mirror of https://github.com/mstorsjo/fdk-aac.git
757 lines
46 KiB
C
757 lines
46 KiB
C
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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/***************************** MPEG-4 AAC Decoder **************************
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Author(s): Manuel Jander
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******************************************************************************/
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/**
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* \file aacdecoder_lib.h
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* \brief FDK AAC decoder library interface header file.
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*
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\page INTRO Introduction
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\section SCOPE Scope
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This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Decoder
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library developed by the Fraunhofer Institute for Integrated Circuits (IIS).
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Depending on the library configuration, it implements decoding of AAC-LC (Low-Complexity),
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HE-AAC (High-Efficiency AAC, v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay).
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All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD
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versions of the library. All references to PS (Parametric Stereo) are only applicable to
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HE-AAC v2 versions of the library.
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\section DecoderBasics Decoder Basics
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This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio
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coding standard. To understand all the terms in this document, you are encouraged to read
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the following documents.
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- ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams.
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- ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams.
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- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004
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MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal
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is partitioned into overlapping portions and transformed into frequency domain. The spectral
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components are then quantized and coded.\n
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An MPEG2 or MPEG4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3),
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the length of individual frames is not restricted to a fixed number of bytes, but can take on
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any length between 1 and 768 bytes.
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\page LIBUSE Library Usage
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\section InterfaceDescritpion API Description
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All API header files are located in the folder /include of the release package. They are described in
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detail in this document. All header files are provided for usage in C/C++ programs. The AAC decoder library
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API functions are located at aacdecoder_lib.h.
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In binary releases the decoder core resides in statically linkable libraries called for example libAACdec.a,
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(Linux) or FDK_aacDec_lib (Microsoft Visual C++).
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\section Calling_Sequence Calling Sequence
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For decoding of ISO/MPEG-2/4 AAC or HE-AAC v2 bitstreams the following sequence is mandatory. Input read
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and output write functions as well as the corresponding open and close functions are left out, since they
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may be implemented differently according to the user's specific requirements. The example implementation in
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main.cpp uses file-based input/output, and in such case call mpegFileRead_Open() to open an input file and
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to allocate memory for the required structures, and the corresponding mpegFileRead_Close() to close opened
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files and to de-allocate associated structures. mpegFileRead_Open() tries to detect the bitstream format and
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in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio
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Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to
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provide this information manually. For any other bitstream formats that are usually applicable in streaming
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applications, the decoder itself will try to synchronize and parse the given bitstream fragment using the
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FDK transport library. Hence, for streaming applications (without file access) this step is not necessary.
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-# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance.
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\dontinclude main.cpp
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\skipline aacDecoder_Open
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-# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config (SMC)) is available, call
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aacDecoder_ConfigRaw() to pass it to the decoder and before the decoding process starts. If this data is
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not available in advance, the decoder will get it from the bitstream and configure itself while decoding
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with aacDecoder_DecodeFrame().
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-# Begin decoding loop.
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\skipline do {
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-# Read data from bitstream file or stream into a client-supplied input buffer ("inBuffer" in main.cpp).
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If it is very small like just 4, aacDecoder_DecodeFrame() will
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repeatedly return ::AAC_DEC_NOT_ENOUGH_BITS until enough bits were fed by aacDecoder_Fill(). Only read data
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when this buffer has completely been processed and is then empty. For file-based input execute
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mpegFileRead_Read() or any other implementation with similar functionality.
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-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied
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external bitstream input buffer.
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\skipline aacDecoder_Fill
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-# Call aacDecoder_DecodeFrame() which writes decoded PCM audio data to a client-supplied buffer. It is the
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client's responsibility to allocate a buffer which is large enough to hold this output data.
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\skipline aacDecoder_DecodeFrame
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If the bitstream's configuration (number of channels, sample rate, frame size) is not known in advance, you may
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call aacDecoder_GetStreamInfo() to retrieve a structure containing this information and then initialize an audio
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output device. In the example main.cpp, if the number of channels or the sample rate has changed since program
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start or since the previously decoded frame, the audio output device will be re-initialized. If WAVE file output
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is chosen, a new WAVE file for each new configuration will be created.
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\skipline aacDecoder_GetStreamInfo
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-# Repeat steps 5 to 7 until no data to decode is available anymore, or if an error occured.
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\skipline } while
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-# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer structures.
