path: root/libAACenc/include/aacenc_lib.h

/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android

� 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
----------------------------------------------------------------------------------------------------------- */

/**************************** MPEG-4 HE-AAC Encoder **************************

  Initial author:       M. Lohwasser
******************************************************************************/

/**
 * \file   aacenc_lib.h
 * \brief  FDK AAC Encoder library interface header file.
 *
\page  FDK AAC Encoder Library Introduction

\section Scope

This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Encoder
library developed by the Fraunhofer Institute for Integrated Circuits (IIS).

The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC Low-Complexity
standard, and depending on the library's configuration, MPEG-4 High-Efficiency AAC v2 and/or AAC-ELD standard.

All references to SBR (Spectral Band Replication) are only applicable to HE-AAC or AAC-ELD versions
of the library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2
versions of the library.

\section encBasics Encoder Basics

This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio coding
standard. To understand all the terms in this document, you are encouraged to read the following documents.

- ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams.
- ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams.
- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004

MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is
partitioned into overlapping portions and transformed into frequency domain. The spectral components
are then quantized and coded. \n
An MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the
length of individual frames is not restricted to a fixed number of bytes, but can take on any length
between 1 and 768 bytes.


\page LIBUSE Library Usage

\section InterfaceDescription API Files

All API header files are located in the folder /include of the release package. All header files
are provided for usage in C/C++ programs. The AAC encoder library API functions are located at
aacenc_lib.h.

In binary releases the encoder core resides in statically linkable libraries called for example
libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual C++) for the plain AAC-LC core encoder
and libSBRenc.a (LINUX) or FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band
Replication) and PS (Parametric Stereo) modules.

\section CallingSequence Calling Sequence

For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. Input read and output
write functions as well as the corresponding open and close functions are left out, since they may be
implemented differently according to the user's specific requirements. The example implementation in
main.cpp uses file-based input/output.

-# Call aacEncOpen() to allocate encoder instance with required \ref encOpen "configuration".\n
\dontinclude main.cpp
\skipline hAacEncoder =
\skipline aacEncOpen
-# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, channelMode, bitrate and transport type are \ref encParams "mandatory".
\code
    ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value);
\endcode
-# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" encoder instance with present parameter set.
\skipline aacEncEncode
-# Call aacEncInfo() to retrieve a configuration data block to be transmitted out of band. This is required when using RFC3640 or RFC3016 like transport.
\dontinclude main.cpp
\skipline encInfo
\skipline aacEncInfo
-# Encode input audio data in loop.
\skip Encode as long as
\skipline do
\until {
Feed \ref feedInBuf "input buffer" with new audio data and provide input/output \ref bufDes "arguments" to aacEncEncode().
\skipline aacEncEncode
\until ;
Write \ref writeOutData "output data" to file or audio device. \skipline while
-# Call aacEncClose() and destroy encoder instance.
\skipline aacEncClose

\section encOpen Encoder Instance Allocation

The assignment of the aacEncOpen() function is very flexible and can be used in the following way.
- If the amount of memory consumption is not an issue, the encoder instance can be allocated
for the maximum number of possible audio channels (for example 6 or 8) with the full functional range supported by the library.
This is the default open procedure for the AAC encoder if memory consumption does not need to be minimized.
\code aacEncOpen(&hAacEncoder,0,0) \endcode
- If the required MPEG-4 AOTs do not call for the full functional range of the library, encoder modules can be allocated selectively.
\verbatim
------------------------------------------------------
 AAC | SBR |  PS | MD |         FLAGS         | value
-----+-----+-----+----+-----------------------+-------
  X  |  -  |  -  |  - | (0x01)                |  0x01
  X  |  X  |  -  |  - | (0x01|0x02)           |  0x03
  X  |  X  |  X  |  - | (0x01|0x02|0x04)      |  0x07
  X  |  -  |  -  |  X | (0x01          |0x10) |  0x11
  X  |  X  |  -  |  X | (0x01|0x02     |0x10) |  0x13
  X  |  X  |  X  |  X | (0x01|0x02|0x04|0x10) |  0x17
------------------------------------------------------
 - AAC: Allocate AAC Core Encoder module.
 - SBR: Allocate Spectral Band Replication module.
 - PS: Allocate Parametric Stereo module.
 - MD: Allocate Meta Data module within AAC encoder.
\endverbatim
\code aacEncOpen(&hAacEncoder,value,0) \endcode
- Specifying the maximum number of channels to be supported in the encoder instance can be done as follows.
 - For example allocate an encoder instance which supports 2 channels for all supported AOTs.
   The library itself may be capable of encoding up to 6 or 8 channels but in this example only 2 channel encoding is required and thus only buffers for 2 channels are allocated to save data memory.
\code aacEncOpen(&hAacEncoder,0,2) \endcode
 - Additionally the maximum number of supported channels in the SBR module can be denoted separately.\n
   In this example the encoder instance provides a maximum of 6 channels out of which up to 2 channels support SBR.
   This encoder instance can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) streams.
   HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels support SBR, which saves data memory.
\code aacEncOpen(&hAacEncoder,0,6|(2<<8)) \endcode
\n

\section bufDes Input/Output Arguments

\subsection allocIOBufs Provide Buffer Descriptors
In the present encoder API, the input and output buffers are described with \ref AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling
of input and output buffers without impact to the actual encoding call. Optional buffers are necessary e.g. for ancillary data, meta data input or additional output
buffers describing superframing data in DAB+ or DRM+.\n
At least one input buffer for audio input data and one output buffer for bitstream data must be allocated. The input buffer size can be a user defined multiple
of the number of input channels. PCM input data will be copied from the user defined PCM buffer to an internal input buffer and so input data can be less than one AAC audio frame.
The output buffer size should be 6144 bits per channel excluding the LFE channel.
If the output data does not fit into the provided buffer, an AACENC_ERROR will be returned by aacEncEncode().
\dontinclude main.cpp
\skipline inputBuffer
\until outputBuffer
All input and output buffer must be clustered in input and output buffer arrays.
\skipline inBuffer
\until outBufferElSize
Allocate buffer descriptors
\skipline AACENC_BufDesc
\skipline AACENC_BufDesc
Initialize input buffer descriptor
\skipline inBufDesc
\until bufElSizes
Initialize output buffer descriptor
\skipline outBufDesc
\until bufElSizes

\subsection argLists Provide Input/Output Argument Lists
The input and output arguments of an aacEncEncode() call are described in argument structures.
\dontinclude main.cpp
\skipline AACENC_InArgs
\skipline AACENC_OutArgs

\section feedInBuf Feed Input Buffer
The input buffer should be handled as a modulo buffer. New audio data in the form of pulse-code-
modulated samples (PCM) must be read from external and be fed to the input buffer depending on its
fill level. The required sample bitrate (represented by the data type INT_PCM which is 16, 24 or 32
bits wide) is fixed and depends on library configuration (usually 16 bit).

