConst;
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unsigned int attackMs, releaseMs, maxAttackMs;
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FIXP_DBL threshold;
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unsigned int channels, maxChannels;
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UINT sampleRate, maxSampleRate;
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FIXP_DBL cor, max;
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FIXP_DBL* maxBuf;
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FIXP_DBL* delayBuf;
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unsigned int maxBufIdx, delayBufIdx;
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FIXP_DBL smoothState0;
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FIXP_DBL minGain;
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FIXP_DBL additionalGainPrev;
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FIXP_DBL additionalGainFilterState;
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FIXP_DBL additionalGainFilterState1;
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};
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typedef enum {
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TDLIMIT_OK = 0,
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TDLIMIT_UNKNOWN = -1,
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__error_codes_start = -100,
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TDLIMIT_INVALID_HANDLE,
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TDLIMIT_INVALID_PARAMETER,
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__error_codes_end
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} TDLIMITER_ERROR;
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struct TDLimiter;
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typedef struct TDLimiter* TDLimiterPtr;
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#define PCM_LIM LONG
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#define FIXP_DBL2PCM_LIM(x) (x)
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#define PCM_LIM2FIXP_DBL(x) (x)
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#define PCM_LIM_BITS 32
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#define FIXP_PCM_LIM FIXP_DBL
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#define SAMPLE_BITS_LIM DFRACT_BITS
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/******************************************************************************
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* pcmLimiter_Reset *
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* limiter: limiter handle *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter);
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/******************************************************************************
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* pcmLimiter_Destroy *
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* limiter: limiter handle *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter);
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/******************************************************************************
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* pcmLimiter_GetDelay *
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* limiter: limiter handle *
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* returns: exact delay caused by the limiter in samples per channel *
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******************************************************************************/
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unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter);
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/******************************************************************************
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* pcmLimiter_GetMaxGainReduction *
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* limiter: limiter handle *
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* returns: maximum gain reduction in last processed block in dB *
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******************************************************************************/
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INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter);
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/******************************************************************************
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* pcmLimiter_SetNChannels *
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* limiter: limiter handle *
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* nChannels: number of channels ( <= maxChannels specified on create) *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter,
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unsigned int nChannels);
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/******************************************************************************
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* pcmLimiter_SetSampleRate *
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* limiter: limiter handle *
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* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter, UINT sampleRate);
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/******************************************************************************
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* pcmLimiter_SetAttack *
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* limiter: limiter handle *
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* attackMs: attack time in ms ( <= maxAttackMs specified on create) *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter,
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unsigned int attackMs);
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/******************************************************************************
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* pcmLimiter_SetRelease *
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* limiter: limiter handle *
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* releaseMs: release time in ms *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter,
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unsigned int releaseMs);
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/******************************************************************************
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* pcmLimiter_GetLibInfo *
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* info: pointer to an allocated and initialized LIB_INFO structure *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info);
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#ifdef __cplusplus
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}
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#endif
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2018-02-26 20:17:00 +01:00
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/******************************************************************************
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* pcmLimiter_Create *
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* maxAttackMs: maximum and initial attack/lookahead time in milliseconds *
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* releaseMs: release time in milliseconds (90% time constant) *
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* threshold: limiting threshold *
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* maxChannels: maximum and initial number of channels *
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* maxSampleRate: maximum and initial sampling rate in Hz *
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* returns: limiter handle *
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******************************************************************************/
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TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs,
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FIXP_DBL threshold, unsigned int maxChannels,
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UINT maxSampleRate);
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/******************************************************************************
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* pcmLimiter_SetThreshold *
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* limiter: limiter handle *
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* threshold: limiter threshold *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter,
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FIXP_DBL threshold);
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/******************************************************************************
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* pcmLimiter_Apply *
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* limiter: limiter handle *
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* pGain : pointer to gains to be applied to the signal before limiting, *
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* which are downscaled by TDL_GAIN_SCALING bit. *
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* These gains are delayed by gain_delay, and smoothed. *
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* Smoothing is done by a butterworth lowpass filter with a cutoff *
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* frequency which is fixed with respect to the sampling rate. *
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* It is a substitute for the smoothing due to windowing and *
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* overlap/add, if a gain is applied in frequency domain. *
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* gain_scale: pointer to scaling exponents to be applied to the signal before *
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* limiting, without delay and without smoothing *
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* gain_size: number of elements in pGain, currently restricted to 1 *
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* gain_delay: delay [samples] with which the gains in pGain shall be applied *
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* gain_delay <= nSamples *
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* samples: input/output buffer containing interleaved samples *
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* precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits *
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* nSamples: number of samples per channel *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
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INT_PCM* samplesOut, FIXP_DBL* pGain,
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const INT* gain_scale, const UINT gain_size,
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const UINT gain_delay, const UINT nSamples);
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#endif /* #ifndef LIMITER_H */
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