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/* -----------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
Forschung e.V. All rights reserved.
1. INTRODUCTION
The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
scheme for digital audio. This FDK AAC Codec software is intended to be used on
a wide variety of Android devices.
AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
general perceptual audio codecs. AAC-ELD is considered the best-performing
full-bandwidth communications codec by independent studies and is widely
deployed. AAC has been standardized by ISO and IEC as part of the MPEG
specifications.
Patent licenses for necessary patent claims for the FDK AAC Codec (including
those of Fraunhofer) may be obtained through Via Licensing
(www.vialicensing.com) or through the respective patent owners individually for
the purpose of encoding or decoding bit streams in products that are compliant
with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
Android devices already license these patent claims through Via Licensing or
directly from the patent owners, and therefore FDK AAC Codec software may
already be covered under those patent licenses when it is used for those
licensed purposes only.
Commercially-licensed AAC software libraries, including floating-point versions
with enhanced sound quality, are also available from Fraunhofer. Users are
encouraged to check the Fraunhofer website for additional applications
information and documentation.
2. COPYRIGHT LICENSE
Redistribution and use in source and binary forms, with or without modification,
are permitted without payment of copyright license fees provided that you
satisfy the following conditions:
You must retain the complete text of this software license in redistributions of
the FDK AAC Codec or your modifications thereto in source code form.
You must retain the complete text of this software license in the documentation
and/or other materials provided with redistributions of the FDK AAC Codec or
your modifications thereto in binary form. You must make available free of
charge copies of the complete source code of the FDK AAC Codec and your
modifications thereto to recipients of copies in binary form.
The name of Fraunhofer may not be used to endorse or promote products derived
from this library without prior written permission.
You may not charge copyright license fees for anyone to use, copy or distribute
the FDK AAC Codec software or your modifications thereto.
Your modified versions of the FDK AAC Codec must carry prominent notices stating
that you changed the software and the date of any change. For modified versions
of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
AAC Codec Library for Android."
3. NO PATENT LICENSE
NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
Fraunhofer provides no warranty of patent non-infringement with respect to this
software.
You may use this FDK AAC Codec software or modifications thereto only for
purposes that are authorized by appropriate patent licenses.
4. DISCLAIMER
This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
including but not limited to the implied warranties of merchantability and
fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
or consequential damages, including but not limited to procurement of substitute
goods or services; loss of use, data, or profits, or business interruption,
however caused and on any theory of liability, whether in contract, strict
liability, or tort (including negligence), arising in any way out of the use of
this software, even if advised of the possibility of such damage.
5. CONTACT INFORMATION
Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany
www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------- */
/**************************** AAC encoder library ******************************
Author(s): M. Lohwasser
Description:
*******************************************************************************/
/**
* \file aacenc_lib.h
* \brief FDK AAC Encoder library interface header file.
*
\mainpage 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 in aacenc_lib.h.
\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 uses
file-based input/output.
-# Call aacEncOpen() to allocate encoder instance with required \ref encOpen
"configuration". \code HANDLE_AACENCODER hAacEncoder = NULL; if ( (ErrorStatus =
aacEncOpen(&hAacEncoder,0,0)) != AACENC_OK ) { \endcode
-# 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. \code ErrorStatus =
aacEncEncode(hAacEncoder, NULL, NULL, NULL, NULL); \endcode
-# Call aacEncInfo() to retrieve a configuration data block to be transmitted
out of band. This is required when using RFC3640 or RFC3016 like transport.
\code
AACENC_InfoStruct encInfo;
aacEncInfo(hAacEncoder, &encInfo);
\endcode
-# Encode input audio data in loop.
\code
do
{
\endcode
Feed \ref feedInBuf "input buffer" with new audio data and provide input/output
\ref bufDes "arguments" to aacEncEncode(). \code ErrorStatus =
aacEncEncode(hAacEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs); \endcode
Write \ref writeOutData "output data" to file or audio device.
\code
} while (ErrorStatus==AACENC_OK);
\endcode
-# Call aacEncClose() and destroy encoder instance.