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\skipline aacDecoder_Close
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\section BufferSystem Buffer System
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There are three main buffers in an AAC decoder application. One external input buffer to hold bitstream
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data from file I/O or elsewhere, one decoder-internal input buffer, and one to hold the decoded output
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PCM sample data, whereas this output buffer may overlap with the external input buffer.
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The external input buffer is set in the example framework main.cpp and its size is defined by ::IN_BUF_SIZE.
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You may freely choose different sizes here. To feed the data to the decoder-internal input buffer, use the
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function aacDecoder_Fill(). This function returns important information about how many bytes in the
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external input buffer have not yet been copied into the internal input buffer (variable bytesValid).
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Once the external buffer has been fully copied, it can be re-filled again.
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In case you want to re-fill it when there are still unprocessed bytes (bytesValid is unequal 0), you
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would have to additionally perform a memcpy(), so that just means unnecessary computational overhead
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and therefore we recommend to re-fill the buffer only when bytesValid is 0.
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\image latex dec_buffer.png "Lifecycle of the external input buffer" width=9cm
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The size of the decoder-internal input buffer is set in tpdec_lib.h (see define ::TRANSPORTDEC_INBUF_SIZE).
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You may choose a smaller size under the following considerations:
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- each input channel requires 768 bytes
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- the whole buffer must be of size 2^n
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So for example a stereo decoder:
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\f[
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TRANSPORTDEC\_INBUF\_SIZE = 2 * 768 = 1536 => 2048
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\f]
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tpdec_lib.h and TRANSPORTDEC_INBUF_SIZE are not part of the decoder's library interface. Therefore
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only source-code clients may change this setting. If you received a library release, please ask us and
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we can change this in order to meet your memory requirements.
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\page OutputFormat Decoder audio output
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\section OutputFormatObtaining Obtaining channel mapping information
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The decoded audio output format is indicated by a set of variables of the CStreamInfo structure.
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While the members sampleRate, frameSize and numChannels might be quite self explaining,
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pChannelType and pChannelIndices might require some more detailed explanation.
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These two arrays indicate what is each output channel supposed to be. Both array have
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CStreamInfo::numChannels cells. Each cell of pChannelType indicates the channel type, described in
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the enum ::AUDIO_CHANNEL_TYPE defined in FDK_audio.h. The cells of pChannelIndices indicate the sub index
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among the channels starting with 0 among all channels of the same audio channel type.
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The indexing scheme is the same as for MPEG-2/4. Thus indices are counted upwards starting from the front
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direction (thus a center channel if any, will always be index 0). Then the indices count up, starting always
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with the left side, pairwise from front toward back. For detailed explanation, please refer to
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ISO/IEC 13818-7:2005(E), chapter 8.5.3.2.
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In case a Program Config is included in the audio configuration, the channel mapping described within
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it will be adopted.
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In case of MPEG-D Surround the channel mapping will follow the same criteria described in ISO/IEC 13818-7:2005(E),
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but adding corresponding top channels to the channel types front, side and back, in order to avoid any
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loss of information.
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\section OutputFormatChange Changing the audio output format
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The channel interleaving scheme and the actual channel order can be changed at runtime through the
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parameters ::AAC_PCM_OUTPUT_INTERLEAVED and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING. See the description of those
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parameters and the decoder library function aacDecoder_SetParam() for more detail.
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\section OutputFormatExample Channel mapping examples
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The following examples illustrate the location of individual audio samples in the audio buffer that
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is passed to aacDecoder_DecodeFrame() and the expected data in the CStreamInfo structure which can be obtained
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by calling aacDecoder_GetStreamInfo().
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\subsection ExamplesStereo Stereo
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In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 0 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
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a AAC-LC bit stream which has channelConfiguration = 2 in its audio specific config would lead
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to the following values in CStreamInfo:
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CStreamInfo::numChannels = 2
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CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT }
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CStreamInfo::pChannelIndices = { 0, 1 }
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Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 0, the audio channels will be located as contiguous blocks
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in the output buffer as follows:
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\verbatim
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<left sample 0> <left sample 1> <left sample 2> ... <left sample N>
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<right sample 0> <right sample 1> <right sample 2> ... <right sample N>
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\endverbatim
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Where N equals to CStreamInfo::frameSize .
