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461 lines
20 KiB
C
461 lines
20 KiB
C
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/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of
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this software, even if advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------- */
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/**************************** PCM utility library ******************************
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Author(s): Christian Griebel
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Description:
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*******************************************************************************/
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/**
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* \file pcmdmx_lib.h
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* \brief FDK PCM audio mixdown library interface header file.
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\page INTRO Introduction
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\section SCOPE Scope
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This document describes the high-level application interface and usage of the
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FDK PCM audio mixdown module library developed by the Fraunhofer Institute for
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Integrated Circuits (IIS). Depending on the library configuration, the module
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can manipulate the number of audio channels of a given PCM signal. It can
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create for example a two channel stereo audio signal from a given multi-channel
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configuration (e.g. 5.1 channels).
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\page ABBREV List of abbreviations
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\li \b AAC - Advanced Audio Coding\n
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Is an audio coding standard for lossy digital audio compression standardized
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by ISO and IEC, as part of the MPEG-2 (ISO/IEC 13818-7:2006) and MPEG-4
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(ISO/IEC 14496-3:2009) specifications.
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\li \b DSE - Data Stream Element\n
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A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream
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standardized in ISO/IEC 14496-3:2009. It can convey any kind of data associated
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to one program.
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\li \b PCE - Program Config Element\n
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A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream
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standardized in ISO/IEC 14496-3:2009 that can define the stream configuration
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for a single program. In addition it can comprise simple downmix meta data.
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*/
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#ifndef PCMDMX_LIB_H
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#define PCMDMX_LIB_H
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#include "machine_type.h"
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#include "common_fix.h"
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#include "FDK_audio.h"
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#include "FDK_bitstream.h"
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/**
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* \enum PCMDMX_ERROR
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*
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* Error codes that can be returned by module interface functions.
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*/
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typedef enum {
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PCMDMX_OK = 0x0, /*!< No error happened. */
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PCMDMX_UNSUPPORTED =
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0x1, /*!< The requested feature/service is unavailable. This can
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occur if the module was built for a wrong configuration. */
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pcm_dmx_fatal_error_start,
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PCMDMX_OUT_OF_MEMORY, /*!< Not enough memory to set up an instance of the
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module. */
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pcm_dmx_fatal_error_end,
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PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */
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PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */
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PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not
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supported and thus no processing was performed.
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*/
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PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */
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PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */
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PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most
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probably the value ist out of range.
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*/
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PCMDMX_CORRUPT_ANC_DATA, /*!< The read ancillary data was corrupt. */
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PCMDMX_OUTPUT_BUFFER_TOO_SMALL /*!< The size of pcm output buffer is too
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small. */
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} PCMDMX_ERROR;
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/** Macro to identify fatal errors. */
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#define PCMDMX_IS_FATAL_ERROR(err) \
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((((err) >= pcm_dmx_fatal_error_start) && \
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((err) <= pcm_dmx_fatal_error_end)) \
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? 1 \
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: 0)
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/**
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* \enum PCMDMX_PARAM
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*
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* Modules dynamic runtime parameters that can be handed to function
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* pcmDmx_SetParam() and pcmDmx_GetParam().
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*/
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typedef enum {
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DMX_PROFILE_SETTING =
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0x01, /*!< Defines which equations, coefficients and default/
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fallback values used for downmixing. See
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::DMX_PROFILE_TYPE type for details. */
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DMX_BS_DATA_EXPIRY_FRAME =
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0x10, /*!< The number of frames without new metadata that
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have to go by before the bitstream data expires.
