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diff --git a/libSBRenc/src/ps_main.h b/libSBRenc/src/ps_main.h new file mode 100644 index 0000000..6180299 --- /dev/null +++ b/libSBRenc/src/ps_main.h @@ -0,0 +1,271 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 +----------------------------------------------------------------------------------------------------------- */ + +/***************************** MPEG Audio Encoder *************************** + + Initial Authors: Markus Multrus + Contents/Description: PS Wrapper, Downmix header file + +******************************************************************************/ + +#ifndef __INCLUDED_PS_MAIN_H +#define __INCLUDED_PS_MAIN_H + +/* Includes ******************************************************************/ +#include "sbr_def.h" +#include "qmf.h" +#include "ps_encode.h" +#include "FDK_bitstream.h" +#include "FDK_hybrid.h" + + +/* Data Types ****************************************************************/ +typedef enum { + PSENC_STEREO_BANDS_INVALID = 0, + PSENC_STEREO_BANDS_10 = 10, + PSENC_STEREO_BANDS_20 = 20 + +} PSENC_STEREO_BANDS_CONFIG; + +typedef enum { + PSENC_NENV_1 = 1, + PSENC_NENV_2 = 2, + PSENC_NENV_4 = 4, + PSENC_NENV_DEFAULT = PSENC_NENV_2, + PSENC_NENV_MAX = PSENC_NENV_4 + +} PSENC_NENV_CONFIG; + +typedef struct { + UINT bitrateFrom; /* inclusive */ + UINT bitrateTo; /* exclusive */ + PSENC_STEREO_BANDS_CONFIG nStereoBands; + PSENC_NENV_CONFIG nEnvelopes; + LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ + +} psTuningTable_t; + +/* Function / Class Declarations *********************************************/ + +typedef struct T_PARAMETRIC_STEREO { + HANDLE_PS_ENCODE hPsEncode; + PS_OUT psOut[2]; + + FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2][MAX_HYBRID_BANDS]; + FIXP_DBL *pHybridData[HYBRID_READ_OFFSET+HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; + + FIXP_QMF qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>1][QMF_CHANNELS]; + int qmfDelayScale; + + INT psDelay; + UINT maxEnvelopes; + UCHAR dynBandScale[PS_MAX_BANDS]; + FIXP_DBL maxBandValue[PS_MAX_BANDS]; + SCHAR dmxScale; + INT initPS; + INT noQmfSlots; + INT noQmfBands; + + FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2*HYBRID_FILTER_LENGTH*HYBRID_MAX_QMF_BANDS]; + FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2*HYBRID_FILTER_DELAY*(QMF_CHANNELS-HYBRID_MAX_QMF_BANDS)]; + FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; + FDK_SYN_HYB_FILTER fdkHybSynFilter; + +} PARAMETRIC_STEREO; + + +typedef struct T_PSENC_CONFIG { + INT frameSize; + INT qmfFilterMode; + INT sbrPsDelay; + PSENC_STEREO_BANDS_CONFIG nStereoBands; + PSENC_NENV_CONFIG maxEnvelopes; + FIXP_DBL iidQuantErrorThreshold; + +} PSENC_CONFIG, *HANDLE_PSENC_CONFIG; + +typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; + + +/** + * \brief Create a parametric stereo encoder instance. + * + * \param phParametricStereo A pointer to a parametric stereo handle to be allocated. Initialized on return. + * + * \return + * - PSENC_OK, on succes. + * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. + */ +FDK_PSENC_ERROR PSEnc_Create( + HANDLE_PARAMETRIC_STEREO *phParametricStereo + ); + + +/** + * \brief Initialize a parametric stereo encoder instance. + * + * \param hParametricStereo Meta Data handle. + * \param hPsEncConfig Filled parametric stereo configuration structure. + * \param noQmfSlots Number of slots within one audio frame. + * \param noQmfBands Number of QMF bands. + * \param dynamic_RAM Pointer to preallocated workbuffer. + * + * \return + * - PSENC_OK, on succes. + * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. + */ +FDK_PSENC_ERROR PSEnc_Init( + HANDLE_PARAMETRIC_STEREO hParametricStereo, + const HANDLE_PSENC_CONFIG hPsEncConfig, + INT noQmfSlots, + INT noQmfBands + ,UCHAR *dynamic_RAM + ); + + +/** + * \brief Destroy parametric stereo encoder instance. + * + * Deallocate instance and free whole memory. + * + * \param phParametricStereo Pointer to the parametric stereo handle to be deallocated. + * + * \return + * - PSENC_OK, on succes. + * - PSENC_INVALID_HANDLE, on failure. + */ +FDK_PSENC_ERROR PSEnc_Destroy( + HANDLE_PARAMETRIC_STEREO *phParametricStereo + ); + + +/** + * \brief Apply parametric stereo processing. + * + * \param hParametricStereo Meta Data handle. + * \param samples Pointer to 2 channel audio input signal. + * \param timeInStride, Stride factor of input buffer. + * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. + * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. + * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. + * \param downsampledOutSignal Pointer to buffer where to write downmixed timesignal. + * \param sbrSynthQmf Pointer to QMF synthesis filterbank. + * \param qmfScale Return scaling factor of the qmf data. + * \param sendHeader Signal whether to write header data. + * + * \return + * - PSENC_OK, on succes. + * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. + */ +FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( + HANDLE_PARAMETRIC_STEREO hParametricStereo, + INT_PCM *samples[2], + UINT timeInStride, + QMF_FILTER_BANK **hQmfAnalysis, + FIXP_QMF **RESTRICT downmixedRealQmfData, + FIXP_QMF **RESTRICT downmixedImagQmfData, + INT_PCM *downsampledOutSignal, + HANDLE_QMF_FILTER_BANK sbrSynthQmf, + SCHAR *qmfScale, + const int sendHeader + ); + + +/** + * \brief Write parametric stereo bitstream. + * + * Write ps_data() element to bitstream and return number of written bits. + * Returns number of written bits only, if hBitstream == NULL. + * + * \param hParametricStereo Meta Data handle. + * \param hBitstream Bitstream buffer handle. + * + * \return + * - number of written bits. + */ +INT FDKsbrEnc_PSEnc_WritePSData( + HANDLE_PARAMETRIC_STEREO hParametricStereo, + HANDLE_FDK_BITSTREAM hBitstream + ); + +#endif /* __INCLUDED_PS_MAIN_H */ |