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Bug: 71430241
Test: CTS DecoderTest and DecoderTestAacDrc

original-Change-Id: Iaa20f749b8a04d553b20247cfe1a8930ebbabe30

Apply clang-format also on header files.

original-Change-Id: I14de1ef16bbc79ec0283e745f98356a10efeb2e4

Fixes for MPEG-D DRC

original-Change-Id: If1de2d74bbbac84b3f67de3b88b83f6a23b8a15c

Catch unsupported tw_mdct at an early stage

original-Change-Id: Ied9dd00d754162a0e3ca1ae3e6b854315d818afe

Fixing PVC transition frames

original-Change-Id: Ib75725abe39252806c32d71176308f2c03547a4e

Move qmf bands sanity check

original-Change-Id: Iab540c3013c174d9490d2ae100a4576f51d8dbc4

Initialize scaling variable

original-Change-Id: I3c4087101b70e998c71c1689b122b0d7762e0f9e

Add 16 qmf band configuration to getSlotNrgHQ()

original-Change-Id: I49a5d30f703a1b126ff163df9656db2540df21f1

Always apply byte alignment at the end of the AudioMuxElement

original-Change-Id: I42d560287506d65d4c3de8bfe3eb9a4ebeb4efc7

Setup SBR element only if no parse error exists

original-Change-Id: I1915b73704bc80ab882b9173d6bec59cbd073676

Additional array index check in HCR

original-Change-Id: I18cc6e501ea683b5009f1bbee26de8ddd04d8267

Fix fade-in index selection in concealment module

original-Change-Id: Ibf802ed6ed8c05e9257e1f3b6d0ac1162e9b81c1

Enable explicit backward compatible parser for AAC_LD

original-Change-Id: I27e9c678dcb5d40ed760a6d1e06609563d02482d

Skip spatial specific config in explicit backward compatible ASC

original-Change-Id: Iff7cc365561319e886090cedf30533f562ea4d6e

Update flags description in decoder API

original-Change-Id: I9a5b4f8da76bb652f5580cbd3ba9760425c43830

Add QMF domain reset function

original-Change-Id: I4f89a8a2c0277d18103380134e4ed86996e9d8d6

DRC upgrade v2.1.0

original-Change-Id: I5731c0540139dab220094cd978ef42099fc45b74

Fix integer overflow in sqrtFixp_lookup()

original-Change-Id: I429a6f0d19aa2cc957e0f181066f0ca73968c914

Fix integer overflow in invSqrtNorm2()

original-Change-Id: I84de5cbf9fb3adeb611db203fe492fabf4eb6155

Fix integer overflow in GenerateRandomVector()

original-Change-Id: I3118a641008bd9484d479e5b0b1ee2b5d7d44d74

Fix integer overflow in adjustTimeSlot_EldGrid()

original-Change-Id: I29d503c247c5c8282349b79df940416a512fb9d5

Fix integer overflow in FDKsbrEnc_codeEnvelope()

original-Change-Id: I6b34b61ebb9d525b0c651ed08de2befc1f801449

Follow-up on: Fix integer overflow in adjustTimeSlot_EldGrid()

original-Change-Id: I6f8f578cc7089e5eb7c7b93e580b72ca35ad689a

Fix integer overflow in get_pk_v2()

original-Change-Id: I63375bed40d45867f6eeaa72b20b1f33e815938c

Fix integer overflow in Syn_filt_zero()

original-Change-Id: Ie0c02fdfbe03988f9d3b20d10cd9fe4c002d1279

Fix integer overflow in CFac_CalcFacSignal()

original-Change-Id: Id2d767c40066c591b51768e978eb8af3b803f0c5

Fix integer overflow in FDKaacEnc_FDKaacEnc_calcPeNoAH()

original-Change-Id: Idcbd0f4a51ae2550ed106aa6f3d678d1f9724841

Fix integer overflow in sbrDecoder_calculateGainVec()

original-Change-Id: I7081bcbe29c5cede9821b38d93de07c7add2d507

Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I4a95ddc18de150102352d4a1845f06094764c881