\code
aacEncClose(&hAacEncoder);
\endcode
\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(). \code static INT_PCM
inputBuffer[8*2048]; static UCHAR ancillaryBuffer[50]; static
AACENC_MetaData metaDataSetup; static UCHAR outputBuffer[8192];
\endcode
All input and output buffer must be clustered in input and output buffer arrays.
\code
static void* inBuffer[] = { inputBuffer, ancillaryBuffer, &metaDataSetup
}; static INT inBufferIds[] = { IN_AUDIO_DATA, IN_ANCILLRY_DATA,
IN_METADATA_SETUP }; static INT inBufferSize[] = { sizeof(inputBuffer),
sizeof(ancillaryBuffer), sizeof(metaDataSetup) }; static INT inBufferElSize[]
= { sizeof(INT_PCM), sizeof(UCHAR), sizeof(AACENC_MetaData) };
static void* outBuffer[] = { outputBuffer };
static INT outBufferIds[] = { OUT_BITSTREAM_DATA };
static INT outBufferSize[] = { sizeof(outputBuffer) };
static INT outBufferElSize[] = { sizeof(UCHAR) };
\endcode
Allocate buffer descriptors
\code
AACENC_BufDesc inBufDesc;
AACENC_BufDesc outBufDesc;
\endcode
Initialize input buffer descriptor
\code
inBufDesc.numBufs = sizeof(inBuffer)/sizeof(void*);
inBufDesc.bufs = (void**)&inBuffer;
inBufDesc.bufferIdentifiers = inBufferIds;
inBufDesc.bufSizes = inBufferSize;
inBufDesc.bufElSizes = inBufferElSize;
\endcode
Initialize output buffer descriptor
\code
outBufDesc.numBufs = sizeof(outBuffer)/sizeof(void*);
outBufDesc.bufs = (void**)&outBuffer;
outBufDesc.bufferIdentifiers = outBufferIds;
outBufDesc.bufSizes = outBufferSize;
outBufDesc.bufElSizes = outBufferElSize;
\endcode
\subsection argLists Provide Input/Output Argument Lists
The input and output arguments of an aacEncEncode() call are described in
argument structures. \code AACENC_InArgs inargs; AACENC_OutArgs outargs;
\endcode
\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). \code inargs.numInSamples
+= WAV_InputRead ( wavIn, &inputBuffer[inargs.numInSamples],
FDKmin(encInfo.inputChannels*encInfo.frameLength,
sizeof(inputBuffer) /
sizeof(INT_PCM)-inargs.numInSamples),
SAMPLE_BITS
);
\endcode
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: \code if (outargs.numInSamples>0) {
FDKmemmove( inputBuffer,
&inputBuffer[outargs.numInSamples],
sizeof(INT_PCM)*(inargs.numInSamples-outargs.numInSamples) );
inargs.numInSamples -= outargs.numInSamples;
}
\endcode
\section writeOutData Output Bitstream Data
If any AAC bitstream data is available, write it to output file or device as
follows. \code if (outargs.numOutBytes>0) { FDKfwrite(outputBuffer,
outargs.numOutBytes, 1, pOutFile);
}
\endcode
\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, 0-3); \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. \code CHANNEL_MODE chMode =
MODE_INVALID;
switch (nChannels) {
case 1: chMode = MODE_1; break;
case 2: chMode = MODE_2; break;
case 3: chMode = MODE_1_2; break;
case 4: chMode = MODE_1_2_1; break;
case 5: chMode = MODE_1_2_2; break;
case 6: chMode = MODE_1_2_2_1; break;
case 7: chMode = MODE_6_1; break;
case 8: chMode = MODE_7_1_BACK; break;
default:
chMode = MODE_INVALID;
}
return chMode;
\endcode
\subsection peakbitrate Peak Bitrate Configuration
In AAC, the default bitreservoir configuration depends on the chosen bitrate per
frame and the number of effective channels. The size can be determined as below.
