ps_main.h - Android社区 - https://www.androidos.net.cn/

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

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 */

C++程序 | 272行 | 11.01 KB

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