Fix integer overflow in Pred_Lt4()

original-Change-Id: I4dbd012b2de7d07c3e70a47b92e3bfae8dbc750a

Fix integer overflow in FDKsbrEnc_InitSbrFastTransientDetector()

original-Change-Id: I788cbec1a4a00f44c2f3a72ad7a4afa219807d04

Fix unsigned integer overflow in FDKaacEnc_WriteBitstream()

original-Change-Id: I68fc75166e7d2cd5cd45b18dbe3d8c2a92f1822a

Fix unsigned integer overflow in FDK_MetadataEnc_Init()

original-Change-Id: Ie8d025f9bcdb2442c704bd196e61065c03c10af4

Fix overflow in pseudo random number generators

original-Change-Id: I3e2551ee01356297ca14e3788436ede80bd5513c

Fix unsigned integer overflow in sbrDecoder_Parse()

original-Change-Id: I3f231b2f437e9c37db4d5b964164686710eee971

Fix unsigned integer overflow in longsub()

original-Change-Id: I73c2bc50415cac26f1f5a29e125bbe75f9180a6e

Fix unsigned integer overflow in CAacDecoder_DecodeFrame()

original-Change-Id: Ifce2db4b1454b46fa5f887e9d383f1cc43b291e4

Fix overflow at CLpdChannelStream_Read()

original-Change-Id: Idb9d822ce3a4272e4794b643644f5434e2d4bf3f

Fix unsigned integer overflow in Hcr_State_BODY_SIGN_ESC__ESC_WORD()

original-Change-Id: I1ccf77c0015684b85534c5eb97162740a870b71c

Fix unsigned integer overflow in UsacConfig_Parse()

original-Change-Id: Ie6d27f84b6ae7eef092ecbff4447941c77864d9f

Fix unsigned integer overflow in aacDecoder_drcParse()

original-Change-Id: I713f28e883eea3d70b6fa56a7b8f8c22bcf66ca0

Fix unsigned integer overflow in aacDecoder_drcReadCompression()

original-Change-Id: Ia34dfeb88c4705c558bce34314f584965cafcf7a

Fix unsigned integer overflow in CDataStreamElement_Read()

original-Change-Id: Iae896cc1d11f0a893d21be6aa90bd3e60a2c25f0

Fix unsigned integer overflow in transportDec_AdjustEndOfAccessUnit()

original-Change-Id: I64cf29a153ee784bb4a16fdc088baabebc0007dc

Fix unsigned integer overflow in transportDec_GetAuBitsRemaining()

original-Change-Id: I975b3420faa9c16a041874ba0db82e92035962e4

Fix unsigned integer overflow in extractExtendedData()

original-Change-Id: I2a59eb09e2053cfb58dfb75fcecfad6b85a80a8f

Fix signed integer overflow in CAacDecoder_ExtPayloadParse()

original-Change-Id: I4ad5ca4e3b83b5d964f1c2f8c5e7b17c477c7929

Fix unsigned integer overflow in CAacDecoder_DecodeFrame()

original-Change-Id: I29a39df77d45c52a0c9c5c83c1ba81f8d0f25090

Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I8fb194ffc073a3432a380845be71036a272d388f

Fix signed integer overflow in _interpolateDrcGain()

original-Change-Id: I879ec9ab14005069a7c47faf80e8bc6e03d22e60

Fix unsigned integer overflow in FDKreadBits()

original-Change-Id: I1f47a6a8037ff70375aa8844947d5681bb4287ad

Fix unsigned integer overflow in FDKbyteAlign()

original-Change-Id: Id5f3a11a0c9e50fc6f76ed6c572dbd4e9f2af766

Fix unsigned integer overflow in FDK_get32()

original-Change-Id: I9d33b8e97e3d38cbb80629cb859266ca0acdce96

Fix unsigned integer overflow in FDK_pushBack()

original-Change-Id: Ic87f899bc8c6acf7a377a8ca7f3ba74c3a1e1c19

Fix unsigned integer overflow in FDK_pushForward()

original-Change-Id: I3b754382f6776a34be1602e66694ede8e0b8effc

Fix unsigned integer overflow in ReadPsData()

original-Change-Id: I25361664ba8139e32bbbef2ca8c106a606ce9c37

Fix signed integer overflow in E_UTIL_residu()

original-Change-Id: I8c3abd1f437ee869caa8fb5903ce7d3d641b6aad

REVERT: Follow-up on: Integer overflow in CLpc_SynthesisLattice().

original-Change-Id: I3d340099acb0414795c8dfbe6362bc0a8f045f9b

Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I4aedb8b3a187064e9f4d985175aa55bb99cc7590

Follow-up on: Fix unsigned integer overflow in aacDecoder_drcParse()

original-Change-Id: I2aa2e13916213bf52a67e8b0518e7bf7e57fb37d

Fix integer overflow in acelp

original-Change-Id: Ie6390c136d84055f8b728aefbe4ebef6e029dc77

Fix unsigned integer overflow in aacDecoder_UpdateBitStreamCounters()

original-Change-Id: I391ffd97ddb0b2c184cba76139bfb356a3b4d2e2

Adjust concealment default settings

original-Change-Id: I6a95db935a327c47df348030bcceafcb29f54b21

Saturate estimatedStartPos

original-Change-Id: I27be2085e0ae83ec9501409f65e003f6bcba1ab6

Negative shift exponent in _interpolateDrcGain()