\dontinclude main.cpp
\skipline WAV_InputRead
\until ;
After the encoder's internal buffer is fed with incoming audio samples, and aacEncEncode()
processed the new input data, update/move remaining samples in input buffer, simulating a modulo buffer:
\skipline outargs.numInSamples>0
\until }

\section writeOutData Output Bitstream Data
If any AAC bitstream data is available, write it to output file or device. This can be done once the
following condition is true:
\dontinclude main.cpp
\skip Valid bitstream available
\skipline outargs

\skipline outBytes>0

If you use file I/O then for example call mpegFileWrite_Write() from the library libMpegFileWrite

\dontinclude main.cpp
\skipline mpegFileWrite_Write

\section cfgMetaData Meta Data Configuration

If the present library is configured with Metadata support, it is possible to insert meta data side info into the generated
audio bitstream while encoding.

To work with meta data the encoder instance has to be \ref encOpen "allocated" with meta data support. The meta data mode must be be configured with
the ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function.
\code aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-2); \endcode

This configuration indicates how to embed meta data into bitstrem. Either no insertion, MPEG or ETSI style.
The meta data itself must be specified within the meta data setup structure AACENC_MetaData.

Changing one of the AACENC_MetaData setup parameters can be achieved from outside the library within ::IN_METADATA_SETUP input
buffer. There is no need to supply meta data setup structure every frame. If there is no new meta setup data available, the
encoder uses the previous setup or the default configuration in initial state.

In general the audio compressor and limiter within the encoder library can be configured with the ::AACENC_METADATA_DRC_PROFILE parameter
AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile.
\n

\section encReconf Encoder Reconfiguration

The encoder library allows reconfiguration of the encoder instance with new settings
continuously between encoding frames. Each parameter to be changed must be set with
a single aacEncoder_SetParam() call. The internal status of each parameter can be
retrieved with an aacEncoder_GetParam() call.\n
There is no stand-alone reconfiguration function available. When parameters were
modified from outside the library, an internal control mechanism triggers the necessary
reconfiguration process which will be applied at the beginning of the following
aacEncEncode() call. This state can be observed from external via the AACENC_INIT_STATUS
and aacEncoder_GetParam() function. The reconfiguration process can also be applied
immediately when all parameters of an aacEncEncode() call are NULL with a valid encoder
handle.\n\n
The internal reconfiguration process can be controlled from extern with the following access.
\code aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); \endcode


\section encParams Encoder Parametrization

All parameteres listed in ::AACENC_PARAM can be modified within an encoder instance.

\subsection encMandatory Mandatory Encoder Parameters
The following parameters must be specified when the encoder instance is initialized.
\code
aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value);
aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value);
aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value);
aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value);
\endcode
Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE parameter
if the parameter was not set from extern. The bitrate depends on the number of effective
channels and sampling rate and is determined as follows.
\code
AAC-LC (AOT_AAC_LC): 1.5 bits per sample
HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr)
HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr)
HE-AAC v2 (AOT_PS): 0.5 bits per sample
\endcode

\subsection channelMode Channel Mode Configuration
The input audio data is described with the ::AACENC_CHANNELMODE parameter in the
aacEncoder_SetParam() call. It is not possible to use the encoder instance with a 'number of
input channels' argument. Instead, the channelMode must be set as follows.
\code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode
The parameter is specified in ::CHANNEL_MODE and can be mapped from the number of input channels
in the following way.
\dontinclude main.cpp
\skip CHANNEL_MODE chMode = MODE_INVALID;
\until return

\subsection encQual Audio Quality Considerations
The default encoder configuration is suggested to be used. Encoder tools such as TNS and PNS
are activated by default and are internally controlled (see \ref BEHAVIOUR_TOOLS).

There is an additional quality parameter called ::AACENC_AFTERBURNER. In the default
configuration this quality switch is deactivated because it would cause a workload
increase which might be significant. If workload is not an issue in the application
we recommended to activate this feature.
\code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 1); \endcode

\subsection encELD ELD Auto Configuration Mode
For ELD configuration a so called auto configurator is available which configures SBR and the SBR ratio by itself.
The configurator is used when the encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set explicitely.

Based on sampling rate and chosen bitrate per channel a reasonable SBR configuration will be used.
\verbatim
------------------------------------------------------------
  Sampling Rate  | Channel Bitrate |  SBR |       SBR Ratio
-----------------+-----------------+------+-----------------
 ]min, 16] kHz   |     min - 27999 |   on | downsampled SBR
                 |   28000 -   max |  off |             ---
-----------------+-----------------+------+-----------------
 ]16 - 24] kHz   |     min - 39999 |   on | downsampled SBR
                 |   40000 -   max |  off |             ---
-----------------+-----------------+------+-----------------
 ]24 - 32] kHz   |     min - 27999 |   on |    dualrate SBR
                 |   28000 - 55999 |   on | downsampled SBR
                 |   56000 -   max |  off |             ---
-----------------+-----------------+------+-----------------
 ]32 - 44.1] kHz |     min - 63999 |   on |    dualrate SBR
                 |   64000 -   max |  off |             ---
-----------------+-----------------+------+-----------------
 ]44.1 - 48] kHz |     min - 63999 |   on |    dualrate SBR
                 |   64000 - max   |  off |             ---
------------------------------------------------------------
\endverbatim


\section audiochCfg Audio Channel Configuration
The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel
Mapping. The configurations 1-7 are predefined in MPEG standard and used for implicit signalling within the encoded bitstream.
For user defined Configurations the Channel Configuration is set to 0 and the Channel Mapping must be explecitly described with an appropriate
Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from
extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow.
\verbatim
-------------------------------------------------------------------------------
 ChannelMode           | ChCfg  | front_El      | side_El  | back_El  | lfe_El
-----------------------+--------+---------------+----------+----------+--------
MODE_1                 |      1 | SCE           |          |          |
MODE_2                 |      2 | CPE           |          |          |
MODE_1_2               |      3 | SCE, CPE      |          |          |
MODE_1_2_1             |      4 | SCE, CPE      |          | SCE      |
MODE_1_2_2             |      5 | SCE, CPE      |          | CPE      |
MODE_1_2_2_1           |      6 | SCE, CPE      |          | CPE      | LFE
MODE_1_2_2_2_1         |      7 | SCE, CPE, CPE |          | CPE      | LFE
-----------------------+--------+---------------+----------+----------+--------
MODE_7_1_REAR_SURROUND |      0 | SCE, CPE      |          | CPE, CPE | LFE
MODE_7_1_FRONT_CENTER  |      0 | SCE, CPE, CPE |          | CPE      | LFE
-------------------------------------------------------------------------------
 - SCE: Single Channel Element.
 - CPE: Channel Pair.
 - SCE: Low Frequency Element.
\endverbatim