\f[
bitreservoir = nEffChannels*6144 - (bitrate*framelength/samplerate)
\f]
Due to audio quality concerns it is not recommended to change the bitreservoir
size to a lower value than the default setting! However, for minimizing the
delay for streaming applications or for achieving a constant size of the
bitstream packages in each frame, it may be necessaray to limit the maximum bits
per frame size. This can be done with the ::AACENC_PEAK_BITRATE parameter. \code
aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value);
\endcode
To achieve acceptable audio quality with a reduced bitreservoir size setting at
least 1000 bits per audio channel is recommended. For a multichannel audio file
with 5.1 channels the bitreservoir reduced to 5000 bits results in acceptable
audio quality.
\subsection vbrmode Variable Bitrate Mode
The encoder provides various Variable Bitrate Modes that differ in audio quality
and average overall bitrate. The given values are averages over time, different
encoder settings and strongly depend on the type of audio signal. The VBR
configurations can be adjusted via ::AACENC_BITRATEMODE encoder parameter.
\verbatim
--------------------------------------------
VBR_MODE | Approx. Bitrate in kbps/channel
| AAC-LC | AAC-LD/AC_ELD
----------+---------------+-----------------
VBR_1 | 32 - 48 | 32 - 56
VBR_2 | 40 - 56 | 40 - 64
VBR_3 | 48 - 64 | 48 - 72
VBR_4 | 64 - 80 | 64 - 88
VBR_5 | 96 - 120 | 112 - 144
--------------------------------------------
\endverbatim
The bitrate ranges apply for individual audio channels. In case of multichannel
configurations the average bitrate might be estimated by multiplying with the
number of effective channels. This corresponds to all audio input channels
exclusively the low frequency channel. At configurations which are making use of
downmix modules the AAC core channels respectively downmix channels shall be
considered. For ::AACENC_AOT which are using SBR, the average bitrate can be
estimated by using the ratio of 0.5 for dualrate SBR and 0.75 for downsampled
SBR configurations.
\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, 0/1); \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
explicitly.
Based on sampling rate and chosen bitrate a reasonable SBR configuration will be
used. \verbatim
------------------------------------------------------------------
Sampling Rate | Total Bitrate | No. of | SBR | SBR Ratio
[kHz] | [bit/s] | Chan | |
| | | |
---------------+-----------------+--------+-----+-----------------
]min, 16[ | min - max | 1 | off | ---
---------------+-----------------+--------------+-----------------
[16] | min - 27999 | 1 | on | downsampled SBR
| 28000 - max | 1 | off | ---
---------------+-----------------+--------------+-----------------
]16 - 24] | min - 39999 | 1 | on | downsampled SBR
| 40000 - max | 1 | off | ---
---------------+-----------------+--------------+-----------------
]24 - 32] | min - 27999 | 1 | on | dualrate SBR
| 28000 - 55999 | 1 | on | downsampled SBR
| 56000 - max | 1 | off | ---
---------------+-----------------+--------------+-----------------
]32 - 44.1] | min - 63999 | 1 | on | dualrate SBR
| 64000 - max | 1 | off | ---
---------------+-----------------+--------------+-----------------
]44.1 - 48] | min - 63999 | 1 | on | dualrate SBR
| 64000 - max | 1 | off | ---
| | | |
---------------+-----------------+--------+-----+-----------------
]min, 16[ | min - max | 2 | off | ---
---------------+-----------------+--------------+-----------------
[16] | min - 31999 | 2 | on | downsampled SBR
| 32000 - 63999 | 2 | on | downsampled SBR
| 64000 - max | 2 | off | ---
---------------+-----------------+--------------+-----------------
]16 - 24] | min - 47999 | 2 | on | downsampled SBR
| 48000 - 79999 | 2 | on | downsampled SBR
| 80000 - max | 2 | off | ---
---------------+-----------------+--------------+-----------------
]24 - 32] | min - 31999 | 2 | on | dualrate SBR
| 32000 - 67999 | 2 | on | dualrate SBR
| 68000 - 95999 | 2 | on | downsampled SBR
| 96000 - max | 2 | off | ---
---------------+-----------------+--------------+-----------------
]32 - 44.1] | min - 43999 | 2 | on | dualrate SBR
| 44000 - 127999 | 2 | on | dualrate SBR
| 128000 - max | 2 | off | ---
---------------+-----------------+--------------+-----------------
]44.1 - 48] | min - 43999 | 2 | on | dualrate SBR
| 44000 - 127999 | 2 | on | dualrate SBR
| 128000 - max | 2 | off | ---
| | |
------------------------------------------------------------------
\endverbatim
\subsection encDsELD Reduced Delay (Downscaled) Mode
The downscaled mode of AAC-ELD reduces the algorithmic delay of AAC-ELD by
virtually increasing the sampling rate. When using the downscaled mode, the
bitrate should be increased for keeping the same audio quality level. For common
signals, the bitrate should be increased by 25% for a downscale factor of 2.
