mirror of https://github.com/mstorsjo/fdk-aac.git
234 lines
12 KiB
C
234 lines
12 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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<EFBFBD> Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur F<EFBFBD>rderung 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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/************************ FDK PCM postprocessor module *********************
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Author(s): Matthias Neusinger
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Description: Hard limiter for clipping prevention
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*******************************************************************************/
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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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#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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typedef enum {
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TDLIMIT_OK = 0,
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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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/******************************************************************************
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* createLimiter *
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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 createLimiter(unsigned int maxAttackMs,
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unsigned int releaseMs,
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INT_PCM threshold,
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unsigned int maxChannels,
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unsigned int maxSampleRate);
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/******************************************************************************
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* resetLimiter *
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* limiter: limiter handle *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter);
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/******************************************************************************
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* destroyLimiter *
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* limiter: limiter handle *
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* returns: error code *
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******************************************************************************/
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TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter);
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/******************************************************************************
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* applyLimiter *
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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 applyLimiter(TDLimiterPtr limiter,
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INT_PCM* samples,
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FIXP_DBL* pGain,
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const INT* gain_scale,
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const UINT gain_size,
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const UINT gain_delay,
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const UINT nSamples);
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/******************************************************************************
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* getLimiterDelay *
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* limiter: limiter handle *
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* returns: exact delay caused by the limiter in samples *
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******************************************************************************/
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unsigned int getLimiterDelay(TDLimiterPtr limiter);
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/******************************************************************************
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* setLimiterNChannels *
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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 setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels);
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/******************************************************************************
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* setLimiterSampleRate *
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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 setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate);
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/******************************************************************************
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* setLimiterAttack *
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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 setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs);
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/******************************************************************************
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* setLimiterRelease *
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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 setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs);
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/******************************************************************************
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* setLimiterThreshold *
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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 setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold);
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#ifdef __cplusplus
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}
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#endif
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#endif //#ifndef _LIMITER_H_
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