original-Change-Id: I18edb26b26d002aafd5e633d4914960f7a359c29

Negative shift exponent in calculateICC()

original-Change-Id: I3dcd2ae98d2eb70ee0d59750863cbb2a6f4f8aba

Too large shift exponent in FDK_put()

original-Change-Id: Ib7d9aaa434d2d8de4a13b720ca0464b31ca9b671

Too large shift exponent in CalcInvLdData()

original-Change-Id: I43e6e78d4cd12daeb1dcd5d82d1798bdc2550262

Member access within null pointer of type SBR_CHANNEL

original-Change-Id: Idc5e4ea8997810376d2f36bbdf628923b135b097

Member access within null pointer of type CpePersistentData

original-Change-Id: Ib6c91cb0d37882768e5baf63324e429589de0d9d

Member access within null pointer FDKaacEnc_psyMain()

original-Change-Id: I7729b7f4479970531d9dc823abff63ca52e01997

Member access within null pointer FDKaacEnc_GetPnsParam()

original-Change-Id: I9aa3b9f3456ae2e0f7483dbd5b3dde95fc62da39

Member access within null pointer FDKsbrEnc_EnvEncodeFrame()

original-Change-Id: I67936f90ea714e90b3e81bc0dd1472cc713eb23a

Add HCR sanity check

original-Change-Id: I6c1d9732ebcf6af12f50b7641400752f74be39f7

Fix memory issue for HBE edge case with 8:3 SBR

original-Change-Id: I11ea58a61e69fbe8bf75034b640baee3011e63e9

Additional SBR parametrization sanity check for ELD

original-Change-Id: Ie26026fbfe174c2c7b3691f6218b5ce63e322140

Add MPEG-D DRC channel layout check

original-Change-Id: Iea70a74f171b227cce636a9eac4ba662777a2f72

Additional out-of-bounds checks in MPEG-D DRC

original-Change-Id: Ife4a8c3452c6fde8a0a09e941154a39a769777d4

Change-Id: Ic63cb2f628720f54fe9b572b0cb528e2599c624e
This commit is contained in:
Fraunhofer IIS FDK
2018-02-26 20:17:00 +01:00
committed by Jean-Michel Trivi
parent 6288a1e34c
commit 6cfabd3536
450 changed files with 164126 additions and 83284 deletions
Download Patch File Download Diff File Expand all files Collapse all files

336
libPCMutils/include/limiter.h
View File

@@ -1,74 +1,85 @@
/* -----------------------------------------------------------------------------------------------------------
/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
<EFBFBD> Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur F<EFBFBD>rderung der angewandten Forschung e.V.
All rights reserved.
© 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.
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.
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.
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.
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:
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 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
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.
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.
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."
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.
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.
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.
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
@@ -79,34 +90,53 @@ Am Wolfsmantel 33
www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------------------------------------- */
----------------------------------------------------------------------------- */
/************************ FDK PCM postprocessor module *********************
/**************************** PCM utility library ******************************
Author(s): Matthias Neusinger
Description: Hard limiter for clipping prevention
*******************************************************************************/
#ifndef _LIMITER_H_
#define _LIMITER_H_
#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. */
#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,
@@ -119,115 +149,133 @@ typedef enum {
struct TDLimiter;
typedef struct TDLimiter* TDLimiterPtr;
/******************************************************************************
* createLimiter *
* 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 createLimiter(unsigned int maxAttackMs,
unsigned int releaseMs,
INT_PCM threshold,
unsigned int maxChannels,
unsigned int maxSampleRate);
#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
/******************************************************************************
* resetLimiter *
* limiter: limiter handle *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter);
* pcmLimiter_Reset *
* limiter: limiter handle *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter);
/******************************************************************************
* destroyLimiter *
* limiter: limiter handle *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter);
* pcmLimiter_Destroy *
* limiter: limiter handle *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter);
/******************************************************************************
* applyLimiter *
* 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 applyLimiter(TDLimiterPtr limiter,
INT_PCM* samples,
FIXP_DBL* pGain,
const INT* gain_scale,
const UINT gain_size,
const UINT gain_delay,
const UINT nSamples);
* pcmLimiter_GetDelay *
* limiter: limiter handle *
* returns: exact delay caused by the limiter in samples per channel *
******************************************************************************/
unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter);
/******************************************************************************
* getLimiterDelay *
* limiter: limiter handle *
* returns: exact delay caused by the limiter in samples *
******************************************************************************/
unsigned int getLimiterDelay(TDLimiterPtr limiter);
* pcmLimiter_GetMaxGainReduction *
* limiter: limiter handle *
* returns: maximum gain reduction in last processed block in dB *
******************************************************************************/
INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter);
/******************************************************************************
* setLimiterNChannels *
* limiter: limiter handle *
* nChannels: number of channels ( <= maxChannels specified on create) *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels);
* 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);
/******************************************************************************
* setLimiterSampleRate *
* limiter: limiter handle *
* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate);
* 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);
/******************************************************************************
* setLimiterAttack *
* limiter: limiter handle *
* attackMs: attack time in ms ( <= maxAttackMs specified on create) *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs);
* 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);
/******************************************************************************
* setLimiterRelease *
* limiter: limiter handle *
* releaseMs: release time in ms *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs);
* pcmLimiter_SetRelease *
* limiter: limiter handle *
* releaseMs: release time in ms *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter,
unsigned int releaseMs);
/******************************************************************************
* setLimiterThreshold *
* limiter: limiter handle *
* threshold: limiter threshold *
* returns: error code *
******************************************************************************/
TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold);
* 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);
#endif //#ifndef _LIMITER_H_
/******************************************************************************
* 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 */