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The value 0 disables expiry. */
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DMX_BS_DATA_DELAY =
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0x11, /*!< The number of delay frames of the output samples
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compared to the bitstream data. */
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MIN_NUMBER_OF_OUTPUT_CHANNELS =
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0x20, /*!< The minimum number of output channels. For all
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input configurations that have less than the given
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channels the module will modify the output
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automatically to obtain the given number of output
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channels. Mono signals will be duplicated. If more
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than two output channels are desired the module
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just adds empty channels. The parameter value must
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be either -1, 0, 1, 2, 6 or 8. If the value is
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greater than zero and exceeds the value of
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parameter ::MAX_NUMBER_OF_OUTPUT_CHANNELS the
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latter will be set to the same value. Both values
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-1 and 0 disable the feature. */
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MAX_NUMBER_OF_OUTPUT_CHANNELS =
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0x21, /*!< The maximum number of output channels. For all
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input configurations that have more than the given
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channels the module will apply a mixdown
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automatically to obtain the given number of output
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channels. The value must be either -1, 0, 1, 2, 6
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or 8. If it's greater than zero and lower or equal
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than the value of ::MIN_NUMBER_OF_OUTPUT_CHANNELS
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parameter the latter will be set to the same value.
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The values -1 and 0 disable the feature. */
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DMX_DUAL_CHANNEL_MODE =
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0x30, /*!< Downmix mode for two channel audio data. See type
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::DUAL_CHANNEL_MODE for details. */
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DMX_PSEUDO_SURROUND_MODE =
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0x31 /*!< Defines how module handles pseudo surround
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compatible signals. See ::PSEUDO_SURROUND_MODE
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type for details. */
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} PCMDMX_PARAM;
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/**
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* \enum DMX_PROFILE_TYPE
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*
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* Valid value list for parameter ::DMX_PROFILE_SETTING.
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*/
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typedef enum {
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DMX_PRFL_STANDARD =
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0x0, /*!< The standard profile creates mixdown signals based on
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the advanced downmix metadata (from a DSE), equations
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and default values defined in ISO/IEC 14496:3
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Ammendment 4. Any other (legacy) downmix metadata will
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be ignored. */
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DMX_PRFL_MATRIX_MIX =
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0x1, /*!< This profile behaves just as the standard profile if
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advanced downmix metadata (from a DSE) is available. If
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not, the matrix_mixdown information embedded in the
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program configuration element (PCE) will be applied. If
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neither is the case the module creates a mixdown using
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the default coefficients defined in MPEG-4 Ammendment 4.
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The profile can be used e.g. to support legacy digital
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TV (e.g. DVB) streams. */
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DMX_PRFL_FORCE_MATRIX_MIX =
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0x2, /*!< Similar to the ::DMX_PRFL_MATRIX_MIX profile but if both
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the advanced (DSE) and the legacy (PCE) MPEG downmix
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metadata are available the latter will be applied. */
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DMX_PRFL_ARIB_JAPAN =
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0x3 /*!< Downmix creation as described in ABNT NBR 15602-2. But
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if advanced downmix metadata is available it will be
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prefered. */
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} DMX_PROFILE_TYPE;
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/**
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* \enum PSEUDO_SURROUND_MODE
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*
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* Valid value list for parameter ::DMX_PSEUDO_SURROUND_MODE.
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*/
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typedef enum {
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NEVER_DO_PS_DMX =
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-1, /*!< Ignore any metadata and do never create a pseudo surround
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compatible downmix. (Default) */
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AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if
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signalled in bitstreams meta data. */
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FORCE_PS_DMX =
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1 /*!< Always create a pseudo surround compatible downmix.
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CAUTION: This can lead to excessive signal cancellations
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and signal level differences for non-compatible signals. */
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} PSEUDO_SURROUND_MODE;
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/**
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* \enum DUAL_CHANNEL_MODE
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*
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* Valid value list for parameter ::DMX_DUAL_CHANNEL_MODE.
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*/
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typedef enum {
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STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */
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CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */
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CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */
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MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two
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channels. */
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} DUAL_CHANNEL_MODE;
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#define DMX_PCM FIXP_DBL
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#define DMX_PCMF FIXP_DBL
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#define DMX_PCM_BITS DFRACT_BITS
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#define FX_DMX2FX_PCM(x) FX_DBL2FX_PCM((FIXP_DBL)(x))
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/* ------------------------ *
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* MODULES INTERFACE: *
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* ------------------------ */
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typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX;
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/*! \addtogroup pcmDmxResetFlags Modules reset flags
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* Macros that can be used as parameter for function pcmDmx_Reset() to specify
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* which parts of the module shall be reset.
