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Update avcodec to 20080825

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git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@27539 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
David McPaul 2008-09-15 13:57:54 +00:00
parent ab1b76ccbe
commit b44406a7fc
18 changed files with 3993 additions and 1157 deletions
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275
src/add-ons/media/plugins/avcodec/libavcodec/aac.h Normal file
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/*
* 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 aac.h
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef FFMPEG_AAC_H
#define FFMPEG_AAC_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
#define PNS_MEAN_ENERGY 3719550720.0f // sqrt(3.0) * 1<<31
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,
};
/**
* 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;
} 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 left channel;
* [2] list of gains for right 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
} 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
* @{
*/
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];
/** @} */
/**
* @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
/** @} */
} AACContext;
#endif /* FFMPEG_AAC_H */

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src/add-ons/media/plugins/avcodec/libavcodec/aac_ac3_parser.h Normal file
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/*
* Common AAC and AC-3 parser prototypes
* Copyright (c) 2003 Fabrice Bellard.
* Copyright (c) 2003 Michael Niedermayer.
*
* 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
*/
#ifndef FFMPEG_AAC_AC3_PARSER_H
#define FFMPEG_AAC_AC3_PARSER_H
#include <stdint.h>
#include "avcodec.h"
#include "parser.h"
typedef struct AACAC3ParseContext {
ParseContext pc;
int frame_size;
int header_size;
int (*sync)(uint64_t state, struct AACAC3ParseContext *hdr_info,
int *need_next_header, int *new_frame_start);
int channels;
int sample_rate;
int bit_rate;
int samples;
int remaining_size;
uint64_t state;
int need_next_header;
} AACAC3ParseContext;
int ff_aac_ac3_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size);
#endif /* FFMPEG_AAC_AC3_PARSER_H */

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src/add-ons/media/plugins/avcodec/libavcodec/aacdectab.h Normal file
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/*
* AAC decoder data
* 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 aacdectab.h
* AAC decoder data
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef FFMPEG_AACDECTAB_H
#define FFMPEG_AACDECTAB_H
#include "aac.h"
#include <stdint.h>
/* @name swb_offsets
* Sample offset into the window indicating the beginning of a scalefactor
* window band
*
* scalefactor window band - term for scalefactor bands within a window,
* given in Table 4.110 to Table 4.128.
*
* scalefactor band - a set of spectral coefficients which are scaled by one
* scalefactor. In case of EIGHT_SHORT_SEQUENCE and grouping a scalefactor band
* may contain several scalefactor window bands of corresponding frequency. For
* all other window_sequences scalefactor bands and scalefactor window bands are
* identical.
* @{
*/
static const uint16_t swb_offset_1024_96[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 48, 52, 56, 64,
72, 80, 88, 96, 108, 120, 132, 144,
156, 172, 188, 212, 240, 276, 320, 384,
