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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libPCMutils/include/limiter.h | |
parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip |
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.
So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.
At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.
DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libPCMutils/include/limiter.h')
-rw-r--r-- | fdk-aac/libPCMutils/include/limiter.h | 281 |
1 files changed, 281 insertions, 0 deletions
diff --git a/fdk-aac/libPCMutils/include/limiter.h b/fdk-aac/libPCMutils/include/limiter.h new file mode 100644 index 0000000..fab7226 --- /dev/null +++ b/fdk-aac/libPCMutils/include/limiter.h @@ -0,0 +1,281 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** PCM utility library ****************************** + + Author(s): Matthias Neusinger + + Description: Hard limiter for clipping prevention + +*******************************************************************************/ + +#ifndef LIMITER_H +#define LIMITER_H + +#include "common_fix.h" +#include "FDK_audio.h" + +#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */ +#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */ + +#define TDL_GAIN_SCALING (15) /* scaling of gain value. */ + +#ifdef __cplusplus +extern "C" { +#endif + +struct TDLimiter { + unsigned int attack; + FIXP_DBL attackConst, releaseConst; + unsigned int attackMs, releaseMs, maxAttackMs; + FIXP_DBL threshold; + unsigned int channels, maxChannels; + UINT sampleRate, maxSampleRate; + FIXP_DBL cor, max; + FIXP_DBL* maxBuf; + FIXP_DBL* delayBuf; + unsigned int maxBufIdx, delayBufIdx; + FIXP_DBL smoothState0; + FIXP_DBL minGain; + + FIXP_DBL additionalGainPrev; + FIXP_DBL additionalGainFilterState; + FIXP_DBL additionalGainFilterState1; +}; + +typedef enum { + TDLIMIT_OK = 0, + TDLIMIT_UNKNOWN = -1, + + __error_codes_start = -100, + + TDLIMIT_INVALID_HANDLE, + TDLIMIT_INVALID_PARAMETER, + + __error_codes_end +} TDLIMITER_ERROR; + +struct TDLimiter; +typedef struct TDLimiter* TDLimiterPtr; + +#define PCM_LIM LONG +#define FIXP_DBL2PCM_LIM(x) (x) +#define PCM_LIM2FIXP_DBL(x) (x) +#define PCM_LIM_BITS 32 +#define FIXP_PCM_LIM FIXP_DBL + +#define SAMPLE_BITS_LIM DFRACT_BITS + +/****************************************************************************** + * pcmLimiter_Reset * + * limiter: limiter handle * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter); + +/****************************************************************************** + * pcmLimiter_Destroy * + * limiter: limiter handle * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter); + +/****************************************************************************** + * pcmLimiter_GetDelay * + * limiter: limiter handle * + * returns: exact delay caused by the limiter in samples per channel * + ******************************************************************************/ +unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter); + +/****************************************************************************** + * pcmLimiter_GetMaxGainReduction * + * limiter: limiter handle * + * returns: maximum gain reduction in last processed block in dB * + ******************************************************************************/ +INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter); + +/****************************************************************************** + * pcmLimiter_SetNChannels * + * limiter: limiter handle * + * nChannels: number of channels ( <= maxChannels specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter, + unsigned int nChannels); + +/****************************************************************************** + * pcmLimiter_SetSampleRate * + * limiter: limiter handle * + * sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter, UINT sampleRate); + +/****************************************************************************** + * pcmLimiter_SetAttack * + * limiter: limiter handle * + * attackMs: attack time in ms ( <= maxAttackMs specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter, + unsigned int attackMs); + +/****************************************************************************** + * pcmLimiter_SetRelease * + * limiter: limiter handle * + * releaseMs: release time in ms * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter, + unsigned int releaseMs); + +/****************************************************************************** + * pcmLimiter_GetLibInfo * + * info: pointer to an allocated and initialized LIB_INFO structure * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info); + +#ifdef __cplusplus +} +#endif + +/****************************************************************************** + * pcmLimiter_Create * + * maxAttackMs: maximum and initial attack/lookahead time in milliseconds * + * releaseMs: release time in milliseconds (90% time constant) * + * threshold: limiting threshold * + * maxChannels: maximum and initial number of channels * + * maxSampleRate: maximum and initial sampling rate in Hz * + * returns: limiter handle * + ******************************************************************************/ +TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, + FIXP_DBL threshold, unsigned int maxChannels, + UINT maxSampleRate); + +/****************************************************************************** + * pcmLimiter_SetThreshold * + * limiter: limiter handle * + * threshold: limiter threshold * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, + FIXP_DBL threshold); + +/****************************************************************************** + * pcmLimiter_Apply * + * limiter: limiter handle * + * pGain : pointer to gains to be applied to the signal before limiting, * + * which are downscaled by TDL_GAIN_SCALING bit. * + * These gains are delayed by gain_delay, and smoothed. * + * Smoothing is done by a butterworth lowpass filter with a cutoff * + * frequency which is fixed with respect to the sampling rate. * + * It is a substitute for the smoothing due to windowing and * + * overlap/add, if a gain is applied in frequency domain. * + * gain_scale: pointer to scaling exponents to be applied to the signal before * + * limiting, without delay and without smoothing * + * gain_size: number of elements in pGain, currently restricted to 1 * + * gain_delay: delay [samples] with which the gains in pGain shall be applied * + * gain_delay <= nSamples * + * samples: input/output buffer containing interleaved samples * + * precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits * + * nSamples: number of samples per channel * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, + INT_PCM* samplesOut, FIXP_DBL* pGain, + const INT* gain_scale, const UINT gain_size, + const UINT gain_delay, const UINT nSamples); + +#endif /* #ifndef LIMITER_H */ |