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\subsection ExamplesSurround Surround 5.1
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In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
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a AAC-LC bit stream which has channelConfiguration = 6 in its audio specific config, would lead
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to the following values in CStreamInfo:
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CStreamInfo::numChannels = 6
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CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_FRONT, ::ACT_LFE, ::ACT_BACK, ::ACT_BACK }
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CStreamInfo::pChannelIndices = { 1, 2, 0, 0, 0, 1 }
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Since ::AAC_PCM_OUTPUT_CHANNEL_MAPPING is 1, WAV file channel ordering will be used. For a 5.1 channel
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scheme, thus the channels would be: front left, front right, center, LFE, surround left, surround right.
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Thus the third channel is the center channel, receiving the index 0. The other front channels are
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front left, front right being placed as first and second channels with indices 1 and 2 correspondingly.
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There is only one LFE, placed as the fourth channel and index 0. Finally both surround
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channels get the type definition ACT_BACK, and the indices 0 and 1.
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Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 1, the audio channels will be placed in the output buffer
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as follows:
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\verbatim
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<front left sample 0> <front right sample 0>
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<center sample 0> <LFE sample 0>
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<surround left sample 0> <surround right sample 0>
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<front left sample 1> <front right sample 1>
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<center sample 1> <LFE sample 1>
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<surround left sample 1> <surround right sample 1>
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...
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<front left sample N> <front right sample N>
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<center sample N> <LFE sample N>
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<surround left sample N> <surround right sample N>
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\endverbatim
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Where N equals to CStreamInfo::frameSize .
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\subsection ExamplesArib ARIB coding mode 2/1
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In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
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in case of a ARIB bit stream using coding mode 2/1 as described in ARIB STD-B32 Part 2 Version 2.1-E1, page 61,
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would lead to the following values in CStreamInfo:
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CStreamInfo::numChannels = 3
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CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT,:: ACT_BACK }
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CStreamInfo::pChannelIndices = { 0, 1, 0 }
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The audio channels will be placed as follows in the audio output buffer:
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\verbatim
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<front left sample 0> <front right sample 0> <mid surround sample 0>
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<front left sample 1> <front right sample 1> <mid surround sample 1>
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...
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<front left sample N> <front right sample N> <mid surround sample N>
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Where N equals to CStreamInfo::frameSize .
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\endverbatim
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*/
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#ifndef AACDECODER_LIB_H
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#define AACDECODER_LIB_H
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#include "machine_type.h"
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#include "FDK_audio.h"
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#include "genericStds.h"
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/**
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* \brief AAC decoder error codes.
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*/
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typedef enum {
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AAC_DEC_OK = 0x0000, /*!< No error occured. Output buffer is valid and error free. */
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AAC_DEC_OUT_OF_MEMORY = 0x0002, /*!< Heap returned NULL pointer. Output buffer is invalid. */
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AAC_DEC_UNKNOWN = 0x0005, /*!< Error condition is of unknown reason, or from a another module. Output buffer is invalid. */
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/* Synchronization errors. Output buffer is invalid. */
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aac_dec_sync_error_start = 0x1000,
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AAC_DEC_TRANSPORT_SYNC_ERROR = 0x1001, /*!< The transport decoder had syncronisation problems. Do not exit decoding. Just feed new
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bitstream data. */
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AAC_DEC_NOT_ENOUGH_BITS = 0x1002, /*!< The input buffer ran out of bits. */
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aac_dec_sync_error_end = 0x1FFF,
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/* Initialization errors. Output buffer is invalid. */
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aac_dec_init_error_start = 0x2000,
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AAC_DEC_INVALID_HANDLE = 0x2001, /*!< The handle passed to the function call was invalid (NULL). */
