libavcodec/aac.h - vendor/opensource/ffmpeg - Git at Google

 /*
  * AAC definitions and structures
  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 /**
  * @file libavcodec/aac.h
  * AAC definitions and structures
  * @author Oded Shimon  ( ods15 ods15 dyndns org )
  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
  */

 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H

 #include "libavutil/internal.h"
 #include "avcodec.h"
 #include "dsputil.h"
 #include "mpeg4audio.h"

 #include <stdint.h>

 #define AAC_INIT_VLC_STATIC(num, size) \
     INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
          ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
         ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
         size);

 #define MAX_CHANNELS 64
 #define MAX_ELEM_ID 16

 #define TNS_MAX_ORDER 20

 enum AudioObjectType {
     AOT_NULL,
                                // Support?                Name
     AOT_AAC_MAIN,              ///< Y                       Main
     AOT_AAC_LC,                ///< Y                       Low Complexity
     AOT_AAC_SSR,               ///< N (code in SoC repo)    Scalable Sample Rate
     AOT_AAC_LTP,               ///< N (code in SoC repo)    Long Term Prediction
     AOT_SBR,                   ///< N (in progress)         Spectral Band Replication
     AOT_AAC_SCALABLE,          ///< N                       Scalable
     AOT_TWINVQ,                ///< N                       Twin Vector Quantizer
     AOT_CELP,                  ///< N                       Code Excited Linear Prediction
     AOT_HVXC,                  ///< N                       Harmonic Vector eXcitation Coding
     AOT_TTSI             = 12, ///< N                       Text-To-Speech Interface
     AOT_MAINSYNTH,             ///< N                       Main Synthesis
     AOT_WAVESYNTH,             ///< N                       Wavetable Synthesis
     AOT_MIDI,                  ///< N                       General MIDI
     AOT_SAFX,                  ///< N                       Algorithmic Synthesis and Audio Effects
     AOT_ER_AAC_LC,             ///< N                       Error Resilient Low Complexity
     AOT_ER_AAC_LTP       = 19, ///< N                       Error Resilient Long Term Prediction
     AOT_ER_AAC_SCALABLE,       ///< N                       Error Resilient Scalable
     AOT_ER_TWINVQ,             ///< N                       Error Resilient Twin Vector Quantizer
     AOT_ER_BSAC,               ///< N                       Error Resilient Bit-Sliced Arithmetic Coding
     AOT_ER_AAC_LD,             ///< N                       Error Resilient Low Delay
     AOT_ER_CELP,               ///< N                       Error Resilient Code Excited Linear Prediction
     AOT_ER_HVXC,               ///< N                       Error Resilient Harmonic Vector eXcitation Coding
     AOT_ER_HILN,               ///< N                       Error Resilient Harmonic and Individual Lines plus Noise
     AOT_ER_PARAM,              ///< N                       Error Resilient Parametric
     AOT_SSC,                   ///< N                       SinuSoidal Coding
 };

 enum RawDataBlockType {
     TYPE_SCE,
     TYPE_CPE,
     TYPE_CCE,
     TYPE_LFE,
     TYPE_DSE,
     TYPE_PCE,
     TYPE_FIL,
     TYPE_END,
 };

 enum ExtensionPayloadID {
     EXT_FILL,
     EXT_FILL_DATA,
     EXT_DATA_ELEMENT,
     EXT_DYNAMIC_RANGE = 0xb,
     EXT_SBR_DATA      = 0xd,
     EXT_SBR_DATA_CRC  = 0xe,
 };

 enum WindowSequence {
     ONLY_LONG_SEQUENCE,
     LONG_START_SEQUENCE,
     EIGHT_SHORT_SEQUENCE,
     LONG_STOP_SEQUENCE,
 };

 enum BandType {
     ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
     FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
     ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
     NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
     INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
     INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 };

 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)

 enum ChannelPosition {
     AAC_CHANNEL_FRONT = 1,
     AAC_CHANNEL_SIDE  = 2,
     AAC_CHANNEL_BACK  = 3,
     AAC_CHANNEL_LFE   = 4,
     AAC_CHANNEL_CC    = 5,
 };