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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): Alfonso Pino Garcia
Description: Functions that perform (de)interleaving combined with format
change
*******************************************************************************/
#if !defined(PCM_UTILS_H)
#define PCM_UTILS_H
#include "common_fix.h"
void FDK_interleave(const FIXP_DBL *RESTRICT pIn, LONG *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_interleave(const FIXP_DBL *RESTRICT pIn, SHORT *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_interleave(const FIXP_SGL *RESTRICT pIn, SHORT *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_deinterleave(const LONG *RESTRICT pIn, SHORT *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_deinterleave(const LONG *RESTRICT pIn, LONG *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_deinterleave(const SHORT *RESTRICT pIn, SHORT *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
void FDK_deinterleave(const SHORT *RESTRICT pIn, LONG *RESTRICT pOut,
const UINT channels, const UINT frameSize,
const UINT length);
#endif /* !defined(PCM_UTILS_H) */

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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): Christian Griebel
Description:
*******************************************************************************/
/**
* \file pcmdmx_lib.h
* \brief FDK PCM audio mixdown library interface header file.
\page INTRO Introduction
\section SCOPE Scope
This document describes the high-level application interface and usage of the
FDK PCM audio mixdown module library developed by the Fraunhofer Institute for
Integrated Circuits (IIS). Depending on the library configuration, the module
can manipulate the number of audio channels of a given PCM signal. It can
create for example a two channel stereo audio signal from a given multi-channel
configuration (e.g. 5.1 channels).
\page ABBREV List of abbreviations
\li \b AAC - Advanced Audio Coding\n
Is an audio coding standard for lossy digital audio compression standardized
by ISO and IEC, as part of the MPEG-2 (ISO/IEC 13818-7:2006) and MPEG-4
(ISO/IEC 14496-3:2009) specifications.
\li \b DSE - Data Stream Element\n
A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream
standardized in ISO/IEC 14496-3:2009. It can convey any kind of data associated
to one program.
\li \b PCE - Program Config Element\n
A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream
standardized in ISO/IEC 14496-3:2009 that can define the stream configuration
for a single program. In addition it can comprise simple downmix meta data.
*/
#ifndef PCMDMX_LIB_H
#define PCMDMX_LIB_H
#include "machine_type.h"
#include "common_fix.h"
#include "FDK_audio.h"
#include "FDK_bitstream.h"
/**
* \enum PCMDMX_ERROR
*
* Error codes that can be returned by module interface functions.
*/
typedef enum {
PCMDMX_OK = 0x0, /*!< No error happened. */
PCMDMX_UNSUPPORTED =
0x1, /*!< The requested feature/service is unavailable. This can
occur if the module was built for a wrong configuration. */
pcm_dmx_fatal_error_start,
PCMDMX_OUT_OF_MEMORY, /*!< Not enough memory to set up an instance of the
module. */
pcm_dmx_fatal_error_end,
PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */
PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */
PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not
supported and thus no processing was performed.
*/
PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */
PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */
PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most
probably the value ist out of range.
*/
PCMDMX_CORRUPT_ANC_DATA, /*!< The read ancillary data was corrupt. */
PCMDMX_OUTPUT_BUFFER_TOO_SMALL /*!< The size of pcm output buffer is too
small. */
} PCMDMX_ERROR;
/** Macro to identify fatal errors. */
#define PCMDMX_IS_FATAL_ERROR(err) \
((((err) >= pcm_dmx_fatal_error_start) && \
((err) <= pcm_dmx_fatal_error_end)) \
? 1 \
: 0)
/**
* \enum PCMDMX_PARAM
*
* Modules dynamic runtime parameters that can be handed to function
* pcmDmx_SetParam() and pcmDmx_GetParam().
*/
typedef enum {
DMX_PROFILE_SETTING =
0x01, /*!< Defines which equations, coefficients and default/
fallback values used for downmixing. See
::DMX_PROFILE_TYPE type for details. */
DMX_BS_DATA_EXPIRY_FRAME =
0x10, /*!< The number of frames without new metadata that
have to go by before the bitstream data expires.
The value 0 disables expiry. */
DMX_BS_DATA_DELAY =
0x11, /*!< The number of delay frames of the output samples
compared to the bitstream data. */
MIN_NUMBER_OF_OUTPUT_CHANNELS =
0x20, /*!< The minimum number of output channels. For all
input configurations that have less than the given