Currently, downscaling factors 2 and 4 are supported.
To enable the downscaled mode in the encoder, the framelength parameter
AACENC_GRANULE_LENGTH must be set accordingly to 256 or 240 for a downscale
factor of 2 or 128 or 120 for a downscale factor of 4. The default values of 512
or 480 mean that no downscaling is applied. \code
aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 256);
aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 128);
\endcode
Downscaled bitstreams are fully backwards compatible. However, the legacy
decoder needs to support high sample rate, e.g. 96kHz. The signaled sampling
rate is multiplied by the downscale factor. Although not required, downscaling
should be applied when decoding downscaled bitstreams. It reduces CPU workload
and the output will have the same sampling rate as the input. In an ideal
configuration both encoder and decoder should run with the same downscale
factor.
The following table shows approximate filter bank delays in ms for common
sampling rates(sr) at framesize(fs), and downscale factor(dsf), based on this
formula: \f[ 1000 * fs / (dsf * sr) \f]
\verbatim
--------------------------------------
| 512/2 | 512/4 | 480/2 | 480/4
------+-------+-------+-------+-------
22050 | 17.41 | 8.71 | 16.33 | 8.16
32000 | 12.00 | 6.00 | 11.25 | 5.62
44100 | 8.71 | 4.35 | 8.16 | 4.08
48000 | 8.00 | 4.00 | 7.50 | 3.75
--------------------------------------
\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 and 11,12,14 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 | Height | front_El | side_El | back_El |
lfe_El
-----------------------+-------+--------+---------------+----------+----------+---------
MODE_1 | 1 | NORM | SCE | | |
MODE_2 | 2 | NORM | CPE | | |
MODE_1_2 | 3 | NORM | SCE, CPE | | |
MODE_1_2_1 | 4 | NORM | SCE, CPE | | SCE |
MODE_1_2_2 | 5 | NORM | SCE, CPE | | CPE |
MODE_1_2_2_1 | 6 | NORM | SCE, CPE | | CPE |
LFE MODE_1_2_2_2_1 | 7 | NORM | SCE, CPE, CPE | | CPE
| LFE MODE_6_1 | 11 | NORM | SCE, CPE | | CPE,
SCE | LFE MODE_7_1_BACK | 12 | NORM | SCE, CPE | |
CPE, CPE | LFE
-----------------------+-------+--------+---------------+----------+----------+---------
MODE_7_1_TOP_FRONT | 14 | NORM | SCE, CPE | | CPE |
LFE | | TOP | CPE | | |
-----------------------+-------+--------+---------------+----------+----------+---------
MODE_7_1_REAR_SURROUND | 0 | NORM | SCE, CPE | | CPE, CPE |
LFE MODE_7_1_FRONT_CENTER | 0 | NORM | SCE, CPE, CPE | | CPE
| LFE
----------------------------------------------------------------------------------------
- NORM: Normal Height Layer. - TOP: Top Height Layer. - BTM: Bottom Height
Layer.