 /**
  * The point during decoding at which channel coupling is applied.
  */
 enum CouplingPoint {
     BEFORE_TNS,
     BETWEEN_TNS_AND_IMDCT,
     AFTER_IMDCT = 3,
 };

 /**
  * Predictor State
  */
 typedef struct {
     float cor0;
     float cor1;
     float var0;
     float var1;
     float r0;
     float r1;
 } PredictorState;

 #define MAX_PREDICTORS 672

 /**
  * Individual Channel Stream
  */
 typedef struct {
     uint8_t max_sfb;            ///< number of scalefactor bands per group
     enum WindowSequence window_sequence[2];
     uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
     int num_window_groups;
     uint8_t group_len[8];
     const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
     int num_swb;                ///< number of scalefactor window bands
     int num_windows;
     int tns_max_bands;
     int predictor_present;
     int predictor_initialized;
     int predictor_reset_group;
     uint8_t prediction_used[41];
 } IndividualChannelStream;

 /**
  * Temporal Noise Shaping
  */
 typedef struct {
     int present;
     int n_filt[8];
     int length[8][4];
     int direction[8][4];
     int order[8][4];
     float coef[8][4][TNS_MAX_ORDER];
 } TemporalNoiseShaping;

 /**
  * Dynamic Range Control - decoded from the bitstream but not processed further.
  */
 typedef struct {
     int pce_instance_tag;                           ///< Indicates with which program the DRC info is associated.
     int dyn_rng_sgn[17];                            ///< DRC sign information; 0 - positive, 1 - negative
     int dyn_rng_ctl[17];                            ///< DRC magnitude information
     int exclude_mask[MAX_CHANNELS];                 ///< Channels to be excluded from DRC processing.
     int band_incr;                                  ///< Number of DRC bands greater than 1 having DRC info.
     int interpolation_scheme;                       ///< Indicates the interpolation scheme used in the SBR QMF domain.
     int band_top[17];                               ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
     int prog_ref_level;                             /**< A reference level for the long-term program audio level for all
                                                      *   channels combined.
                                                      */
 } DynamicRangeControl;

 typedef struct {
     int num_pulse;
     int pos[4];
     int amp[4];
 } Pulse;

 /**
  * coupling parameters
  */
 typedef struct {
     enum CouplingPoint coupling_point;  ///< The point during decoding at which coupling is applied.
     int num_coupled;       ///< number of target elements
     enum RawDataBlockType type[8];   ///< Type of channel element to be coupled - SCE or CPE.
     int id_select[8];      ///< element id
     int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
                             *   [2] list of gains for left channel; [3] lists of gains for both channels
                             */
     float gain[16][120];
 } ChannelCoupling;

 /**
  * Single Channel Element - used for both SCE and LFE elements.
  */
 typedef struct {
     IndividualChannelStream ics;
     TemporalNoiseShaping tns;
     enum BandType band_type[120];             ///< band types
     int band_type_run_end[120];               ///< band type run end points
     float sf[120];                            ///< scalefactors
     DECLARE_ALIGNED_16(float, coeffs[1024]);  ///< coefficients for IMDCT
     DECLARE_ALIGNED_16(float, saved[512]);    ///< overlap
     DECLARE_ALIGNED_16(float, ret[1024]);     ///< PCM output
     PredictorState predictor_state[MAX_PREDICTORS];
 } SingleChannelElement;

 /**
  * channel element - generic struct for SCE/CPE/CCE/LFE
  */
 typedef struct {
     // CPE specific
     uint8_t ms_mask[120];     ///< Set if mid/side stereo is used for each scalefactor window band
     // shared
     SingleChannelElement ch[2];
     // CCE specific
     ChannelCoupling coup;
 } ChannelElement;

 /**
  * main AAC context
  */
 typedef struct {
     AVCodecContext * avccontext;

     MPEG4AudioConfig m4ac;

     int is_saved;                 ///< Set if elements have stored overlap from previous frame.
     DynamicRangeControl che_drc;