channels the module will modify the output
automatically to obtain the given number of output
channels. Mono signals will be duplicated. If more
than two output channels are desired the module
just adds empty channels. The parameter value must
be either -1, 0, 1, 2, 6 or 8. If the value is
greater than zero and exceeds the value of
parameter ::MAX_NUMBER_OF_OUTPUT_CHANNELS the
latter will be set to the same value. Both values
-1 and 0 disable the feature. */
MAX_NUMBER_OF_OUTPUT_CHANNELS =
0x21, /*!< The maximum number of output channels. For all
input configurations that have more than the given
channels the module will apply a mixdown
automatically to obtain the given number of output
channels. The value must be either -1, 0, 1, 2, 6
or 8. If it's greater than zero and lower or equal
than the value of ::MIN_NUMBER_OF_OUTPUT_CHANNELS
parameter the latter will be set to the same value.
The values -1 and 0 disable the feature. */
DMX_DUAL_CHANNEL_MODE =
0x30, /*!< Downmix mode for two channel audio data. See type
::DUAL_CHANNEL_MODE for details. */
DMX_PSEUDO_SURROUND_MODE =
0x31 /*!< Defines how module handles pseudo surround
compatible signals. See ::PSEUDO_SURROUND_MODE
type for details. */
} PCMDMX_PARAM;
/**
* \enum DMX_PROFILE_TYPE
*
* Valid value list for parameter ::DMX_PROFILE_SETTING.
*/
typedef enum {
DMX_PRFL_STANDARD =
0x0, /*!< The standard profile creates mixdown signals based on
the advanced downmix metadata (from a DSE), equations
and default values defined in ISO/IEC 14496:3
Ammendment 4. Any other (legacy) downmix metadata will
be ignored. */
DMX_PRFL_MATRIX_MIX =
0x1, /*!< This profile behaves just as the standard profile if
advanced downmix metadata (from a DSE) is available. If
not, the matrix_mixdown information embedded in the
program configuration element (PCE) will be applied. If
neither is the case the module creates a mixdown using
the default coefficients defined in MPEG-4 Ammendment 4.
The profile can be used e.g. to support legacy digital
TV (e.g. DVB) streams. */
DMX_PRFL_FORCE_MATRIX_MIX =
0x2, /*!< Similar to the ::DMX_PRFL_MATRIX_MIX profile but if both
the advanced (DSE) and the legacy (PCE) MPEG downmix
metadata are available the latter will be applied. */
DMX_PRFL_ARIB_JAPAN =
0x3 /*!< Downmix creation as described in ABNT NBR 15602-2. But
if advanced downmix metadata is available it will be
prefered. */
} DMX_PROFILE_TYPE;
/**
* \enum PSEUDO_SURROUND_MODE
*
* Valid value list for parameter ::DMX_PSEUDO_SURROUND_MODE.
*/
typedef enum {
NEVER_DO_PS_DMX =
-1, /*!< Ignore any metadata and do never create a pseudo surround
compatible downmix. (Default) */
AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if
signalled in bitstreams meta data. */
FORCE_PS_DMX =
1 /*!< Always create a pseudo surround compatible downmix.
CAUTION: This can lead to excessive signal cancellations
and signal level differences for non-compatible signals. */
} PSEUDO_SURROUND_MODE;
/**
* \enum DUAL_CHANNEL_MODE
*
* Valid value list for parameter ::DMX_DUAL_CHANNEL_MODE.
*/
typedef enum {
STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */
CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */
CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */
MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two
channels. */
} DUAL_CHANNEL_MODE;
#define DMX_PCM FIXP_DBL
#define DMX_PCMF FIXP_DBL
#define DMX_PCM_BITS DFRACT_BITS
#define FX_DMX2FX_PCM(x) FX_DBL2FX_PCM((FIXP_DBL)(x))
/* ------------------------ *
* MODULES INTERFACE: *
* ------------------------ */
typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX;
/*! \addtogroup pcmDmxResetFlags Modules reset flags
* Macros that can be used as parameter for function pcmDmx_Reset() to specify
* which parts of the module shall be reset.
* @{
*
* \def PCMDMX_RESET_PARAMS
* Only reset the user specific parameters that have been modified with
* pcmDmx_SetParam().
*
* \def PCMDMX_RESET_BS_DATA
* Delete the meta data that has been fed with the appropriate interface
* functions.
*
* \def PCMDMX_RESET_FULL
* Reset the complete module instance to the state after pcmDmx_Open() had been
* called.
*/
#define PCMDMX_RESET_PARAMS (1)
#define PCMDMX_RESET_BS_DATA (2)
#define PCMDMX_RESET_FULL (PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA)
/*! @} */
#ifdef __cplusplus
extern "C" {
#endif
/** Open and initialize an instance of the PCM downmix module
* @param[out] pSelf Pointer to a buffer receiving the handle of the new
*instance.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_Open(HANDLE_PCM_DOWNMIX *pSelf);
/** Set one parameter for a single instance of the PCM downmix module.
* @param[in] self Handle of PCM downmix instance.