Moreover, the Table describes all fixed Channel Elements for each Channel Mode which are assigned to a speaker arrangement. The
arrangement includes front, side, back and lfe Audio Channel Elements.\n
This mapping of Audio Channel Elements is defined in MPEG standard for Channel Config 1-7. The Channel assignment for MODE_1_1,
MODE_2_2 and MODE_2_1 is used from the ARIB standard. All other configurations are defined as suggested in MPEG.\n
In case of Channel Config 0 or writing matrix mixdown coefficients, the encoder enables the writing of Program Config Element
itself as described in \ref encPCE. The configuration used in Program Config Element refers to the denoted Table.\n
Beside the Channel Element assignment the Channel Modes are resposible for audio input data channel mapping. The Channel Mapping
of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n
Following Table describes the complete channel mapping for both Channel Order configurations.
\verbatim
---------------------------------------------------------------------------------------
ChannelMode            |  MPEG-Channelorder            |  WAV-Channelorder
-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
MODE_1                 | 0 |   |   |   |   |   |   |   | 0 |   |   |   |   |   |   |
MODE_2                 | 0 | 1 |   |   |   |   |   |   | 0 | 1 |   |   |   |   |   |
MODE_1_2               | 0 | 1 | 2 |   |   |   |   |   | 2 | 0 | 1 |   |   |   |   |
MODE_1_2_1             | 0 | 1 | 2 | 3 |   |   |   |   | 2 | 0 | 1 | 3 |   |   |   |
MODE_1_2_2             | 0 | 1 | 2 | 3 | 4 |   |   |   | 2 | 0 | 1 | 3 | 4 |   |   |
MODE_1_2_2_1           | 0 | 1 | 2 | 3 | 4 | 5 |   |   | 2 | 0 | 1 | 4 | 5 | 3 |   |
MODE_1_2_2_2_1         | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3
MODE_7_1_FRONT_CENTER  | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
---------------------------------------------------------------------------------------
\endverbatim

The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio
channels are distributed MPEG like starting at the front channels and ending at the back channels. The distribution is used as
described in Table concering Channel Config and fix channel elements. Please see the following example for clarification.

\verbatim
Example: MODE_1_2_2_1 - WAV-Channelorder 5.1
------------------------------------------
 Input Channel      | Coder Channel
--------------------+---------------------
 2 (front center)   | 0 (SCE channel)
 0 (left center)    | 1 (1st of 1st CPE)
 1 (right center)   | 2 (2nd of 1st CPE)
 4 (left surround)  | 3 (1st of 2nd CPE)
 5 (right surround) | 4 (2nd of 2nd CPE)
 3 (LFE)            | 5 (LFE)
------------------------------------------
\endverbatim


\section suppBitrates Supported Bitrates

The FDK AAC Encoder provides a wide range of supported bitrates.
The minimum and maximum allowed bitrate depends on the Audio Object Type. For AAC-LC the minimum
bitrate is the bitrate that is required to write the most basic and minimal valid bitstream.
It consists of the bitstream format header information and other static/mandatory information
within the AAC payload. The maximum AAC framesize allowed by the MPEG-4 standard
determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal
look-up table is used.

A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5
bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample
for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2.
For example for one channel with a sampling frequency of 48 kHz, the range from
48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC.

For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the
AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz.
HE-AAC v2 requires stereo input audio data.

Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are
appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo
audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio
quality at that bitrate than HE-AAC or HE-AAC v2.

\section reommendedConfig Recommended Sampling Rate and Bitrate Combinations

The following table provides an overview of recommended encoder configuration parameters
which we determined by virtue of numerous listening tests.

\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode.
\verbatim
-----------------------------------------------------------------------------------
Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
                   |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
                   |                  |                [kHz]  |      Rate  |
                   |                  |                       |     [kHz]  |
-------------------+------------------+-----------------------+------------+-------
AAC LC + SBR + PS  |   8000 -  11999  |         22.05, 24.00  |     24.00  |      2
AAC LC + SBR + PS  |  12000 -  17999  |                32.00  |     32.00  |      2
AAC LC + SBR + PS  |  18000 -  39999  |  32.00, 44.10, 48.00  |     44.10  |      2
AAC LC + SBR + PS  |  40000 -  56000  |  32.00, 44.10, 48.00  |     48.00  |      2
-------------------+------------------+-----------------------+------------+-------
AAC LC + SBR       |   8000 -  11999  |         22.05, 24.00  |     24.00  |      1
AAC LC + SBR       |  12000 -  17999  |                32.00  |     32.00  |      1
AAC LC + SBR       |  18000 -  39999  |  32.00, 44.10, 48.00  |     44.10  |      1
AAC LC + SBR       |  40000 -  56000  |  32.00, 44.10, 48.00  |     48.00  |      1
AAC LC + SBR       |  16000 -  27999  |  32.00, 44.10, 48.00  |     32.00  |      2
AAC LC + SBR       |  28000 -  63999  |  32.00, 44.10, 48.00  |     44.10  |      2
AAC LC + SBR       |  64000 - 128000  |  32.00, 44.10, 48.00  |     48.00  |      2
-------------------+------------------+-----------------------+------------+-------
AAC LC + SBR       |  64000 -  69999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
AAC LC + SBR       |  70000 - 159999  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
AAC LC + SBR       | 160000 - 245999  |  32.00, 44.10, 48.00  |     48.00  |      5
AAC LC + SBR       | 160000 - 265999  |  32.00, 44.10, 48.00  |     48.00  |    5.1
-------------------+------------------+-----------------------+------------+-------
AAC LC             |   8000 -  15999  | 11.025, 12.00, 16.00  |     12.00  |      1
AAC LC             |  16000 -  23999  |                16.00  |     16.00  |      1
AAC LC             |  24000 -  31999  |  16.00, 22.05, 24.00  |     24.00  |      1
AAC LC             |  32000 -  55999  |                32.00  |     32.00  |      1
AAC LC             |  56000 - 160000  |  32.00, 44.10, 48.00  |     44.10  |      1
AAC LC             | 160001 - 288000  |                48.00  |     48.00  |      1
-------------------+------------------+-----------------------+------------+-------
AAC LC             |  16000 -  23999  | 11.025, 12.00, 16.00  |     12.00  |      2
AAC LC             |  24000 -  31999  |                16.00  |     16.00  |      2
AAC LC             |  32000 -  39999  |  16.00, 22.05, 24.00  |     22.05  |      2
AAC LC             |  40000 -  95999  |                32.00  |     32.00  |      2
AAC LC             |  96000 - 111999  |  32.00, 44.10, 48.00  |     32.00  |      2
AAC LC             | 112000 - 320001  |  32.00, 44.10, 48.00  |     44.10  |      2
AAC LC             | 320002 - 576000  |                48.00  |     48.00  |      2
-------------------+------------------+-----------------------+------------+-------
AAC LC             | 160000 - 239999  |                32.00  |     32.00  | 5, 5.1
AAC LC             | 240000 - 279999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
AAC LC             | 280000 - 800000  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
-----------------------------------------------------------------------------------
\endverbatim \n