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AAC_DEC_UNSUPPORTED_AOT = 0x2002, /*!< The AOT found in the configuration is not supported. */
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AAC_DEC_UNSUPPORTED_FORMAT = 0x2003, /*!< The bitstream format is not supported. */
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AAC_DEC_UNSUPPORTED_ER_FORMAT = 0x2004, /*!< The error resilience tool format is not supported. */
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AAC_DEC_UNSUPPORTED_EPCONFIG = 0x2005, /*!< The error protection format is not supported. */
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AAC_DEC_UNSUPPORTED_MULTILAYER = 0x2006, /*!< More than one layer for AAC scalable is not supported. */
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AAC_DEC_UNSUPPORTED_CHANNELCONFIG = 0x2007, /*!< The channel configuration (either number or arrangement) is not supported. */
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AAC_DEC_UNSUPPORTED_SAMPLINGRATE = 0x2008, /*!< The sample rate specified in the configuration is not supported. */
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AAC_DEC_INVALID_SBR_CONFIG = 0x2009, /*!< The SBR configuration is not supported. */
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AAC_DEC_SET_PARAM_FAIL = 0x200A, /*!< The parameter could not be set. Either the value was out of range or the parameter does
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not exist. */
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AAC_DEC_NEED_TO_RESTART = 0x200B, /*!< The decoder needs to be restarted, since the requiered configuration change cannot be
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performed. */
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aac_dec_init_error_end = 0x2FFF,
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/* Decode errors. Output buffer is valid but concealed. */
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aac_dec_decode_error_start = 0x4000,
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AAC_DEC_TRANSPORT_ERROR = 0x4001, /*!< The transport decoder encountered an unexpected error. */
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AAC_DEC_PARSE_ERROR = 0x4002, /*!< Error while parsing the bitstream. Most probably it is corrupted, or the system crashed. */
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AAC_DEC_UNSUPPORTED_EXTENSION_PAYLOAD = 0x4003, /*!< Error while parsing the extension payload of the bitstream. The extension payload type
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found is not supported. */
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AAC_DEC_DECODE_FRAME_ERROR = 0x4004, /*!< The parsed bitstream value is out of range. Most probably the bitstream is corrupt, or
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the system crashed. */
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AAC_DEC_CRC_ERROR = 0x4005, /*!< The embedded CRC did not match. */
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AAC_DEC_INVALID_CODE_BOOK = 0x4006, /*!< An invalid codebook was signalled. Most probably the bitstream is corrupt, or the system
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crashed. */
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AAC_DEC_UNSUPPORTED_PREDICTION = 0x4007, /*!< Predictor found, but not supported in the AAC Low Complexity profile. Most probably the
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bitstream is corrupt, or has a wrong format. */
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AAC_DEC_UNSUPPORTED_CCE = 0x4008, /*!< A CCE element was found which is not supported. Most probably the bitstream is corrupt, or
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has a wrong format. */
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AAC_DEC_UNSUPPORTED_LFE = 0x4009, /*!< A LFE element was found which is not supported. Most probably the bitstream is corrupt, or
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has a wrong format. */
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AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA = 0x400A, /*!< Gain control data found but not supported. Most probably the bitstream is corrupt, or has
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a wrong format. */
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AAC_DEC_UNSUPPORTED_SBA = 0x400B, /*!< SBA found, but currently not supported in the BSAC profile. */
|
|
AAC_DEC_TNS_READ_ERROR = 0x400C, /*!< Error while reading TNS data. Most probably the bitstream is corrupt or the system
|
|
crashed. */
|
|
AAC_DEC_RVLC_ERROR = 0x400D, /*!< Error while decoding error resillient data. */
|
|
aac_dec_decode_error_end = 0x4FFF,
|
|
|
|
/* Ancillary data errors. Output buffer is valid. */
|
|
aac_dec_anc_data_error_start = 0x8000,
|
|
AAC_DEC_ANC_DATA_ERROR = 0x8001, /*!< Non severe error concerning the ancillary data handling. */
|
|
AAC_DEC_TOO_SMALL_ANC_BUFFER = 0x8002, /*!< The registered ancillary data buffer is too small to receive the parsed data. */
|
|
AAC_DEC_TOO_MANY_ANC_ELEMENTS = 0x8003, /*!< More than the allowed number of ancillary data elements should be written to buffer. */
|
|
aac_dec_anc_data_error_end = 0x8FFF
|
|
|
|
|
|
} AAC_DECODER_ERROR;
|
|
|
|
|
|
/** Macro to identify initialization errors. */
|
|
#define IS_INIT_ERROR(err) ( (((err)>=aac_dec_init_error_start) && ((err)<=aac_dec_init_error_end)) ? 1 : 0)
|
|
/** Macro to identify decode errors. */
|
|
#define IS_DECODE_ERROR(err) ( (((err)>=aac_dec_decode_error_start) && ((err)<=aac_dec_decode_error_end)) ? 1 : 0)
|
|
/** Macro to identify if the audio output buffer contains valid samples after calling aacDecoder_DecodeFrame(). */
|
|
#define IS_OUTPUT_VALID(err) ( ((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err) )
|
|
|
|
/**
|
|
* \brief AAC decoder setting parameters
|
|
*/
|
|
typedef enum
|
|
{
|
|
AAC_PCM_OUTPUT_INTERLEAVED = 0x0000, /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */
|
|
AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE = 0x0002, /*!< Defines how the decoder processes two channel signals: \n
|
|
0: Leave both signals as they are (default). \n
|
|
1: Create a dual mono output signal from channel 1. \n
|
|
2: Create a dual mono output signal from channel 2. \n
|
|
3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
|
|
AAC_PCM_OUTPUT_CHANNEL_MAPPING = 0x0003, /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */
|
|
AAC_PCM_LIMITER_ENABLE = 0x0004, /*!< Enable signal level limiting. \n
|
|
-1: Auto-config. Enable limiter for all non-lowdelay configurations by default. \n
|
|
0: Disable limiter in general. \n
|
|
1: Enable limiter always.
