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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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