\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode.
\verbatim
-----------------------------------------------------------------------------------
Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
                   |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
                   |                  |                [kHz]  |      Rate  |
                   |                  |                       |     [kHz]  |
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          |  18000 -  24999  |        32.00 - 44.10  |     32.00  |      1
ELD + SBR          |  25000 -  31999  |        32.00 - 48.00  |     32.00  |      1
ELD + SBR          |  32000 -  64000  |        32.00 - 48.00  |     48.00  |      1
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          |  32000 -  51999  |        32.00 - 48.00  |     44.10  |      2
ELD + SBR          |  52000 - 128000  |        32.00 - 48.00  |     48.00  |      2
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          |  72000 - 160000  |        44.10 - 48.00  |     48.00  |      3
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          |  96000 - 212000  |        44.10 - 48.00  |     48.00  |      4
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          | 120000 - 246000  |        44.10 - 48.00  |     48.00  |      5
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          | 120000 - 266000  |        44.10 - 48.00  |     48.00  |    5.1
-------------------+------------------+-----------------------+------------+-------
LD, ELD            |  16000 -  19999  |        16.00 - 24.00  |     16.00  |      1
LD, ELD            |  20000 -  39999  |        16.00 - 32.00  |     24.00  |      1
LD, ELD            |  40000 -  49999  |        22.05 - 32.00  |     32.00  |      1
LD, ELD            |  50000 -  61999  |        24.00 - 44.10  |     32.00  |      1
LD, ELD            |  62000 -  84999  |        32.00 - 48.00  |     44.10  |      1
LD, ELD            |  85000 - 192000  |        44.10 - 48.00  |     48.00  |      1
-------------------+------------------+-----------------------+------------+-------
LD, ELD            |  64000 -  75999  |        24.00 - 32.00  |     32.00  |      2
LD, ELD            |  76000 -  97999  |        24.00 - 44.10  |     32.00  |      2
LD, ELD            |  98000 - 135999  |        32.00 - 48.00  |     44.10  |      2
LD, ELD            | 136000 - 384000  |        44.10 - 48.00  |     48.00  |      2
-------------------+------------------+-----------------------+------------+-------
LD, ELD            |  96000 - 113999  |        24.00 - 32.00  |     32.00  |      3
LD, ELD            | 114000 - 146999  |        24.00 - 44.10  |     32.00  |      3
LD, ELD            | 147000 - 203999  |        32.00 - 48.00  |     44.10  |      3
LD, ELD            | 204000 - 576000  |        44.10 - 48.00  |     48.00  |      3
-------------------+------------------+-----------------------+------------+-------
LD, ELD            | 128000 - 151999  |        24.00 - 32.00  |     32.00  |      4
LD, ELD            | 152000 - 195999  |        24.00 - 44.10  |     32.00  |      4
LD, ELD            | 196000 - 271999  |        32.00 - 48.00  |     44.10  |      4
LD, ELD            | 272000 - 768000  |        44.10 - 48.00  |     48.00  |      4
-------------------+------------------+-----------------------+------------+-------
LD, ELD            | 160000 - 189999  |        24.00 - 32.00  |     32.00  |      5
LD, ELD            | 190000 - 244999  |        24.00 - 44.10  |     32.00  |      5
LD, ELD            | 245000 - 339999  |        32.00 - 48.00  |     44.10  |      5
LD, ELD            | 340000 - 960000  |        44.10 - 48.00  |     48.00  |      5
-----------------------------------------------------------------------------------
\endverbatim \n

\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode.
\verbatim
-----------------------------------------------------------------------------------
Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
                   |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
                   |                  |                [kHz]  |      Rate  |
                   |                  |                       |     [kHz]  |
-------------------+------------------+-----------------------+------------+-------
ELD + SBR          |  18000 -  24999  |        16.00 - 22.05  |     22.05  |      1
(downsampled SBR)  |  25000 -  35999  |        22.05 - 32.00  |     24.00  |      1
                   |  36000 -  64000  |        32.00 - 48.00  |     32.00  |      1
-----------------------------------------------------------------------------------
\endverbatim \n


\page ENCODERBEHAVIOUR Encoder Behaviour

\section BEHAVIOUR_BANDWIDTH Bandwidth

The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth
according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the
file bandwidth.cpp (if available).

The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly.
\code
aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value);
\endcode

However it is not recommended to change these settings, because they are based on numerious listening
tests and careful tweaks to ensure the best overall encoding quality.

Theoretically a signal of for example 48 kHz can contain frequencies up to 24 kHz, but to use this full range
in an audio encoder usually does not make sense. Usually the encoder has a very limited amount of
bits to spend (typically 128 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would
waste a lot of these bits for frequencies the human ear is hardly able to perceive anyway, if at all. Hence it
is wise to use the available bits for the really important frequency range and just skip the rest.
At lower bitrates (e. g. <= 80 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller
bandwidth, because an encoded signal with smaller bandwidth and hence less artifacts sounds better than a signal
with higher bandwidth but then more coding artefacts across all frequencies. These artefacts would occur if
small bitrates and high bandwidths are chosen because the available bits are just not enough to encode all
frequencies well.