|
|
It is recommended to call the decoder with a AACDEC_CLRHIST flag to reset all states when
|
|
the limiter switch is changed explicitly. */
|
|
AAC_PCM_LIMITER_ATTACK_TIME = 0x0005, /*!< Signal level limiting attack time in ms.
|
|
Default confguration is 15 ms. Adjustable range from 1 ms to 15 ms. */
|
|
AAC_PCM_LIMITER_RELEAS_TIME = 0x0006, /*!< Signal level limiting release time in ms.
|
|
Default configuration is 50 ms. Adjustable time must be larger than 0 ms. */
|
|
AAC_PCM_MIN_OUTPUT_CHANNELS = 0x0011, /*!< Minimum number of PCM output channels. If higher than the number of encoded audio channels,
|
|
a simple channel extension is applied. \n
|
|
-1, 0: Disable channel extenstion feature. The decoder output contains the same number of
|
|
channels as the encoded bitstream. \n
|
|
1: This value is currently needed only together with the mix-down feature. See
|
|
::AAC_PCM_MAX_OUTPUT_CHANNELS and note 2 below. \n
|
|
2: Encoded mono signals will be duplicated to achieve a 2/0/0.0 channel output
|
|
configuration. \n
|
|
6: The decoder trys to reorder encoded signals with less than six channels to achieve
|
|
a 3/0/2.1 channel output signal. Missing channels will be filled with a zero signal.
|
|
If reordering is not possible the empty channels will simply be appended. Only
|
|
available if instance is configured to support multichannel output. \n
|
|
8: The decoder trys to reorder encoded signals with less than eight channels to
|
|
achieve a 3/0/4.1 channel output signal. Missing channels will be filled with a
|
|
zero signal. If reordering is not possible the empty channels will simply be
|
|
appended. Only available if instance is configured to support multichannel output.\n
|
|
NOTE: \n
|
|
1. The channel signalling (CStreamInfo::pChannelType and CStreamInfo::pChannelIndices)
|
|
will not be modified. Added empty channels will be signalled with channel type
|
|
AUDIO_CHANNEL_TYPE::ACT_NONE. \n
|
|
2. If the parameter value is greater than that of ::AAC_PCM_MAX_OUTPUT_CHANNELS both will
|
|
be set to the same value. \n
|
|
3. This parameter does not affect MPEG Surround processing. */
|
|
AAC_PCM_MAX_OUTPUT_CHANNELS = 0x0012, /*!< Maximum number of PCM output channels. If lower than the number of encoded audio channels,
|
|
downmixing is applied accordingly. If dedicated metadata is available in the stream it
|
|
will be used to achieve better mixing results. \n
|
|
-1, 0: Disable downmixing feature. The decoder output contains the same number of channels
|
|
as the encoded bitstream. \n
|
|
1: All encoded audio configurations with more than one channel will be mixed down to
|
|
one mono output signal. \n
|
|
2: The decoder performs a stereo mix-down if the number encoded audio channels is
|
|
greater than two. \n
|
|
6: If the number of encoded audio channels is greater than six the decoder performs a
|
|
mix-down to meet the target output configuration of 3/0/2.1 channels. Only
|
|
available if instance is configured to support multichannel output. \n
|
|
8: This value is currently needed only together with the channel extension feature.