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* @{
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*
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* \def PCMDMX_RESET_PARAMS
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* Only reset the user specific parameters that have been modified with
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* pcmDmx_SetParam().
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*
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* \def PCMDMX_RESET_BS_DATA
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* Delete the meta data that has been fed with the appropriate interface
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* functions.
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*
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* \def PCMDMX_RESET_FULL
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* Reset the complete module instance to the state after pcmDmx_Open() had been
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* called.
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*/
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#define PCMDMX_RESET_PARAMS (1)
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#define PCMDMX_RESET_BS_DATA (2)
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#define PCMDMX_RESET_FULL (PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA)
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/*! @} */
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#ifdef __cplusplus
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extern "C" {
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#endif
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/** Open and initialize an instance of the PCM downmix module
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* @param[out] pSelf Pointer to a buffer receiving the handle of the new
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*instance.
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* @returns Returns an error code of type ::PCMDMX_ERROR.
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**/
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PCMDMX_ERROR pcmDmx_Open(HANDLE_PCM_DOWNMIX *pSelf);
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/** Set one parameter for a single instance of the PCM downmix module.
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* @param[in] self Handle of PCM downmix instance.
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* @param[in] param Parameter to be set. Can be one from the ::PCMDMX_PARAM
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*list.
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* @param[in] value Parameter value.
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* @returns Returns an error code of type ::PCMDMX_ERROR.
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**/
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PCMDMX_ERROR pcmDmx_SetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param,
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const INT value);
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/** Get one parameter value of a single PCM downmix module instance.
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* @param[in] self Handle of PCM downmix module instance.
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* @param[in] param Parameter to query. Can be one from the ::PCMDMX_PARAM
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*list.
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* @param[out] pValue Pointer to buffer receiving the parameter value.
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* @returns Returns an error code of type ::PCMDMX_ERROR.
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**/
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PCMDMX_ERROR pcmDmx_GetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param,
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INT *const pValue);
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/** \cond
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* Extract relevant downmix meta-data directly from a given bitstream. The
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*function can handle both data specified in ETSI TS 101 154 or ISO/IEC
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*14496-3:2009/Amd.4:2013.
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* @param[in] self Handle of PCM downmix instance.
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* @param[in] hBitStream Handle of FDK bitstream buffer.
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* @param[in] ancDataBits Length of ancillary data in bits.
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* @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a
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*MPEG-1/2 or a MPEG-4 stream.
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* @returns Returns an error code of type ::PCMDMX_ERROR.
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**/
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PCMDMX_ERROR pcmDmx_Parse(HANDLE_PCM_DOWNMIX self,
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HANDLE_FDK_BITSTREAM hBitStream, UINT ancDataBits,
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int isMpeg2);
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/** \endcond */
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/** Read from a given ancillary data buffer and extract the relevant downmix
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*meta-data. The function can handle both data specified in ETSI TS 101 154 or
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*ISO/IEC 14496-3:2009/Amd.4:2013.
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* @param[in] self Handle of PCM downmix instance.
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* @param[in] pAncDataBuf Pointer to ancillary buffer holding the data.
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* @param[in] ancDataBytes Size of ancillary data in bytes.
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* @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a
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*MPEG-1/2 or a MPEG-4 stream.
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* @returns Returns an error code of type ::PCMDMX_ERROR.
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**/
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PCMDMX_ERROR pcmDmx_ReadDvbAncData(HANDLE_PCM_DOWNMIX self, UCHAR *pAncDataBuf,
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UINT ancDataBytes, int isMpeg2);
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/** Set the matrix mixdown information extracted from the PCE of an AAC
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*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 */
|