Unfortunately some people evaluate encoding quality based on possible bandwidth as well, but it is a two-sided
sword considering the trade-off described above.

Another aspect is workload consumption. The higher the allowed bandwidth, the more frequency lines have to be
processed, which in turn increases the workload.

\section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir

For AAC there is a difference between constant bit rate and constant frame
length due to the so-called bit reservoir technique, which allows the encoder to use less
bits in an AAC frame for those audio signal sections which are easy to encode,
and then spend them at a later point in
time for more complex audio sections. The extent to which this "bit exchange"
is done is limited to allow for reliable and relatively low delay real time
streaming.
Over a longer period in time the bitrate will be constant in the AAC constant
bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream
frame will in general have a different length in bytes but over time it
will reach the target bitrate. One could also make an MPEG compliant
AAC encoder which always produces constant length packages for each AAC frame,
but the audio quality would be considerably worse since the bit reservoir
technique would have to be switched off completely. A higher bit rate would have
to be used to get the same audio quality as with an enabled bit reservoir.

The maximum AAC frame length, regardless of the available bit reservoir, is defined
as 6144 bits per channel.

For mp3 by the way, the same bit reservoir technique exists, but there each bit
stream frame has a constant length for a given bit rate (ignoring the
padding byte). In mp3 there is a so-called "back pointer" which tells
the decoder which bits belong to the current mp3 frame - and in general some or
many bits have been transmitted in an earlier mp3 frame. Basically this leads to
the same "bit exchange between mp3 frames" as in AAC but with virtually constant
length frames.

This variable frame length at "constant bit rate" is not something special
in this Fraunhofer IIS AAC encoder. AAC has been designed in that way.

\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes

A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is
also one mode with 1920 samples per channel but this is only for special purposes
such as DAB+ digital radio).

The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is:

\f[
N\_FRAMES = 44100 / 2048 = 21.5332
\f]

At a bit rate of 8 kbps the average number of bits per frame \f$N\_BITS\_PER\_FRAME\f$ is:

\f[
N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52
\f]

which is about 46.44 bytes per encoded frame.

At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it is:

\f[
N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486
\f]

which is about 185.76 bytes per encoded frame.

These bits/frame figures are average figures where each AAC frame generally has a different
size in bytes. To calculate the same for AAC-LC just use 1024 instead of 2048 PCM samples per
frame and channel.
For AAC-LD/ELD it is either 480 or 512 PCM samples per frame and channel.


\section BEHAVIOUR_TOOLS Encoder Tools

The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools depending on the audio signal and
the encoder configuration (i.e. bitrate or AOT). It is not required to configure these tools manually.

PNS improves encoding quality only for certain bitrates. Therefore it makes sense to activate PNS only for
these bitrates and save the processing power required for PNS (about 10 % of the encoder) when using other
bitrates. This is done automatically inside the encoder library. PNS is disabled inside the encoder library if
an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature.

If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed,
the encoder deactivates PNS calculation internally.

*/

#ifndef _AAC_ENC_LIB_H_
#define _AAC_ENC_LIB_H_

#include "machine_type.h"
#include "FDK_audio.h"

#define AACENCODER_LIB_VL0 3
#define AACENCODER_LIB_VL1 4
#define AACENCODER_LIB_VL2 12

/**
 *  AAC encoder error codes.
 */
typedef enum {
    AACENC_OK                     = 0x0000,  /*!< No error happened. All fine. */

    AACENC_INVALID_HANDLE         = 0x0020,  /*!< Handle passed to function call was invalid. */
    AACENC_MEMORY_ERROR           = 0x0021,  /*!< Memory allocation failed. */
    AACENC_UNSUPPORTED_PARAMETER  = 0x0022,  /*!< Parameter not available. */
    AACENC_INVALID_CONFIG         = 0x0023,  /*!< Configuration not provided. */

    AACENC_INIT_ERROR             = 0x0040,  /*!< General initialization error. */
    AACENC_INIT_AAC_ERROR         = 0x0041,  /*!< AAC library initialization error. */
    AACENC_INIT_SBR_ERROR         = 0x0042,  /*!< SBR library initialization error. */
    AACENC_INIT_TP_ERROR          = 0x0043,  /*!< Transport library initialization error. */
    AACENC_INIT_META_ERROR        = 0x0044,  /*!< Meta data library initialization error. */

    AACENC_ENCODE_ERROR           = 0x0060,  /*!< The encoding process was interrupted by an unexpected error. */

    AACENC_ENCODE_EOF             = 0x0080   /*!< End of file reached. */

} AACENC_ERROR;


/**
 *  AAC encoder buffer descriptors identifier.
 *  This identifier are used within buffer descriptors AACENC_BufDesc::bufferIdentifiers.
 */
typedef enum {
    /* Input buffer identifier. */
    IN_AUDIO_DATA      = 0,                  /*!< Audio input buffer, interleaved INT_PCM samples. */
    IN_ANCILLRY_DATA   = 1,                  /*!< Ancillary data to be embedded into bitstream. */
    IN_METADATA_SETUP  = 2,                  /*!< Setup structure for embedding meta data. */

    /* Output buffer identifier. */
    OUT_BITSTREAM_DATA = 3,                  /*!< Buffer holds bitstream output data. */
    OUT_AU_SIZES       = 4                   /*!< Buffer contains sizes of each access unit. This information
                                                  is necessary for superframing. */

} AACENC_BufferIdentifier;


/**
 *  AAC encoder handle.
 */
typedef struct AACENCODER *HANDLE_AACENCODER;


/**
 *  Provides some info about the encoder configuration.
 */
typedef struct {

    UINT                maxOutBufBytes;      /*!< Maximum number of encoder bitstream bytes within one frame.
                                                  Size depends on maximum number of supported channels in encoder instance.
                                                  For superframing (as used for example in DAB+), size has to be a multiple accordingly. */

    UINT                maxAncBytes;         /*!< Maximum number of ancillary data bytes which can be inserted into
                                                  bitstream within one frame. */

    UINT                inBufFillLevel;      /*!< Internal input buffer fill level in samples per channel. This parameter
                                                  will automatically be cleared if samplingrate or channel(Mode/Order) changes. */

    UINT                inputChannels;       /*!< Number of input channels expected in encoding process. */

    UINT                frameLength;         /*!< Amount of input audio samples consumed each frame per channel, depending
                                                  on audio object type configuration. */

    UINT                encoderDelay;        /*!< Codec delay in PCM samples/channel. Depends on framelength and AOT. Does not
                                                  include framing delay for filling up encoder PCM input buffer. */