* @param[in] param Parameter to be set. Can be one from the ::PCMDMX_PARAM
*list.
* @param[in] value Parameter value.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_SetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param,
const INT value);
/** Get one parameter value of a single PCM downmix module instance.
* @param[in] self Handle of PCM downmix module instance.
* @param[in] param Parameter to query. Can be one from the ::PCMDMX_PARAM
*list.
* @param[out] pValue Pointer to buffer receiving the parameter value.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_GetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param,
INT *const pValue);
/** \cond
* Extract relevant downmix meta-data directly from a given bitstream. The
*function can handle both data specified in ETSI TS 101 154 or ISO/IEC
*14496-3:2009/Amd.4:2013.
* @param[in] self Handle of PCM downmix instance.
* @param[in] hBitStream Handle of FDK bitstream buffer.
* @param[in] ancDataBits Length of ancillary data in bits.
* @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a
*MPEG-1/2 or a MPEG-4 stream.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_Parse(HANDLE_PCM_DOWNMIX self,
HANDLE_FDK_BITSTREAM hBitStream, UINT ancDataBits,
int isMpeg2);
/** \endcond */
/** Read from a given ancillary data buffer and extract the relevant downmix
*meta-data. The function can handle both data specified in ETSI TS 101 154 or
*ISO/IEC 14496-3:2009/Amd.4:2013.
* @param[in] self Handle of PCM downmix instance.
* @param[in] pAncDataBuf Pointer to ancillary buffer holding the data.
* @param[in] ancDataBytes Size of ancillary data in bytes.
* @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a
*MPEG-1/2 or a MPEG-4 stream.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_ReadDvbAncData(HANDLE_PCM_DOWNMIX self, UCHAR *pAncDataBuf,
UINT ancDataBytes, int isMpeg2);
/** Set the matrix mixdown information extracted from the PCE of an AAC
*bitstream.
* @param[in] self Handle of PCM downmix instance.
* @param[in] matrixMixdownPresent Matrix mixdown index present flag extracted
*from PCE.
* @param[in] matrixMixdownIdx The 2 bit matrix mixdown index extracted
*from PCE.
* @param[in] pseudoSurroundEnable The pseudo surround enable flag extracted
*from PCE.
* @returns Returns an error code of type
*::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce(HANDLE_PCM_DOWNMIX self,
int matrixMixdownPresent,
int matrixMixdownIdx,
int pseudoSurroundEnable);
/** Reset the module.
* @param[in] self Handle of PCM downmix instance.
* @param[in] flags Flags telling which parts of the module shall be reset.
* See \ref pcmDmxResetFlags for details.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_Reset(HANDLE_PCM_DOWNMIX self, UINT flags);
/** Create a mixdown, bypass or extend the output signal depending on the
*modules settings and the respective given input configuration.
*
* \param[in] self Handle of PCM downmix module instance.
* \param[in,out] pPcmBuf Pointer to time buffer with PCM samples.
* \param[in] pcmBufSize Size of pPcmBuf buffer.
* \param[in] frameSize The I/O block size which is the number of samples per channel.
* \param[in,out] nChannels Pointer to buffer that holds the number of input channels and
* where the amount of output channels is written
*to.
* \param[in] fInterleaved Input and output samples are processed interleaved.
* \param[in,out] channelType Array were the corresponding channel type for each output audio
* channel is stored into.
* \param[in,out] channelIndices Array were the corresponding channel type index for each output
* audio channel is stored into.
* \param[in] mapDescr Pointer to a FDK channel mapping descriptor that contains the
* channel mapping to be used.
* \param[out] pDmxOutScale Pointer on a field receiving the scale factor that has to be
* applied on all samples afterwards. If the
*handed pointer is NULL the final scaling is done internally.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_ApplyFrame(HANDLE_PCM_DOWNMIX self, DMX_PCM *pPcmBuf,
const int pcmBufSize, UINT frameSize,
INT *nChannels, INT fInterleaved,
AUDIO_CHANNEL_TYPE channelType[],
UCHAR channelIndices[],
const FDK_channelMapDescr *const mapDescr,
INT *pDmxOutScale);
/** Close an instance of the PCM downmix module.
* @param[in,out] pSelf Pointer to a buffer containing the handle of the
*instance.
* @returns Returns an error code of type ::PCMDMX_ERROR.
**/
PCMDMX_ERROR pcmDmx_Close(HANDLE_PCM_DOWNMIX *pSelf);
/** Get library info for this module.
* @param[out] info Pointer to an allocated LIB_INFO structure.
* @returns Returns an error code of type ::PCMDMX_ERROR.
*/
PCMDMX_ERROR pcmDmx_GetLibInfo(LIB_INFO *info);
#ifdef __cplusplus
}
#endif
#endif /* PCMDMX_LIB_H */