- SCE: Single Channel Element. - CPE: Channel Pair. - LFE: Low Frequency
Element. \endverbatim
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 in the normal height layer, possibly followed by
front, side, and back elements in the top and bottom layer (Channel
Configuration 14). \n This mapping of Audio Channel Elements is defined in MPEG
standard for Channel Config 1-7 and 11,12,14.\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_6_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 |
| 2 | 0 | 1 | 4 | 5 | 6 | 3 | MODE_7_1_BACK | 0 | 1 | 2 | 3 | 4 | 5 | 6
| 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 MODE_7_1_TOP_FRONT | 0 | 1 | 2 | 3 | 4 |
5 | 6 | 7 | 2 | 0 | 1 | 4 | 5 | 3 | 6 | 7
-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
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 - 64000 | 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 - 64000 | 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 - 239999 | 32.00, 44.10, 48.00 | 44.10
| 5, 5.1 AAC LC + SBR | 240000 - 319999 | 32.00, 44.10, 48.00 |
48.00 | 5, 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. Unlike to HE-AAC configuration the SBR is not covered by ELD audio object
type and needs to be enabled explicitly. Use ::AACENC_SBR_MODE to configure SBR
and its samplingrate ratio with ::AACENC_SBR_RATIO parameter. \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 | 78000 - 160000 | 32.00 - 48.00 | 48.00 | 3
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 104000 - 212000 | 32.00 - 48.00 | 48.00 | 4
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 130000 - 246000 | 32.00 - 48.00 | 48.00 |
5, 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, 5.1 LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00
| 5, 5.1 LD, ELD | 245000 - 339999 | 32.00 - 48.00 |
44.10 | 5, 5.1 LD, ELD | 340000 - 960000 | 44.10 - 48.00 |
48.00 | 5, 5.1
-----------------------------------------------------------------------------------
\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 - 31999 | 16.00 - 24.00 | 24.00 | 1
| 32000 - 47999 | 22.05 - 32.00 | 32.00 | 1
| 48000 - 64000 | 22.05 - 48.00 | 32.00 | 1
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 32000 - 51999 | 16.00 - 24.00 | 24.00 | 2
(downsampled SBR) | 52000 - 59999 | 22.05 - 24.00 | 24.00 | 2
| 60000 - 95999 | 22.05 - 32.00 | 32.00 | 2
| 96000 - 128000 | 22.05 - 48.00 | 32.00 | 2
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 78000 - 99999 | 22.05 - 24.00 | 24.00 | 3
(downsampled SBR) | 100000 - 143999 | 22.05 - 32.00 | 32.00 | 3
| 144000 - 159999 | 22.05 - 48.00 | 32.00 | 3
| 160000 - 192000 | 32.00 - 48.00 | 32.00 | 3
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 104000 - 149999 | 22.05 - 24.00 | 24.00 | 4
(downsampled SBR) | 150000 - 191999 | 22.05 - 32.00 | 32.00 | 4
| 192000 - 211999 | 22.05 - 48.00 | 32.00 | 4
| 212000 - 256000 | 32.00 - 48.00 | 32.00 | 4
-------------------+------------------+-----------------------+------------+-------
ELD + SBR | 130000 - 171999 | 22.05 - 24.00 | 24.00 |
5, 5.1 (downsampled SBR) | 172000 - 239999 | 22.05 - 32.00 | 32.00
| 5, 5.1 | 240000 - 320000 | 32.00 - 48.00 | 32.00 | 5, 5.1
-----------------------------------------------------------------------------------
\endverbatim \n
\subsection reommendedConfigELDv2 AAC-ELD v2, AAC-ELD v2 with SBR.
The ELD v2 212 configuration must be configured explicitly with
::AACENC_CHANNELMODE parameter according MODE_212 value. SBR can be configured
separately through ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO parameter. Following
configurations shall apply to both framelengths 480 and 512. For ELD v2
configuration without SBR and framelength 480 the supported sampling rate is
restricted to the range from 16 kHz up to 24 kHz. \verbatim
-----------------------------------------------------------------------------------
Audio Object Type | Bit Rate Range | Supported | Preferred | No.