    UCHAR               confBuf[64];         /*!< Configuration buffer in binary format as an AudioSpecificConfig
                                                  or StreamMuxConfig according to the selected transport type. */

    UINT                confSize;            /*!< Number of valid bytes in confBuf. */

} AACENC_InfoStruct;


/**
 *  Describes the input and output buffers for an aacEncEncode() call.
 */
typedef struct {
    INT                 numBufs;             /*!< Number of buffers. */
    void              **bufs;                /*!< Pointer to vector containing buffer addresses. */
    INT                *bufferIdentifiers;   /*!< Identifier of each buffer element. See ::AACENC_BufferIdentifier. */
    INT                *bufSizes;            /*!< Size of each buffer in 8-bit bytes. */
    INT                *bufElSizes;          /*!< Size of each buffer element in bytes. */

} AACENC_BufDesc;


/**
 *  Defines the input arguments for an aacEncEncode() call.
 */
typedef struct {
    INT                 numInSamples;        /*!< Number of valid input audio samples (multiple of input channels). */
    INT                 numAncBytes;         /*!< Number of ancillary data bytes to be encoded. */

} AACENC_InArgs;


/**
 *  Defines the output arguments for an aacEncEncode() call.
 */
typedef struct {
    INT                 numOutBytes;         /*!< Number of valid bitstream bytes generated during aacEncEncode(). */
    INT                 numInSamples;        /*!< Number of input audio samples consumed by the encoder. */
    INT                 numAncBytes;         /*!< Number of ancillary data bytes consumed by the encoder. */

} AACENC_OutArgs;


/**
 *  Meta Data Compression Profiles.
 */
typedef enum {
    AACENC_METADATA_DRC_NONE          = 0,   /*!< None. */
    AACENC_METADATA_DRC_FILMSTANDARD  = 1,   /*!< Film standard. */
    AACENC_METADATA_DRC_FILMLIGHT     = 2,   /*!< Film light. */
    AACENC_METADATA_DRC_MUSICSTANDARD = 3,   /*!< Music standard. */
    AACENC_METADATA_DRC_MUSICLIGHT    = 4,   /*!< Music light. */
    AACENC_METADATA_DRC_SPEECH        = 5    /*!< Speech. */

} AACENC_METADATA_DRC_PROFILE;


/**
 *  Meta Data setup structure.
 */
typedef struct {

  AACENC_METADATA_DRC_PROFILE drc_profile;             /*!< MPEG DRC compression profile. See ::AACENC_METADATA_DRC_PROFILE. */
  AACENC_METADATA_DRC_PROFILE comp_profile;            /*!< ETSI heavy compression profile. See ::AACENC_METADATA_DRC_PROFILE. */

  INT                         drc_TargetRefLevel;      /*!< Used to define expected level to:
                                                            Scaled with 16 bit. x*2^16. */
  INT                         comp_TargetRefLevel;     /*!< Adjust limiter to avoid overload.
                                                            Scaled with 16 bit. x*2^16. */

  INT                         prog_ref_level_present;  /*!< Flag, if prog_ref_level is present */
  INT                         prog_ref_level;          /*!< Programme Reference Level = Dialogue Level:
                                                            -31.75dB .. 0 dB ; stepsize: 0.25dB
                                                            Scaled with 16 bit. x*2^16.*/

  UCHAR                       PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in programme config element */
  UCHAR                       ETSI_DmxLvl_present;     /*!< Flag, if dmx-lvl should be written in ETSI-ancData */

  SCHAR                       centerMixLevel;          /*!< Center downmix level (0...7, according to table) */
  SCHAR                       surroundMixLevel;        /*!< Surround downmix level (0...7, according to table) */

  UCHAR                       dolbySurroundMode;       /*!< Indication for Dolby Surround Encoding Mode.
                                                            - 0: Dolby Surround mode not indicated
                                                            - 1: 2-ch audio part is not Dolby surround encoded
                                                            - 2: 2-ch audio part is Dolby surround encoded */
} AACENC_MetaData;


/**
 * AAC encoder control flags.
 *
 * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to get information about the internal
 * initialization process. It is also possible to overwrite the internal state from extern when necessary.
 */
typedef enum
{
    AACENC_INIT_NONE              = 0x0000,  /*!< Do not trigger initialization. */
    AACENC_INIT_CONFIG            = 0x0001,  /*!< Initialize all encoder modules configuration. */
    AACENC_INIT_STATES            = 0x0002,  /*!< Reset all encoder modules history buffer. */
    AACENC_INIT_TRANSPORT         = 0x1000,  /*!< Initialize transport lib with new parameters. */
    AACENC_RESET_INBUFFER         = 0x2000,  /*!< Reset fill level of internal input buffer. */
    AACENC_INIT_ALL               = 0xFFFF   /*!< Initialize all. */
}
AACENC_CTRLFLAGS;


/**
 * \brief  AAC encoder setting parameters.
 *
 * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() function to read
 * the internal status of the following parameters.
 */
typedef enum
{
  AACENC_AOT                      = 0x0100,  /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h.
                                                  - 2: MPEG-4 AAC Low Complexity.
                                                  - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication (HE-AAC).
                                                  - 29: MPEG-4 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
                                                        This configuration can be used only with stereo input audio data.
                                                  - 23: MPEG-4 AAC Low-Delay.
                                                  - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in
                                                        combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter.
                                                  - 129: MPEG-2 AAC Low Complexity.
                                                  - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC).
                                                  - 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
                                                         This configuration can be used only with stereo input audio data. */

  AACENC_BITRATE                  = 0x0101,  /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
                                                  - CBR: Bitrate in bits/second.
                                                    See \ref suppBitrates for details. */

  AACENC_BITRATEMODE              = 0x0102,  /*!< Bitrate mode. Configuration can be different kind of bitrate configurations:
                                                  - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default)
                                                       Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir.
                                                       In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small.
                                                  - 8: LD/ELD full bitreservoir for packet based transmission. */

  AACENC_SAMPLERATE               = 0x0103,  /*!< Audio input data sampling rate. Encoder supports following sampling rates:
                                                  8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */

  AACENC_SBR_MODE                 = 0x0104,  /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE.
                                                  This parameter is for ELD audio object type only.
                                                  - -1: Use ELD SBR auto configurator (default).
                                                  - 0: Disable Spectral Band Replication.
                                                  - 1: Enable Spectral Band Replication. */