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@@ -1,334 +0,0 @@
/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
<EFBFBD> Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur F<>rderung 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
----------------------------------------------------------------------------------------------------------- */
/************************ FDK PCM up/downmixing module *********************
Author(s): Christian Griebel
Description: Declares functions to interface with the PCM downmix processing
module.
*******************************************************************************/
#ifndef _PCMUTILS_LIB_H_
#define _PCMUTILS_LIB_H_
#include "machine_type.h"
#include "common_fix.h"
#include "FDK_audio.h"
#include "FDK_bitstream.h"
/* ------------------------ *
* ERROR CODES: *
* ------------------------ */
typedef enum
{
PCMDMX_OK = 0x0, /*!< No error happened. */
pcm_dmx_fatal_error_start,
PCMDMX_OUT_OF_MEMORY = 0x2, /*!< Not enough memory to set up an instance of the module. */
PCMDMX_UNKNOWN = 0x5, /*!< Error condition is of unknown reason, or from a third
party module. */
pcm_dmx_fatal_error_end,
PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */
PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */
PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not supported and thus
no processing was performed. */
PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */
PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */
PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most probably the
value ist out of range. */
PCMDMX_CORRUPT_ANC_DATA /*!< The read ancillary data was corrupt. */
} PCMDMX_ERROR;
/** Macro to identify fatal errors. */
#define PCMDMX_IS_FATAL_ERROR(err) ( (((err)>=pcm_dmx_fatal_error_start) && ((err)<=pcm_dmx_fatal_error_end)) ? 1 : 0)
/* ------------------------ *
* RUNTIME PARAMS: *
* ------------------------ */
typedef enum
{
DMX_BS_DATA_EXPIRY_FRAME, /*!< The number of frames without new metadata that have to go
by before the bitstream data expires. The value 0 disables
expiry. */
DMX_BS_DATA_DELAY, /*!< The number of delay frames of the output samples compared
to the bitstream data. */
MIN_NUMBER_OF_OUTPUT_CHANNELS, /*!< The minimum number of output channels. For all input
configurations that have less than the given channels the
module will modify the output automatically to obtain the
given number of output channels. Mono signals will be
duplicated. If more than two output channels are desired
the module just adds empty channels. The parameter value
must be either -1, 0, 1, 2, 6 or 8. If the value is
greater than zero and exceeds the value of parameter
MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to
the same value. Both values -1 and 0 disable the feature. */
MAX_NUMBER_OF_OUTPUT_CHANNELS, /*!< The maximum number of output channels. For all input
configurations that have more than the given channels the
module will apply a mixdown automatically to obtain the
given number of output channels. The value must be either
-1, 0, 1, 2, 6 or 8. If it is greater than zero and lower
or equal than the value of MIN_NUMBER_OF_OUTPUT_CHANNELS
parameter the latter will be set to the same value.
The values -1 and 0 disable the feature. */
DMX_DUAL_CHANNEL_MODE, /*!< Downmix mode for two channel audio data. */
DMX_PSEUDO_SURROUND_MODE /*!< Defines how module handles pseudo surround compatible
signals. See PSEUDO_SURROUND_MODE type for details. */
} PCMDMX_PARAM;
/* Parameter value types */
typedef enum
{
NEVER_DO_PS_DMX = -1, /*!< Never create a pseudo surround compatible downmix. */
AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if
signalled in bitstreams meta data. (Default) */
FORCE_PS_DMX = 1 /*!< Always create a pseudo surround compatible downmix.
CAUTION: This can lead to excessive signal cancellations
and signal level differences for non-compatible signals. */
} PSEUDO_SURROUND_MODE;
typedef enum
{
STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */
CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */
CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */
MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two
channels. */
} DUAL_CHANNEL_MODE;
/* ------------------------ *
* MODULES INTERFACE: *
* ------------------------ */
typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX;