448, 512, 576, 640, 704, 768, 832, 896,
960, 1024
};
static const uint16_t swb_offset_128_96[] = {
0, 4, 8, 12, 16, 20, 24, 32, 40, 48, 64, 92, 128
};
static const uint16_t swb_offset_1024_64[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 48, 52, 56, 64,
72, 80, 88, 100, 112, 124, 140, 156,
172, 192, 216, 240, 268, 304, 344, 384,
424, 464, 504, 544, 584, 624, 664, 704,
744, 784, 824, 864, 904, 944, 984, 1024
};
static const uint16_t swb_offset_1024_48[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 48, 56, 64, 72, 80,
88, 96, 108, 120, 132, 144, 160, 176,
196, 216, 240, 264, 292, 320, 352, 384,
416, 448, 480, 512, 544, 576, 608, 640,
672, 704, 736, 768, 800, 832, 864, 896,
928, 1024
};
static const uint16_t swb_offset_128_48[] = {
0, 4, 8, 12, 16, 20, 28, 36,
44, 56, 68, 80, 96, 112, 128
};
static const uint16_t swb_offset_1024_32[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 48, 56, 64, 72, 80,
88, 96, 108, 120, 132, 144, 160, 176,
196, 216, 240, 264, 292, 320, 352, 384,
416, 448, 480, 512, 544, 576, 608, 640,
672, 704, 736, 768, 800, 832, 864, 896,
928, 960, 992, 1024
};
static const uint16_t swb_offset_1024_24[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 52, 60, 68, 76,
84, 92, 100, 108, 116, 124, 136, 148,
160, 172, 188, 204, 220, 240, 260, 284,
308, 336, 364, 396, 432, 468, 508, 552,
600, 652, 704, 768, 832, 896, 960, 1024
};
static const uint16_t swb_offset_128_24[] = {
0, 4, 8, 12, 16, 20, 24, 28,
36, 44, 52, 64, 76, 92, 108, 128
};
static const uint16_t swb_offset_1024_16[] = {
0, 8, 16, 24, 32, 40, 48, 56,
64, 72, 80, 88, 100, 112, 124, 136,
148, 160, 172, 184, 196, 212, 228, 244,
260, 280, 300, 320, 344, 368, 396, 424,
456, 492, 532, 572, 616, 664, 716, 772,
832, 896, 960, 1024
};
static const uint16_t swb_offset_128_16[] = {
0, 4, 8, 12, 16, 20, 24, 28,
32, 40, 48, 60, 72, 88, 108, 128
};
static const uint16_t swb_offset_1024_8[] = {
0, 12, 24, 36, 48, 60, 72, 84,
96, 108, 120, 132, 144, 156, 172, 188,
204, 220, 236, 252, 268, 288, 308, 328,
348, 372, 396, 420, 448, 476, 508, 544,
580, 620, 664, 712, 764, 820, 880, 944,
1024
};
static const uint16_t swb_offset_128_8[] = {
0, 4, 8, 12, 16, 20, 24, 28,
36, 44, 52, 60, 72, 88, 108, 128
};
static const uint16_t *swb_offset_1024[] = {
swb_offset_1024_96, swb_offset_1024_96, swb_offset_1024_64,
swb_offset_1024_48, swb_offset_1024_48, swb_offset_1024_32,
swb_offset_1024_24, swb_offset_1024_24, swb_offset_1024_16,
swb_offset_1024_16, swb_offset_1024_16, swb_offset_1024_8
};
static const uint16_t *swb_offset_128[] = {
/* The last entry on the following row is swb_offset_128_64 but is a
duplicate of swb_offset_128_96. */
swb_offset_128_96, swb_offset_128_96, swb_offset_128_96,
swb_offset_128_48, swb_offset_128_48, swb_offset_128_48,
swb_offset_128_24, swb_offset_128_24, swb_offset_128_16,
swb_offset_128_16, swb_offset_128_16, swb_offset_128_8
};
// @}
/* @name tns_max_bands
* The maximum number of scalefactor bands on which TNS can operate for the long
* and short transforms respectively. The index to these tables is related to
* the sample rate of the audio.
* @{
*/
static const uint8_t tns_max_bands_1024[] = {
31, 31, 34, 40, 42, 51, 46, 46, 42, 42, 42, 39
};
static const uint8_t tns_max_bands_128[] = {
9, 9, 10, 14, 14, 14, 14, 14, 14, 14, 14, 14
};
// @}
/* @name tns_tmp2_map
* Tables of the tmp2[] arrays of LPC coefficients used for TNS.
* The suffix _M_N[] indicate the values of coef_compress and coef_res
* respectively.