     /**
      * @defgroup elements Channel element related data.
      * @{
      */
     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
                                                    *   first index as the first 4 raw data block types
                                                    */
     ChannelElement * che[4][MAX_ELEM_ID];
     ChannelElement * tag_che_map[4][MAX_ELEM_ID];
     int tags_mapped;
     /** @} */

     /**
      * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
      * @{
      */
     DECLARE_ALIGNED_16(float, buf_mdct[1024]);
     /** @} */

     /**
      * @defgroup tables   Computed / set up during initialization.
      * @{
      */
     MDCTContext mdct;
     MDCTContext mdct_small;
     DSPContext dsp;
     int random_state;
     /** @} */

     /**
      * @defgroup output   Members used for output interleaving.
      * @{
      */
     float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
     float add_bias;                                   ///< offset for dsp.float_to_int16
     float sf_scale;                                   ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
     int sf_offset;                                    ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
     /** @} */

     DECLARE_ALIGNED(16, float, temp[128]);
 } AACContext;

 #endif /* AVCODEC_AAC_H */
	/*
	* AAC definitions and structures
	* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
	* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	/**
	* @file libavcodec/aac.h
	* AAC definitions and structures
	* @author Oded Shimon ( ods15 ods15 dyndns org )
	* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
	*/

	#ifndef AVCODEC_AAC_H
	#define AVCODEC_AAC_H

	#include "libavutil/internal.h"
	#include "avcodec.h"
	#include "dsputil.h"
	#include "mpeg4audio.h"

	#include <stdint.h>

	#define AAC_INIT_VLC_STATIC(num, size) \
	INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
	ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
	ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
	size);

	#define MAX_CHANNELS 64
	#define MAX_ELEM_ID 16

	#define TNS_MAX_ORDER 20

	enum AudioObjectType {
	AOT_NULL,
	// Support? Name
	AOT_AAC_MAIN, ///< Y Main
	AOT_AAC_LC, ///< Y Low Complexity
	AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
	AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
	AOT_SBR, ///< N (in progress) Spectral Band Replication
	AOT_AAC_SCALABLE, ///< N Scalable
	AOT_TWINVQ, ///< N Twin Vector Quantizer
	AOT_CELP, ///< N Code Excited Linear Prediction
	AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
	AOT_TTSI = 12, ///< N Text-To-Speech Interface
	AOT_MAINSYNTH, ///< N Main Synthesis
	AOT_WAVESYNTH, ///< N Wavetable Synthesis
	AOT_MIDI, ///< N General MIDI
	AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
	AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
	AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
	AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
	AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
	AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
	AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
	AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
	AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
	AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
	AOT_ER_PARAM, ///< N Error Resilient Parametric
	AOT_SSC, ///< N SinuSoidal Coding
	};

	enum RawDataBlockType {
	TYPE_SCE,
	TYPE_CPE,
	TYPE_CCE,
	TYPE_LFE,
	TYPE_DSE,
	TYPE_PCE,
	TYPE_FIL,
	TYPE_END,
	};

	enum ExtensionPayloadID {
	EXT_FILL,
	EXT_FILL_DATA,
	EXT_DATA_ELEMENT,
	EXT_DYNAMIC_RANGE = 0xb,
	EXT_SBR_DATA = 0xd,
	EXT_SBR_DATA_CRC = 0xe,
	};

	enum WindowSequence {
	ONLY_LONG_SEQUENCE,
	LONG_START_SEQUENCE,
	EIGHT_SHORT_SEQUENCE,
	LONG_STOP_SEQUENCE,
	};

	enum BandType {
	ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
	FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
	ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
	NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
	INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions.
	INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions.
	};

	#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)

	enum ChannelPosition {
	AAC_CHANNEL_FRONT = 1,
	AAC_CHANNEL_SIDE = 2,
	AAC_CHANNEL_BACK = 3,
	AAC_CHANNEL_LFE = 4,
	AAC_CHANNEL_CC = 5,
	};