|
|
See ::AAC_PCM_MIN_OUTPUT_CHANNELS and note 2 below. Only available if instance is
|
|
configured to support multichannel output. \n
|
|
NOTE: \n
|
|
1. Down-mixing of any seven or eight channel configuration not defined in ISO/IEC 14496-3
|
|
PDAM 4 is not supported by this software version. \n
|
|
2. If the parameter value is greater than zero but smaller than ::AAC_PCM_MIN_OUTPUT_CHANNELS
|
|
both will be set to same value. \n
|
|
3. The operating mode of the MPEG Surround module will be set accordingly. \n
|
|
4. Setting this param with any value will disable the binaural processing of the MPEG
|
|
Surround module (::AAC_MPEGS_BINAURAL_ENABLE=0). */
|
|
|
|
AAC_CONCEAL_METHOD = 0x0100, /*!< Error concealment: Processing method. \n
|
|
0: Spectral muting. \n
|
|
1: Noise substitution (see ::CONCEAL_NOISE). \n
|
|
2: Energy interpolation (adds additional signal delay of one frame, see ::CONCEAL_INTER). \n */
|
|
|
|
AAC_DRC_BOOST_FACTOR = 0x0200, /*!< Dynamic Range Control: Scaling factor for boosting gain values.
|
|
Defines how the boosting DRC factors (conveyed in the bitstream) will be applied to the
|
|
decoded signal. The valid values range from 0 (don't apply boost factors) to 127 (fully
|
|
apply all boosting factors). */
|
|
AAC_DRC_ATTENUATION_FACTOR = 0x0201, /*!< Dynamic Range Control: Scaling factor for attenuating gain values. Same as
|
|
AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */
|
|
AAC_DRC_REFERENCE_LEVEL = 0x0202, /*!< Dynamic Range Control: Target reference level. Defines the level below full-scale
|
|
(quantized in steps of 0.25dB) to which the output audio signal will be normalized to by
|
|
the DRC module. The valid values range from 0 (full-scale) to 127 (31.75 dB below
|
|
full-scale). The value smaller than 0 switches off normalization. */
|
|
AAC_DRC_HEAVY_COMPRESSION = 0x0203, /*!< Dynamic Range Control: En-/Disable DVB specific heavy compression (aka RF mode).
|
|
If set to 1, the decoder will apply the compression values from the DVB specific ancillary
|
|
data field. At the same time the MPEG-4 Dynamic Range Control tool will be disabled. By
|
|
default heavy compression is disabled. */
|
|
|
|
AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n
|
|
-1: Use internal default. Implies MPEG Surround partially complex accordingly. \n
|
|
0: Use complex QMF data mode. \n
|
|
1: Use real (low power) QMF data mode. \n */
|
|
|
|
AAC_MPEGS_ENABLE = 0x0500, /*!< MPEG Surround: Allow/Disable decoding of MPS content. Available only for decoders with MPEG
|
|
Surround support. */
|
|
|
|
AAC_TPDEC_CLEAR_BUFFER = 0x0603 /*!< Clear internal bit stream buffer of transport layers. The decoder will start decoding
|
|
at new data passed after this event and any previous data is discarded. */
|
|
|
|
} AACDEC_PARAM;
|
|
|
|
/**
|
|
* \brief This structure gives information about the currently decoded audio data.
|
|
* All fields are read-only.
|
|
*/
|
|
typedef struct
|
|
{
|
|
/* These five members are the only really relevant ones for the user. */
|
|
INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */
|
|
INT frameSize; /*!< The frame size of the decoded PCM audio signal. \n
|
|
1024 or 960 for AAC-LC \n
|
|
2048 or 1920 for HE-AAC (v2) \n
|
|
512 or 480 for AAC-LD and AAC-ELD */
|
|
INT numChannels; /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */
|
|
AUDIO_CHANNEL_TYPE *pChannelType; /*!< Audio channel type of each output audio channel. */
|
|
UCHAR *pChannelIndices; /*!< Audio channel index for each output audio channel.
|
|
See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
|
|
/* Decoder internal members. */
|
|
INT aacSampleRate; /*!< Sampling rate in Hz without SBR (from configuration info). */
|
|
INT profile; /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)). */
|
|
AUDIO_OBJECT_TYPE aot; /*!< Audio Object Type (from ASC): is set to the appropriate value for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */
|
|
INT channelConfig; /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ... */
|
|
INT bitRate; /*!< Instantaneous bit rate. */
|
|
INT aacSamplesPerFrame; /*!< Samples per frame for the AAC core (from ASC). \n
|
|
1024 or 960 for AAC-LC \n
|
|
512 or 480 for AAC-LD and AAC-ELD */
|
|
INT aacNumChannels; /*!< The number of audio channels after AAC core processing (before PS or MPS processing).