* @{
*/
static const float tns_tmp2_map_1_3[4] = {
0.00000000, 0.43388373, -0.64278758, -0.34202015,
};
static const float tns_tmp2_map_0_3[8] = {
0.00000000, 0.43388373, 0.78183150, 0.97492790,
-0.98480773, -0.86602539, -0.64278758, -0.34202015,
};
static const float tns_tmp2_map_1_4[8] = {
0.00000000, 0.20791170, 0.40673664, 0.58778524,
-0.67369562, -0.52643216, -0.36124167, -0.18374951,
};
static const float tns_tmp2_map_0_4[16] = {
0.00000000, 0.20791170, 0.40673664, 0.58778524,
0.74314481, 0.86602539, 0.95105654, 0.99452192,
-0.99573416, -0.96182561, -0.89516330, -0.79801720,
-0.67369562, -0.52643216, -0.36124167, -0.18374951,
};
static const float *tns_tmp2_map[4] = {
tns_tmp2_map_0_3,
tns_tmp2_map_0_4,
tns_tmp2_map_1_3,
tns_tmp2_map_1_4
};
// @}
#endif /* FFMPEG_AACDECTAB_H */

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src/add-ons/media/plugins/avcodec/libavcodec/aacpsy.h Normal file
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/*
* AAC encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* 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
*/
#ifndef FFMPEG_AACPSY_H
#define FFMPEG_AACPSY_H
#include "avcodec.h"
#include "aac.h"
#include "lowpass.h"
enum AACPsyModelType{
AAC_PSY_TEST, ///< a sample model to exercise encoder
AAC_PSY_3GPP, ///< model following recommendations from 3GPP TS 26.403
AAC_NB_PSY_MODELS ///< total number of psychoacoustic models, since it's not a part of the ABI new models can be added freely
};
/**
* context used by psychoacoustic model
*/
typedef struct AACPsyContext {
AVCodecContext *avctx; ///< encoder context
}AACPsyContext;
/**
* Cleanup model context at the end.
*
* @param ctx model context
*/
void ff_aac_psy_end(AACPsyContext *ctx);
#endif /* FFMPEG_AACPSY_H */

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/*
* AAC data declarations
* 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 aactab.h
* AAC data declarations
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef FFMPEG_AACTAB_H
#define FFMPEG_AACTAB_H
#include "mem.h"
#include "aac.h"
#include <stdint.h>
/* NOTE:
* Tables in this file are used by the AAC decoder and will be used by the AAC
* encoder.
*/
/* @name window coefficients
* @{
*/
DECLARE_ALIGNED(16, extern float, ff_aac_kbd_long_1024[1024]);
DECLARE_ALIGNED(16, extern float, ff_aac_kbd_short_128[128]);
// @}
/* @name number of scalefactor window bands for long and short transform windows respectively
* @{
*/
extern const uint8_t ff_aac_num_swb_1024[];
extern const uint8_t ff_aac_num_swb_128 [];
// @}
extern const uint32_t ff_aac_scalefactor_code[121];
extern const uint8_t ff_aac_scalefactor_bits[121];
extern const uint16_t *ff_aac_spectral_codes[11];
extern const uint8_t *ff_aac_spectral_bits [11];
extern const uint16_t ff_aac_spectral_sizes[11];
extern const float *ff_aac_codebook_vectors[];
#ifdef CONFIG_HARDCODED_TABLES
extern const float ff_aac_pow2sf_tab[316];
#else
extern float ff_aac_pow2sf_tab[316];
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* FFMPEG_AACTAB_H */

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src/add-ons/media/plugins/avcodec/libavcodec/ac3.h
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/*
* Common code between AC3 encoder and decoder
* Common code between the AC-3 encoder and decoder
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard.
*
* This library is free software; you can redistribute it and/or
* 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 of the License, or (at your option) any later version.
* version 2.1 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file ac3.h
* Common code between AC3 encoder and decoder.
* Common code between the AC-3 encoder and decoder.
*/
#ifndef FFMPEG_AC3_H
#define FFMPEG_AC3_H
#include "ac3tab.h"
#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
#define AC3_MAX_CHANNELS 6 /* including LFE channel */
#define NB_BLOCKS 6 /* number of PCM blocks inside an AC3 frame */
#define NB_BLOCKS 6 /* number of PCM blocks inside an AC-3 frame */
#define AC3_FRAME_SIZE (NB_BLOCKS * 256)
/* exponent encoding strategy */
@ -36,26 +43,143 @@
#define EXP_D25 2
#define EXP_D45 3
/** Delta bit allocation strategy */
typedef enum {
DBA_REUSE = 0,
DBA_NEW,
DBA_NONE,
DBA_RESERVED
} AC3DeltaStrategy;
/** Channel mode (audio coding mode) */
typedef enum {
AC3_CHMODE_DUALMONO = 0,
AC3_CHMODE_MONO,
AC3_CHMODE_STEREO,
AC3_CHMODE_3F,
AC3_CHMODE_2F1R,
AC3_CHMODE_3F1R,
AC3_CHMODE_2F2R,
AC3_CHMODE_3F2R
} AC3ChannelMode;
typedef struct AC3BitAllocParameters {
int fscod; /* frequency */
int halfratecod;
int sgain, sdecay, fdecay, dbknee, floor;
int cplfleak, cplsleak;
int sr_code;
int sr_shift;
int slow_gain, slow_decay, fast_decay, db_per_bit, floor;
int cpl_fast_leak, cpl_slow_leak;
} AC3BitAllocParameters;
extern const uint16_t ac3_freqs[3];
extern const uint16_t ac3_bitratetab[19];
extern const int16_t ac3_window[256];
extern const uint8_t sdecaytab[4];
extern const uint8_t fdecaytab[4];
extern const uint16_t sgaintab[4];
extern const uint16_t dbkneetab[4];
extern const uint16_t floortab[8];
extern const uint16_t fgaintab[8];
/**
* @struct AC3HeaderInfo
* Coded AC-3 header values up to the lfeon element, plus derived values.