  AACENC_GRANULE_LENGTH           = 0x0105,  /*!< Core encoder (AAC) audio frame length in samples:
                                                  - 1024: Default configuration.
                                                  - 960: DRM/DAB+.
                                                  - 512: Default LD/ELD configuration.
                                                  - 480: Optional length in LD/ELD configuration. */

  AACENC_CHANNELMODE              = 0x0106,  /*!< Set explicit channel mode. Channel mode must match with number of input channels.
                                                  - 1-7 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */

  AACENC_CHANNELORDER             = 0x0107,  /*!< Input audio data channel ordering scheme:
                                                  - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default)
                                                  - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */

  AACENC_SBR_RATIO                = 0x0108,  /*!<  Controls activation of downsampled SBR. With downsampled SBR, the delay will be
                                                   shorter. On the other hand, for achieving the same quality level, downsampled SBR
                                                   needs more bits than dual-rate SBR.
                                                   With downsampled SBR, the AAC encoder will work at the same sampling rate as the
                                                   SBR encoder (single rate).
                                                   Downsampled SBR is supported for AAC-ELD and HE-AACv1.
                                                   - 1: Downsampled SBR (default for ELD).
                                                   - 2: Dual-rate SBR   (default for HE-AAC). */

  AACENC_AFTERBURNER              = 0x0200,  /*!< This parameter controls the use of the afterburner feature.
                                                  The afterburner is a type of analysis by synthesis algorithm which increases the
                                                  audio quality but also the required processing power. It is recommended to always
                                                  activate this if additional memory consumption and processing power consumption
                                                  is not a problem. If increased MHz and memory consumption are an issue then the MHz
                                                  and memory cost of this optional module need to be evaluated against the improvement
                                                  in audio quality on a case by case basis.
                                                  - 0: Disable afterburner (default).
                                                  - 1: Enable afterburner. */

  AACENC_BANDWIDTH                = 0x0203,  /*!< Core encoder audio bandwidth:
                                                  - 0: Determine bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH).
                                                  - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
                                                               touch this value to avoid degraded audio quality) */

  AACENC_TRANSMUX                 = 0x0300,  /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
                                                  types can be configured in encoder library:
                                                  - 0: raw access units
                                                  - 1: ADIF bitstream format
                                                  - 2: ADTS bitstream format
                                                  - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1
                                                  - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig
                                                  - 10: Audio Sync Stream (LOAS) */

  AACENC_HEADER_PERIOD            = 0x0301,  /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS
                                                  transport layer. Additionally this parameter configures the PCE repetition period
                                                  in raw_data_block(). See \ref encPCE.
                                                  - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0.
                                                  - n: Frame count period. */

  AACENC_SIGNALING_MODE           = 0x0302,  /*!< Signaling mode of the extension AOT:
                                                  - 0: Implicit backward compatible signaling (default for non-MPEG-4 based
                                                       AOT's and for the transport formats ADIF and ADTS)
                                                       - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
                                                       - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
                                                       - This method works with all transport formats
                                                       - This method does not work with downsampled SBR
                                                  - 1: Explicit backward compatible signaling
                                                       - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
                                                       - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
                                                       - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part.
                                                         If the stream contained PS, the result will be a a decoded mono downmix
                                                       - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1
                                                       - This method does work with downsampled SBR
                                                  - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS)
                                                       - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders
                                                       - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling
                                                       - A decoder not capable of decoding PS will not decode the stream at all if it contained PS
                                                       - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format
                                                       - This method does work with downsampled SBR

                                                   For making sure that the listener always experiences the best audio quality,
                                                   explicit hierarchical signaling should be used.
                                                   This makes sure that only a full HE-AAC-capable decoder will decode those streams.
                                                   The audio is played at full bandwidth.
                                                   For best backwards compatibility, it is recommended to encode with implicit SBR signaling.
                                                   A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded
                                                   audio will sound band-limited.

                                                   For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible.

                                                   For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1.
                                                   The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC.
                                                   A decoder that is only capable of decoding AAC-LC will skip this part.
                                                   Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length.
                                                   Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the
                                                   length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a
                                                   LOAS/LATM stream that was being encoded with AudioMuxVersion==0.

                                                   For downsampled SBR, explicit signaling is mandatory. The reason for this is that the
                                                   extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part)
                                                   can only be signaled in explicit mode.

                                                   For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the
                                                   AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/

  AACENC_TPSUBFRAMES              = 0x0303,  /*!< Number of sub frames in a transport frame for LOAS/LATM, DAB+ or ADTS (default 1).
                                                  - ADTS: Maximum number of sub frames restricted to 4.
                                                  - DAB+: Maximum number of sub frames restricted to 6.
                                                  - LOAS/LATM: Maximum number of sub frames restricted to 2.*/

  AACENC_PROTECTION               = 0x0306,  /*!< Configure protection in tranpsort layer:
                                                  - 0: No protection. (default)
                                                  - 1: CRC active for ADTS bitstream format. */

  AACENC_ANCILLARY_BITRATE        = 0x0500,  /*!< Constant ancillary data bitrate in bits/second.
                                                  - 0: Either no ancillary data or insert exact number of bytes, denoted via
                                                       input parameter, numAncBytes in AACENC_InArgs.
                                                  - else: Insert ancillary data with specified bitrate. */

  AACENC_METADATA_MODE            = 0x0600,  /*!< Configure Meta Data. See ::AACENC_MetaData for further details:
                                                  - 0: Do not embed any metadata.
                                                  - 1: Embed MPEG defined metadata only.
                                                  - 2: Embed all metadata. */

  AACENC_CONTROL_STATE            = 0xFF00,  /*!< There is an automatic process which internally reconfigures the encoder instance
                                                  when a configuration parameter changed or an error occured. This paramerter allows
                                                  overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */

  AACENC_NONE                     = 0xFFFF   /*!< ------ */

} AACENC_PARAM;