|
|
CAUTION: This are not the final number of output channels! */
|
|
AUDIO_OBJECT_TYPE extAot; /*!< Extension Audio Object Type (from ASC) */
|
|
INT extSamplingRate; /*!< Extension sampling rate in Hz (from ASC) */
|
|
|
|
UINT outputDelay; /*!< The number of samples the output is additionally delayed by the decoder. */
|
|
|
|
UINT flags; /*!< Copy of internal flags. Only to be written by the decoder, and only to be read externally. */
|
|
|
|
SCHAR epConfig; /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.) */
|
|
|
|
/* Statistics */
|
|
INT numLostAccessUnits; /*!< This integer will reflect the estimated amount of lost access units in case aacDecoder_DecodeFrame()
|
|
returns AAC_DEC_TRANSPORT_SYNC_ERROR. It will be < 0 if the estimation failed. */
|
|
|
|
UINT numTotalBytes; /*!< This is the number of total bytes that have passed through the decoder. */
|
|
UINT numBadBytes; /*!< This is the number of total bytes that were considered with errors from numTotalBytes. */
|
|
UINT numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */
|
|
UINT numBadAccessUnits; /*!< This is the number of total access units that were considered with errors from numTotalBytes. */
|
|
|
|
/* Metadata */
|
|
SCHAR drcProgRefLev; /*!< DRC program reference level. Defines the reference level below full-scale.
|
|
It is quantized in steps of 0.25dB. The valid values range from 0 (0 dBFS) to 127 (-31.75 dBFS).
|
|
It is used to reflect the average loudness of the audio in LKFS accoring to ITU-R BS 1770.
|
|
If no level has been found in the bitstream the value is -1. */
|
|
SCHAR drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154, this field indicates whether
|
|
light (MPEG-4 Dynamic Range Control tool) or heavy compression (DVB heavy compression)
|
|
dynamic range control shall take priority on the outputs.
|
|
For details, see ETSI TS 101 154, table C.33. Possible values are: \n
|
|
-1: No corresponding metadata found in the bitstream \n
|
|
0: DRC presentation mode not indicated \n
|
|
1: DRC presentation mode 1 \n
|
|
2: DRC presentation mode 2 \n
|
|
3: Reserved */
|
|
|
|
} CStreamInfo;
|
|
|
|
|
|
typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER; /*!< Pointer to a AAC decoder instance. */
|
|
|
|
#ifdef __cplusplus
|
|
extern "C"
|
|
{
|
|
#endif
|
|
|
|
/**
|
|
* \brief Initialize ancillary data buffer.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param buffer Pointer to (external) ancillary data buffer.
|
|
* \param size Size of the buffer pointed to by buffer.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_AncDataInit ( HANDLE_AACDECODER self,
|
|
UCHAR *buffer,
|
|
int size );
|
|
|
|
/**
|
|
* \brief Get one ancillary data element.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param index Index of the ancillary data element to get.
|
|
* \param ptr Pointer to a buffer receiving a pointer to the requested ancillary data element.
|
|
* \param size Pointer to a buffer receiving the length of the requested ancillary data element.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_AncDataGet ( HANDLE_AACDECODER self,
|
|
int index,
|
|
UCHAR **ptr,
|
|
int *size );
|
|
|
|
/**
|
|
* \brief Set one single decoder parameter.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param param Parameter to be set.
|
|
* \param value Parameter value.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_SetParam ( const HANDLE_AACDECODER self,
|
|
const AACDEC_PARAM param,
|
|
const INT value );
|
|
|
|
|
|
/**
|
|
* \brief Get free bytes inside decoder internal buffer
|
|
* \param self Handle of AAC decoder instance
|
|
* \param pFreeBytes Pointer to variable receving amount of free bytes inside decoder internal buffer
|
|
* \return Error code
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_GetFreeBytes ( const HANDLE_AACDECODER self,
|
|
UINT *pFreeBytes);
|
|
|
|
/**
|
|
* \brief Open an AAC decoder instance
|
|
* \param transportFmt The transport type to be used
|
|
* \return AAC decoder handle
|
|
*/
|
|
LINKSPEC_H HANDLE_AACDECODER
|
|
aacDecoder_Open ( TRANSPORT_TYPE transportFmt, UINT nrOfLayers );
|
|
|
|
/**
|
|
* \brief Explicitly configure the decoder by passing a raw AudioSpecificConfig (ASC) or a StreamMuxConfig (SMC),
|
|
* contained in a binary buffer. This is required for MPEG-4 and Raw Packets file format bitstreams
|
|
* as well as for LATM bitstreams with no in-band SMC. If the transport format is LATM with or without
|
|
* LOAS, configuration is assumed to be an SMC, for all other file formats an ASC.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param conf Pointer to an unsigned char buffer containing the binary configuration buffer (either ASC or SMC).