*/
typedef struct {
/** @defgroup coded Coded elements
* @{
*/
uint16_t sync_word;
uint16_t crc1;
uint8_t sr_code;
uint8_t bitstream_id;
uint8_t channel_mode;
uint8_t lfe_on;
uint8_t frame_type;
int substreamid; ///< substream identification
int center_mix_level; ///< Center mix level index
int surround_mix_level; ///< Surround mix level index
uint16_t channel_map;
int num_blocks; ///< number of audio blocks
/** @} */
/** @defgroup derived Derived values
* @{
*/
uint8_t sr_shift;
uint16_t sample_rate;
uint32_t bit_rate;
uint8_t channels;
uint16_t frame_size;
/** @} */
} AC3HeaderInfo;
typedef enum {
EAC3_FRAME_TYPE_INDEPENDENT = 0,
EAC3_FRAME_TYPE_DEPENDENT,
EAC3_FRAME_TYPE_AC3_CONVERT,
EAC3_FRAME_TYPE_RESERVED
} EAC3FrameType;
void ac3_common_init(void);
/**
* Calculates the log power-spectral density of the input signal.
* This gives a rough estimate of signal power in the frequency domain by using
* the spectral envelope (exponents). The psd is also separately grouped
* into critical bands for use in the calculating the masking curve.
* 128 units in psd = -6 dB. The dbknee parameter in AC3BitAllocParameters
* determines the reference level.
*
* @param[in] exp frequency coefficient exponents
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[out] psd signal power for each frequency bin
* @param[out] band_psd signal power for each critical band
*/
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
int16_t *band_psd);
/**
* Calculates the masking curve.
* First, the excitation is calculated using parameters in \p s and the signal
* power in each critical band. The excitation is compared with a predefined
* hearing threshold table to produce the masking curve. If delta bit
* allocation information is provided, it is used for adjusting the masking
* curve, usually to give a closer match to a better psychoacoustic model.
*
* @param[in] s adjustable bit allocation parameters
* @param[in] band_psd signal power for each critical band
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[in] fast_gain fast gain (estimated signal-to-mask ratio)
* @param[in] is_lfe whether or not the channel being processed is the LFE
* @param[in] dba_mode delta bit allocation mode (none, reuse, or new)
* @param[in] dba_nsegs number of delta segments
* @param[in] dba_offsets location offsets for each segment
* @param[in] dba_lengths length of each segment
* @param[in] dba_values delta bit allocation for each segment
* @param[out] mask calculated masking curve
*/
void ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int start, int end, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
uint8_t *dba_lengths, uint8_t *dba_values,
int16_t *mask);
/**
* Calculates bit allocation pointers.
* The SNR is the difference between the masking curve and the signal. AC-3
* uses this value for each frequency bin to allocate bits. The \p snroffset
* parameter is a global adjustment to the SNR for all bins.