	/**
	* The point during decoding at which channel coupling is applied.
	*/
	enum CouplingPoint {
	BEFORE_TNS,
	BETWEEN_TNS_AND_IMDCT,
	AFTER_IMDCT = 3,
	};

	/**
	* Predictor State
	*/
	typedef struct {
	float cor0;
	float cor1;
	float var0;
	float var1;
	float r0;
	float r1;
	} PredictorState;

	#define MAX_PREDICTORS 672

	/**
	* Individual Channel Stream
	*/
	typedef struct {
	uint8_t max_sfb; ///< number of scalefactor bands per group
	enum WindowSequence window_sequence[2];
	uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
	int num_window_groups;
	uint8_t group_len[8];
	const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
	int num_swb; ///< number of scalefactor window bands
	int num_windows;
	int tns_max_bands;
	int predictor_present;
	int predictor_initialized;
	int predictor_reset_group;
	uint8_t prediction_used[41];
	} IndividualChannelStream;

	/**
	* Temporal Noise Shaping
	*/
	typedef struct {
	int present;
	int n_filt[8];
	int length[8][4];
	int direction[8][4];
	int order[8][4];
	float coef[8][4][TNS_MAX_ORDER];
	} TemporalNoiseShaping;

	/**
	* Dynamic Range Control - decoded from the bitstream but not processed further.
	*/
	typedef struct {
	int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
	int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
	int dyn_rng_ctl[17]; ///< DRC magnitude information
	int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
	int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
	int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
	int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
	int prog_ref_level; /**< A reference level for the long-term program audio level for all
	* channels combined.
	*/
	} DynamicRangeControl;

	typedef struct {
	int num_pulse;
	int pos[4];
	int amp[4];
	} Pulse;

	/**
	* coupling parameters
	*/
	typedef struct {
	enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
	int num_coupled; ///< number of target elements
	enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
	int id_select[8]; ///< element id
	int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
	* [2] list of gains for left channel; [3] lists of gains for both channels
	*/
	float gain[16][120];
	} ChannelCoupling;

	/**
	* Single Channel Element - used for both SCE and LFE elements.
	*/
	typedef struct {
	IndividualChannelStream ics;
	TemporalNoiseShaping tns;
	enum BandType band_type[120]; ///< band types
	int band_type_run_end[120]; ///< band type run end points
	float sf[120]; ///< scalefactors
	DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
	DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
	DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
	PredictorState predictor_state[MAX_PREDICTORS];
	} SingleChannelElement;

	/**
	* channel element - generic struct for SCE/CPE/CCE/LFE
	*/
	typedef struct {
	// CPE specific
	uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
	// shared
	SingleChannelElement ch[2];
	// CCE specific
	ChannelCoupling coup;
	} ChannelElement;

	/**
	* main AAC context
	*/
	typedef struct {
	AVCodecContext * avccontext;

	MPEG4AudioConfig m4ac;

	int is_saved; ///< Set if elements have stored overlap from previous frame.
	DynamicRangeControl che_drc;

	/**
	* @defgroup elements Channel element related data.
	* @{
	*/
	enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
	* first index as the first 4 raw data block types
	*/
	ChannelElement * che[4][MAX_ELEM_ID];
	ChannelElement * tag_che_map[4][MAX_ELEM_ID];
	int tags_mapped;
	/** @} */

	/**
	* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
	* @{
	*/
	DECLARE_ALIGNED_16(float, buf_mdct[1024]);
	/** @} */

	/**
	* @defgroup tables Computed / set up during initialization.
	* @{
	*/
	MDCTContext mdct;
	MDCTContext mdct_small;
	DSPContext dsp;
	int random_state;
	/** @} */

	/**
	* @defgroup output Members used for output interleaving.
	* @{
	*/
	float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
	float add_bias; ///< offset for dsp.float_to_int16
	float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
	int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
	/** @} */

	DECLARE_ALIGNED(16, float, temp[128]);
	} AACContext;

	#endif /* AVCODEC_AAC_H */