|
|
* \param length Length of the configuration buffer in bytes.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_ConfigRaw ( HANDLE_AACDECODER self,
|
|
UCHAR *conf[],
|
|
const UINT length[] );
|
|
|
|
|
|
/**
|
|
* \brief Fill AAC decoder's internal input buffer with bitstream data from the external input buffer.
|
|
* The function only copies such data as long as the decoder-internal input buffer is not full.
|
|
* So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a
|
|
* subsequent call of %aacDecoder_Fill(), the right position in pBuffer can be determined to
|
|
* grab the next data.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param pBuffer Pointer to external input buffer.
|
|
* \param bufferSize Size of external input buffer. This argument is required because decoder-internally
|
|
* we need the information to calculate the offset to pBuffer, where the next
|
|
* available data is, which is then fed into the decoder-internal buffer (as much
|
|
* as possible). Our example framework implementation fills the buffer at pBuffer
|
|
* again, once it contains no available valid bytes anymore (meaning bytesValid equal 0).
|
|
* \param bytesValid Number of bitstream bytes in the external bitstream buffer that have not yet been
|
|
* copied into the decoder's internal bitstream buffer by calling this function.
|
|
* The value is updated according to the amount of newly copied bytes.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_Fill ( HANDLE_AACDECODER self,
|
|
UCHAR *pBuffer[],
|
|
const UINT bufferSize[],
|
|
UINT *bytesValid );
|
|
|
|
#define AACDEC_CONCEAL 1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment module \
|
|
to generate a substitute signal for one lost frame. New input data will not be
|
|
considered. */
|
|
#define AACDEC_FLUSH 2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed audio \
|
|
without having new input data. Thus new input data will not be considered.*/
|
|
#define AACDEC_INTR 4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data discontinuity. \
|
|
Resync any internals as necessary. */
|
|
#define AACDEC_CLRHIST 8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history buffers.\
|
|
CAUTION: This can cause discontinuities in the output signal. */
|
|
|
|
/**
|
|
* \brief Decode one audio frame
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \param pTimeData Pointer to external output buffer where the decoded PCM samples will be stored into.
|
|
* \param flags Bit field with flags for the decoder: \n
|
|
* (flags & AACDEC_CONCEAL) == 1: Do concealment. \n
|
|
* (flags & AACDEC_FLUSH) == 2: Discard input data. Flush filter banks (output delayed audio). \n
|
|
* (flags & AACDEC_INTR) == 4: Input data is discontinuous. Resynchronize any internals as necessary.
|
|
* \return Error code.
|
|
*/
|
|
LINKSPEC_H AAC_DECODER_ERROR
|
|
aacDecoder_DecodeFrame ( HANDLE_AACDECODER self,
|
|
INT_PCM *pTimeData,
|
|
const INT timeDataSize,
|
|
const UINT flags );
|
|
|
|
/**
|
|
* \brief De-allocate all resources of an AAC decoder instance.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \return void
|
|
*/
|
|
LINKSPEC_H void aacDecoder_Close ( HANDLE_AACDECODER self );
|
|
|
|
/**
|
|
* \brief Get CStreamInfo handle from decoder.
|
|
*
|
|
* \param self AAC decoder handle.
|
|
* \return Reference to requested CStreamInfo.
|
|
*/
|
|
LINKSPEC_H CStreamInfo* aacDecoder_GetStreamInfo( HANDLE_AACDECODER self );
|
|
|
|
/**
|
|
* \brief Get decoder library info.
|
|
*
|
|
* \param info Pointer to an allocated LIB_INFO structure.
|
|
* \return 0 on success
|
|
*/
|
|
LINKSPEC_H INT aacDecoder_GetLibInfo( LIB_INFO *info );
|
|
|
|
|
|
#ifdef __cplusplus
|
|
}
|
|
#endif
|
|
|
|
#endif /* AACDECODER_LIB_H */
|