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authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libPCMutils/include/limiter.h
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
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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.
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+/* -----------------------------------------------------------------------------
+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 */