of | [bit/s] | Sampling Rates | Sampl. | Chan. |
| [kHz] | Rate | | |
| [kHz] |
-------------------+------------------+-----------------------+------------+-------
ELD-212 | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 2
(without SBR) | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 2
| 40000 - 49999 | 22.05 - 32.00 | 32.00 | 2
| 50000 - 61999 | 24.00 - 44.10 | 32.00 | 2
| 62000 - 84999 | 32.00 - 48.00 | 44.10 | 2
| 85000 - 192000 | 44.10 - 48.00 | 48.00 | 2
-------------------+------------------+-----------------------+------------+-------
ELD-212 + SBR | 18000 - 20999 | 32.00 | 32.00 | 2
(dualrate SBR) | 21000 - 25999 | 32.00 - 44.10 | 32.00 | 2
| 26000 - 31999 | 32.00 - 48.00 | 44.10 | 2
| 32000 - 64000 | 32.00 - 48.00 | 48.00 | 2
-------------------+------------------+-----------------------+------------+-------
ELD-212 + SBR | 18000 - 19999 | 16.00 - 22.05 | 22.05 | 2
(downsampled SBR) | 20000 - 24999 | 16.00 - 24.00 | 22.05 | 2
| 25000 - 31999 | 16.00 - 24.00 | 24.00 | 2
| 32000 - 64000 | 24.00 - 24.00 | 24.00 | 2
-------------------+------------------+-----------------------+------------+-------
\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 numerous listening tests and careful tweaks to ensure the best overall
encoding quality. Also, the maximum bandwidth that can be set manually by the
user is 20kHz or fs/2, whichever value is smaller.
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 double-edged 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. Therefore, for
AAC-ELD, the bitreservoir is limited. It varies between 500 and 4000 bits/frame,
depending on the bitrate/channel.
- For a bitrate of 12kbps/channel and below, the AAC-ELD bitreservoir is 500
bits/frame.
- For a bitrate of 70kbps/channel and above, the AAC-ELD bitreservoir is 4000
bits/frame.
- Between 12kbps/channel and 70kbps/channel, the AAC-ELD bitrervoir is increased
linearly.
- For AAC-LC, the bitrate is only limited by the maximum AAC frame length. It
is, regardless of the available bit reservoir, defined as 6144 bits per channel.
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.
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.
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 AACENC_LIB_H
#define AACENC_LIB_H
#include "machine_type.h"
#include "FDK_audio.h"
/**
* 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_INIT_MPS_ERROR = 0x0045, /*!< MPS 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. */
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 nDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength
and AOT. Does not include framing delay for filling up encoder
PCM input buffer. */
UINT nDelayCore; /*!< Codec delay in PCM samples/channel, w/o delay caused by
the decoder SBR module. This delay is needed to correctly
write edit lists for gapless playback. The decoder may not
know how much delay is introdcued by SBR, since it may not
know if SBR is active at all (implicit signaling),
therefore the deocder must take into account any delay
caused by the SBR module. */
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.
*/
INT bitResState; /*!< State of the bit reservoir in bits. */
} 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_NOT_PRESENT =
256 /*!< Disable writing gain factor (used for comp_profile only). */
} 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 */
UCHAR drcPresentationMode; /*!< Indicatin for DRC Presentation Mode.
- 0: Presentation mode not inticated
- 1: Presentation mode 1
- 2: Presentation mode 2 */
struct {
/* extended ancillary data */
UCHAR extAncDataEnable; /*< Indicates if MPEG4_ext_ancillary_data() exists.
- 0: No MPEG4_ext_ancillary_data().
- 1: Insert MPEG4_ext_ancillary_data(). */
UCHAR
extDownmixLevelEnable; /*< Indicates if ext_downmixing_levels() exists.
- 0: No ext_downmixing_levels().
- 1: Insert ext_downmixing_levels(). */
UCHAR extDownmixLevel_A; /*< Downmix level index A (0...7, according to
table) */
UCHAR extDownmixLevel_B; /*< Downmix level index B (0...7, according to
table) */
UCHAR dmxGainEnable; /*< Indicates if ext_downmixing_global_gains() exists.
- 0: No ext_downmixing_global_gains().
- 1: Insert ext_downmixing_global_gains(). */
INT dmxGain5; /*< Gain factor for downmix to 5 channels.
-15.75dB .. -15.75dB; stepsize: 0.25dB
Scaled with 16 bit. x*2^16.*/
INT dmxGain2; /*< Gain factor for downmix to 2 channels.