#ifdef __cplusplus
extern "C" {
#endif

/**
 * \brief  Open an instance of the encoder.
 *
 * Allocate memory for an encoder instance with a functional range denoted by the function parameters.
 * Preinitialize encoder instance with default configuration.
 *
 * \param phAacEncoder  A pointer to an encoder handle. Initialized on return.
 * \param encModules    Specify encoder modules to be supported in this encoder instance:
 *                      - 0x0: Allocate memory for all available encoder modules.
 *                      - else: Select memory allocation regarding encoder modules. Following flags are possible and can be combined.
 *                              - 0x01: AAC module.
 *                              - 0x02: SBR module.
 *                              - 0x04: PS module.
 *                              - 0x10: Metadata module.
 *                              - example: (0x01|0x02|0x04|0x10) allocates all modules and is equivalent to default configuration denotet by 0x0.
 * \param maxChannels   Number of channels to be allocated. This parameter can be used in different ways:
 *                      - 0: Allocate maximum number of AAC and SBR channels as supported by the library.
 *                      - nChannels: Use same maximum number of channels for allocating memory in AAC and SBR module.
 *                      - nChannels | (nSbrCh<<8): Number of SBR channels can be different to AAC channels to save data memory.
 *
 * \return
 *          - AACENC_OK, on succes.
 *          - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, on failure.
 */
AACENC_ERROR aacEncOpen(
        HANDLE_AACENCODER        *phAacEncoder,
        const UINT                encModules,
        const UINT                maxChannels
        );


/**
 * \brief  Close the encoder instance.
 *
 * Deallocate encoder instance and free whole memory.
 *
 * \param phAacEncoder  Pointer to the encoder handle to be deallocated.
 *
 * \return
 *          - AACENC_OK, on success.
 *          - AACENC_INVALID_HANDLE, on failure.
 */
AACENC_ERROR aacEncClose(
        HANDLE_AACENCODER        *phAacEncoder
        );


/**
 * \brief Encode audio data.
 *
 * This function is mainly for encoding audio data. In addition the function can be used for an encoder (re)configuration
 * process.
 * - PCM input data will be retrieved from external input buffer until the fill level allows encoding a single frame.
 *   This functionality allows an external buffer with reduced size in comparison to the AAC or HE-AAC audio frame length.
 * - If the value of the input samples argument is zero, just internal reinitialization will be applied if it is
 *   requested.
 * - At the end of a file the flushing process can be triggerd via setting the value of the input samples argument to -1.
 *   The encoder delay lines are fully flushed when the encoder returns no valid bitstream data AACENC_OutArgs::numOutBytes.
 *   Furthermore the end of file is signaled by the return value AACENC_ENCODE_EOF.
 * - If an error occured in the previous frame or any of the encoder parameters changed, an internal reinitialization
 *   process will be applied before encoding the incoming audio samples.
 * - The function can also be used for an independent reconfiguration process without encoding. The first parameter has to be a
 *   valid encoder handle and all other parameters can be set to NULL.
 * - If the size of the external bitbuffer in outBufDesc is not sufficient for writing the whole bitstream, an internal
 *   error will be the return value and a reconfiguration will be triggered.
 *
 * \param hAacEncoder           A valid AAC encoder handle.
 * \param inBufDesc             Input buffer descriptor, see AACENC_BufDesc:
 *                              - At least one input buffer with audio data is expected.
 *                              - Optionally a second input buffer with ancillary data can be fed.
 * \param outBufDesc            Output buffer descriptor, see AACENC_BufDesc:
 *                              - Provide one output buffer for the encoded bitstream.
 * \param inargs                Input arguments, see AACENC_InArgs.
 * \param outargs               Output arguments, AACENC_OutArgs.
 *
 * \return
 *          - AACENC_OK, on success.
 *          - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding process.
 *          - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR,
 *            AACENC_INIT_META_ERROR, on failure in encoder initialization.
 *          - AACENC_ENCODE_EOF, when flushing fully concluded.
 */
AACENC_ERROR aacEncEncode(
        const HANDLE_AACENCODER   hAacEncoder,
        const AACENC_BufDesc     *inBufDesc,
        const AACENC_BufDesc     *outBufDesc,
        const AACENC_InArgs      *inargs,
        AACENC_OutArgs           *outargs
        );


/**
 * \brief  Acquire info about present encoder instance.
 *
 * This function retrieves information of the encoder configuration. In addition to informative internal states,
 * a configuration data block of the current encoder settings will be returned. The format is either Audio Specific Config
 * in case of Raw Packets transport format or StreamMuxConfig in case of LOAS/LATM transport format. The configuration
 * data block is binary coded as specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 File Format
 * or RFC3016 or RFC3640 applications.
 *
 * \param hAacEncoder           A valid AAC encoder handle.
 * \param pInfo                 Pointer to AACENC_InfoStruct. Filled on return.
 *
 * \return
 *          - AACENC_OK, on succes.
 *          - AACENC_INIT_ERROR, on failure.
 */
AACENC_ERROR aacEncInfo(
        const HANDLE_AACENCODER   hAacEncoder,
        AACENC_InfoStruct        *pInfo
        );


/**
 * \brief  Set one single AAC encoder parameter.
 *
 * This function allows configuration of all encoder parameters specified in ::AACENC_PARAM. Each parameter must be
 * set with a separate function call. An internal validation of the configuration value range will be done and an
 * internal reconfiguration will be signaled. The actual configuration adoption is part of the subsequent aacEncEncode() call.
 *
 * \param hAacEncoder           A valid AAC encoder handle.
 * \param param                 Parameter to be set. See ::AACENC_PARAM.
 * \param value                 Parameter value. See parameter description in ::AACENC_PARAM.
 *
 * \return
 *          - AACENC_OK, on success.
 *          - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, AACENC_INVALID_CONFIG, on failure.
 */
AACENC_ERROR aacEncoder_SetParam(
        const HANDLE_AACENCODER   hAacEncoder,
        const AACENC_PARAM        param,
        const UINT                value
        );


/**
 * \brief  Get one single AAC encoder parameter.
 *
 * This function is the complement to aacEncoder_SetParam(). After encoder reinitialization with user defined settings,
 * the internal status can be obtained of each parameter, specified with ::AACENC_PARAM.
 *
 * \param hAacEncoder           A valid AAC encoder handle.
 * \param param                 Parameter to be returned. See ::AACENC_PARAM.
 *
 * \return  Internal configuration value of specifed parameter ::AACENC_PARAM.
 */
UINT aacEncoder_GetParam(
        const HANDLE_AACENCODER   hAacEncoder,
        const AACENC_PARAM        param
        );


/**
 * \brief  Get information about encoder library build.
 *
 * Fill a given LIB_INFO structure with library version information.
 *
 * \param info  Pointer to an allocated LIB_INFO struct.
 *
 * \return
 *          - AACENC_OK, on success.
 *          - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
 */
AACENC_ERROR aacEncGetLibInfo(
        LIB_INFO                 *info
        );


#ifdef __cplusplus
}
#endif

#endif   /* _AAC_ENC_LIB_H_ */