/* Modules reset flags */
#define PCMDMX_RESET_PARAMS ( 1 )
#define PCMDMX_RESET_BS_DATA ( 2 )
#define PCMDMX_RESET_FULL ( PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA )
#ifdef __cplusplus
extern "C"
{
#endif
/** Open and initialize an instance of the PCM downmix module
* @param [out] Pointer to a buffer receiving the handle of the new instance.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Open (
HANDLE_PCM_DOWNMIX *pSelf
);
/** Set one parameter for one instance of the PCM downmix module.
* @param [in] Handle of PCM downmix instance.
* @param [in] Parameter to be set.
* @param [in] Parameter value.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_SetParam (
HANDLE_PCM_DOWNMIX self,
const PCMDMX_PARAM param,
const INT value
);
/** Get one parameter value of one PCM downmix module instance.
* @param [in] Handle of PCM downmix module instance.
* @param [in] Parameter to be set.
* @param [out] Pointer to buffer receiving the parameter value.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_GetParam (
HANDLE_PCM_DOWNMIX self,
const PCMDMX_PARAM param,
INT * const pValue
);
/** Read downmix meta-data directly from a given bitstream.
* @param [in] Handle of PCM downmix instance.
* @param [in] Handle of FDK bitstream buffer.
* @param [in] Length of ancillary data in bits.
* @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Parse (
HANDLE_PCM_DOWNMIX self,
HANDLE_FDK_BITSTREAM hBitStream,
UINT ancDataBits,
int isMpeg2
);
/** Read downmix meta-data from a given data buffer.
* @param [in] Handle of PCM downmix instance.
* @param [in] Pointer to ancillary data buffer.
* @param [in] Size of ancillary data in bytes.
* @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_ReadDvbAncData (
HANDLE_PCM_DOWNMIX self,
UCHAR *pAncDataBuf,
UINT ancDataBytes,
int isMpeg2
);
/** Set the matrix mixdown information extracted from the PCE of an AAC bitstream.
* @param [in] Handle of PCM downmix instance.
* @param [in] Matrix mixdown index present flag extracted from PCE.
* @param [in] The 2 bit matrix mixdown index extracted from PCE.
* @param [in] The pseudo surround enable flag extracted from PCE.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce (
HANDLE_PCM_DOWNMIX self,
int matrixMixdownPresent,
int matrixMixdownIdx,
int pseudoSurroundEnable
);
/** Reset the module.
* @param [in] Handle of PCM downmix instance.
* @param [in] Flags telling which parts of the module shall be reset.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Reset (
HANDLE_PCM_DOWNMIX self,
UINT flags
);
/** Create a mixdown, bypass or extend the output signal depending on the modules settings and the
* respective given input configuration.
*
* \param [in] Handle of PCM downmix module instance.
* \param [inout] Pointer to time buffer with decoded PCM samples.
* \param [in] The I/O block size which is the number of samples per channel.
* \param [inout] Pointer to buffer that holds the number of input channels and where the
* amount of output channels is written to.
* \param [in] Flag which indicates if output time data is writtern interleaved or as
* subsequent blocks.
* \param [inout] Array were the corresponding channel type for each output audio channel is
* stored into.
* \param [inout] Array were the corresponding channel type index for each output audio channel
* is stored into.
* \param [in] Array containing the output channel mapping to be used (from MPEG PCE ordering
* to whatever is required).
* \param [out] Pointer on a field receiving the scale factor that has to be applied on all
* samples afterwards. If the handed pointer is NULL the final scaling is done
* internally.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_ApplyFrame (
HANDLE_PCM_DOWNMIX self,
INT_PCM *pPcmBuf,
UINT frameSize,
INT *nChannels,
int fInterleaved,
AUDIO_CHANNEL_TYPE channelType[],
UCHAR channelIndices[],
const UCHAR channelMapping[][8],
INT *pDmxOutScale
);
/** Close an instance of the PCM downmix module.
* @param [inout] Pointer to a buffer containing the handle of the instance.
* @returns Returns an error code.
**/
PCMDMX_ERROR pcmDmx_Close (
HANDLE_PCM_DOWNMIX *pSelf
);
/** Get library info for this module.
* @param [out] Pointer to an allocated LIB_INFO structure.
* @returns Returns an error code.
*/
PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info );
#ifdef __cplusplus
}
#endif
#endif /* _PCMUTILS_LIB_H_ */