*
* @param[in] mask masking curve
* @param[in] psd signal power for each frequency bin
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[in] snr_offset SNR adjustment
* @param[in] floor noise floor
* @param[in] bap_tab look-up table for bit allocation pointers
* @param[out] bap bit allocation pointers
*/
void ff_ac3_bit_alloc_calc_bap(int16_t *mask, int16_t *psd, int start, int end,
int snr_offset, int floor,
const uint8_t *bap_tab, uint8_t *bap);
void ac3_parametric_bit_allocation(AC3BitAllocParameters *s, uint8_t *bap,
int8_t *exp, int start, int end,
int snroffset, int fgain, int is_lfe,
int deltbae,int deltnseg,
uint8_t *deltoffst, uint8_t *deltlen, uint8_t *deltba);
int snr_offset, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs,
uint8_t *dba_offsets, uint8_t *dba_lengths,
uint8_t *dba_values);
#endif /* FFMPEG_AC3_H */

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@ -0,0 +1,61 @@
/*
* AC-3 parser prototypes
* Copyright (c) 2003 Fabrice Bellard.
* Copyright (c) 2003 Michael Niedermayer.
*
* 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
*/
#ifndef FFMPEG_AC3_PARSER_H
#define FFMPEG_AC3_PARSER_H
#include "ac3.h"
#include "bitstream.h"
typedef enum {
AC3_PARSE_ERROR_SYNC = -1,
AC3_PARSE_ERROR_BSID = -2,
AC3_PARSE_ERROR_SAMPLE_RATE = -3,
AC3_PARSE_ERROR_FRAME_SIZE = -4,
AC3_PARSE_ERROR_FRAME_TYPE = -5,
AC3_PARSE_ERROR_CRC = -6,
} AC3ParseError;
/**
* Parses AC-3 frame header.
* Parses the header up to the lfeon element, which is the first 52 or 54 bits
* depending on the audio coding mode.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return Returns 0 on success, -1 if there is a sync word mismatch,
* -2 if the bsid (version) element is invalid, -3 if the fscod (sample rate)
* element is invalid, or -4 if the frmsizecod (bit rate) element is invalid.
*/
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr);
/**
* Parses AC-3 frame header and sets channel_map
* Parses the header up to the lfeon (channel_map in E-AC-3)
* element, which is the first 52, 54 or 104 bits depending
* on the audio coding mode.
* @param gbc[in] BitContext containing the first 54 bits of the frame.
* @param hdr[out] Pointer to struct where header info is written.
* @return value returned by ff_ac3_parse_header
*/
int ff_ac3_parse_header_full(GetBitContext *gbc, AC3HeaderInfo *hdr);
#endif /* FFMPEG_AC3_PARSER_H */

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@ -0,0 +1,171 @@
/*
* Common code between the AC-3 and E-AC-3 decoders
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@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 ac3.h
* Common code between the AC-3 and E-AC-3 decoders.
*/
#ifndef FFMPEG_AC3DEC_H
#define FFMPEG_AC3DEC_H
#include "lfg.h"
#include "ac3.h"
#include "bitstream.h"
#include "dsputil.h"
/* override ac3.h to include coupling channel */
#undef AC3_MAX_CHANNELS
#define AC3_MAX_CHANNELS 7
#define CPL_CH 0
#define AC3_OUTPUT_LFEON 8
#define AC3_MAX_COEFS 256
#define AC3_BLOCK_SIZE 256
#define MAX_BLOCKS 6
typedef struct {
AVCodecContext *avctx; ///< parent context
GetBitContext gbc; ///< bitstream reader
uint8_t *input_buffer; ///< temp buffer to prevent overread
///@defgroup bsi bit stream information
///@{
int frame_type; ///< frame type (strmtyp)
int substreamid; ///< substream identification
int frame_size; ///< current frame size, in bytes
int bit_rate; ///< stream bit rate, in bits-per-second
int sample_rate; ///< sample frequency, in Hz
int num_blocks; ///< number of audio blocks
int channel_mode; ///< channel mode (acmod)
int lfe_on; ///< lfe channel in use
int channel_map; ///< custom channel map