-15.75dB .. -15.75dB; stepsize: 0.25dB
Scaled with 16 bit. x*2^16.*/
UCHAR lfeDmxEnable; /*< Indicates if ext_downmixing_lfe_level() exists.
- 0: No ext_downmixing_lfe_level().
- 1: Insert ext_downmixing_lfe_level(). */
UCHAR lfeDmxLevel; /*< Downmix level index for LFE (0..15, according to
table) */
} ExtMetaData;
} 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. The ELD
v2 212 configuration can be configured by ::AACENC_CHANNELMODE
parameter.
- 129: MPEG-2 AAC Low Complexity.
- 132: MPEG-2 AAC Low Complexity with Spectral Band
Replication (HE-AAC).
Please note that the virtual MPEG-2 AOT's basically disables
non-existing Perceptual Noise Substitution tool in AAC encoder
and controls the MPEG_ID flag in adts header. The virtual
MPEG-2 AOT doesn't prohibit specific transport formats. */
AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is
mandatory and interacts with ::AACENC_BITRATEMODE.
- CBR: Bitrate in bits/second.
- VBR: Variable bitrate. Bitrate argument will
be ignored. 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.
- 1: Variable bitrate mode, \ref vbrmode
"very low bitrate".
- 2: Variable bitrate mode, \ref vbrmode
"low bitrate".
- 3: Variable bitrate mode, \ref vbrmode
"medium bitrate".
- 4: Variable bitrate mode, \ref vbrmode
"high bitrate".
- 5: Variable bitrate mode, \ref vbrmode
"very high bitrate". */
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.
- 512: Default length in LD/ELD configuration.
- 480: Length in LD/ELD configuration.
- 256: Length for ELD reduced delay mode (x2).
- 240: Length for ELD reduced delay mode (x2).
- 128: Length for ELD reduced delay mode (x4).
- 120: Length for ELD reduced delay mode (x4). */
AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must
match with number of input channels.
- 1-7, 11,12,14 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 audio bandwidth internally
(default, see chapter \ref BEHAVIOUR_BANDWIDTH).
- 1 to fs/2: Audio bandwidth in Hertz. Limited
to 20kHz max. Not usable if SBR is active. This
setting is for experts only, better do not touch
this value to avoid degraded audio quality. */
AACENC_PEAK_BITRATE =
0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits
per audio frame. Bitrate is in bits/second. The peak bitrate
will internally be limited to the chosen bitrate
::AACENC_BITRATE as lower limit and the
number_of_effective_channels*6144 bit as upper limit.
Setting the peak bitrate equal to ::AACENC_BITRATE does not
necessarily mean that the audio frames will be of constant
size. Since the peak bitate is in bits/second, the frame sizes
can vary by one byte in one or the other direction over various
frames. However, it is not recommended to reduce the peak
pitrate to ::AACENC_BITRATE - it would disable the
bitreservoir, which would affect the audio quality by a large
amount. */
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 or
ADTS (default 1).
- ADTS: Maximum number of sub frames restricted to 4.
- LOAS/LATM: Maximum number of sub frames restricted to 2.*/
AACENC_AUDIOMUXVER =
0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA,
currently not implemented):
- 0: Default, no transmission of tara Buffer fullness, no ASC
length and including actual latm Buffer fullnes.
- 1: Transmission of tara Buffer fullness, ASC length and
actual latm Buffer fullness.
- 2: Transmission of tara Buffer fullness, ASC length and
maximum level of latm Buffer fullness. */
AACENC_PROTECTION = 0x0306, /*!< Configure protection in transport layer:
- 0: No protection. (default)
- 1: CRC active for ADTS transport 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 dynamic_range_info metadata.
- 2: Embed dynamic_range_info and
ancillary_data metadata.
- 3: Embed ancillary_data 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.
* - 0x08: MPS module.
* - 0x10: Metadata module.
* - example: (0x01|0x02|0x04|0x08|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,
* AACENC_INIT_MPS_ERROR, on failure in encoder initialization.
* - AACENC_UNSUPPORTED_PARAMETER, on incorrect input or output buffer
* descriptor 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 /* AACENC_LIB_H */
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