int center_mix_level; ///< Center mix level index
int surround_mix_level; ///< Surround mix level index
int eac3; ///< indicates if current frame is E-AC-3
///@}
///@defgroup audfrm frame syntax parameters
int snr_offset_strategy; ///< SNR offset strategy (snroffststr)
int block_switch_syntax; ///< block switch syntax enabled (blkswe)
int dither_flag_syntax; ///< dither flag syntax enabled (dithflage)
int bit_allocation_syntax; ///< bit allocation model syntax enabled (bamode)
int fast_gain_syntax; ///< fast gain codes enabled (frmfgaincode)
int dba_syntax; ///< delta bit allocation syntax enabled (dbaflde)
int skip_syntax; ///< skip field syntax enabled (skipflde)
///@}
///@defgroup cpl standard coupling
int cpl_in_use[MAX_BLOCKS]; ///< coupling in use (cplinu)
int cpl_strategy_exists[MAX_BLOCKS]; ///< coupling strategy exists (cplstre)
int channel_in_cpl[AC3_MAX_CHANNELS]; ///< channel in coupling (chincpl)
int phase_flags_in_use; ///< phase flags in use (phsflginu)
int phase_flags[18]; ///< phase flags (phsflg)
int num_cpl_subbands; ///< number of coupling sub bands (ncplsubnd)
int num_cpl_bands; ///< number of coupling bands (ncplbnd)
int cpl_band_struct[18]; ///< coupling band structure (cplbndstrc)
int firstchincpl; ///< first channel in coupling
int first_cpl_coords[AC3_MAX_CHANNELS]; ///< first coupling coordinates states (firstcplcos)
int cpl_coords[AC3_MAX_CHANNELS][18]; ///< coupling coordinates (cplco)
///@}
///@defgroup aht adaptive hybrid transform
int channel_uses_aht[AC3_MAX_CHANNELS]; ///< channel AHT in use (chahtinu)
int pre_mantissa[AC3_MAX_CHANNELS][AC3_MAX_COEFS][MAX_BLOCKS]; ///< pre-IDCT mantissas
///@}
///@defgroup channel channel
int fbw_channels; ///< number of full-bandwidth channels
int channels; ///< number of total channels
int lfe_ch; ///< index of LFE channel
float downmix_coeffs[AC3_MAX_CHANNELS][2]; ///< stereo downmix coefficients
int downmixed; ///< indicates if coeffs are currently downmixed
int output_mode; ///< output channel configuration
int out_channels; ///< number of output channels
///@}
///@defgroup dynrng dynamic range
float dynamic_range[2]; ///< dynamic range
///@}
///@defgroup bandwidth bandwidth
int start_freq[AC3_MAX_CHANNELS]; ///< start frequency bin (strtmant)
int end_freq[AC3_MAX_CHANNELS]; ///< end frequency bin (endmant)
///@}
///@defgroup rematrixing rematrixing
int num_rematrixing_bands; ///< number of rematrixing bands (nrematbnd)
int rematrixing_flags[4]; ///< rematrixing flags (rematflg)
///@}
///@defgroup exponents exponents
int num_exp_groups[AC3_MAX_CHANNELS]; ///< Number of exponent groups (nexpgrp)
int8_t dexps[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< decoded exponents
int exp_strategy[MAX_BLOCKS][AC3_MAX_CHANNELS]; ///< exponent strategies (expstr)
///@}
///@defgroup bitalloc bit allocation
AC3BitAllocParameters bit_alloc_params; ///< bit allocation parameters
int first_cpl_leak; ///< first coupling leak state (firstcplleak)
int snr_offset[AC3_MAX_CHANNELS]; ///< signal-to-noise ratio offsets (snroffst)
int fast_gain[AC3_MAX_CHANNELS]; ///< fast gain values/SMR's (fgain)
uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< bit allocation pointers
int16_t psd[AC3_MAX_CHANNELS][AC3_MAX_COEFS]; ///< scaled exponents
int16_t band_psd[AC3_MAX_CHANNELS][50]; ///< interpolated exponents
int16_t mask[AC3_MAX_CHANNELS][50]; ///< masking curve values
int dba_mode[AC3_MAX_CHANNELS]; ///< delta bit allocation mode
int dba_nsegs[AC3_MAX_CHANNELS]; ///< number of delta segments
uint8_t dba_offsets[AC3_MAX_CHANNELS][8]; ///< delta segment offsets
uint8_t dba_lengths[AC3_MAX_CHANNELS][8]; ///< delta segment lengths
uint8_t dba_values[AC3_MAX_CHANNELS][8]; ///< delta values for each segment
///@}
///@defgroup dithering zero-mantissa dithering
int dither_all; ///< true if all channels are dithered
int dither_flag[AC3_MAX_CHANNELS]; ///< dither flags (dithflg)
AVLFG dith_state; ///< for dither generation
///@}
///@defgroup imdct IMDCT
int block_switch[AC3_MAX_CHANNELS]; ///< block switch flags (blksw)
MDCTContext imdct_512; ///< for 512 sample IMDCT
MDCTContext imdct_256; ///< for 256 sample IMDCT
///@}
///@defgroup opt optimization
DSPContext dsp; ///< for optimization
float add_bias; ///< offset for float_to_int16 conversion
float mul_bias; ///< scaling for float_to_int16 conversion
///@}
DECLARE_ALIGNED_16(int, fixed_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]); ///> fixed-point transform coefficients
///@defgroup arrays aligned arrays
DECLARE_ALIGNED_16(float, transform_coeffs[AC3_MAX_CHANNELS][AC3_MAX_COEFS]); ///< transform coefficients
DECLARE_ALIGNED_16(float, delay[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]); ///< delay - added to the next block
DECLARE_ALIGNED_16(float, window[AC3_BLOCK_SIZE]); ///< window coefficients
DECLARE_ALIGNED_16(float, tmp_output[AC3_BLOCK_SIZE]); ///< temporary storage for output before windowing
DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]); ///< output after imdct transform and windowing
///@}
} AC3DecodeContext;
#endif /* FFMPEG_AC3DEC_H */

40
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@ -0,0 +1,40 @@
/*
* AC-3 and E-AC-3 decoder tables
* Copyright (c) 2007 Bartlomiej Wolowiec <bartek.wolowiec@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
*/
#ifndef FFMPEG_AC3DEC_DATA_H
#define FFMPEG_AC3DEC_DATA_H
#include "common.h"
extern const uint8_t ff_ac3_ungroup_3_in_5_bits_tab[32][3];
extern const uint8_t ff_eac3_hebap_tab[64];
extern const uint8_t ff_eac3_bits_vs_hebap[20];
extern const int16_t ff_eac3_gaq_remap_1[12];
extern const int16_t ff_eac3_gaq_remap_2_4_a[9][2];
extern const int8_t ff_eac3_gaq_remap_2_4_b[9][2];
extern const int16_t (*ff_eac3_vq_hebap[8])[6];
extern const uint8_t ff_eac3_frm_expstr[32][6];
extern const uint8_t ff_eac3_default_cpl_band_struct[18];
extern const uint8_t ff_ac3_rematrix_band_tab[5];
#endif /* FFMPEG_AC3DEC_DATA_H */

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@ -1,186 +1,59 @@
/**
* @file ac3tab.h
* tables taken directly from AC3 spec.
/*
* AC-3 tables
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard.
*
* 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
*/
/* possible frequencies */
const uint16_t ac3_freqs[3] = { 48000, 44100, 32000 };
#ifndef FFMPEG_AC3TAB_H
#define FFMPEG_AC3TAB_H
/* possible bitrates */
const uint16_t ac3_bitratetab[19] = {
32, 40, 48, 56, 64, 80, 96, 112, 128,
160, 192, 224, 256, 320, 384, 448, 512, 576, 640
#include "common.h"
extern const uint16_t ff_ac3_frame_size_tab[38][3];
extern const uint8_t ff_ac3_channels_tab[8];
extern const uint16_t ff_ac3_sample_rate_tab[3];
extern const uint16_t ff_ac3_bitrate_tab[19];
extern const int16_t ff_ac3_window[256];
extern const uint8_t ff_ac3_log_add_tab[260];
extern const uint16_t ff_ac3_hearing_threshold_tab[50][3];
extern const uint8_t ff_ac3_bap_tab[64];
extern const uint8_t ff_ac3_slow_decay_tab[4];
extern const uint8_t ff_ac3_fast_decay_tab[4];
extern const uint16_t ff_ac3_slow_gain_tab[4];
extern const uint16_t ff_ac3_db_per_bit_tab[4];
extern const int16_t ff_ac3_floor_tab[8];
extern const uint16_t ff_ac3_fast_gain_tab[8];
extern const uint8_t ff_ac3_critical_band_size_tab[50];
extern const uint16_t ff_eac3_default_chmap[8];
/** Custom channel map locations bitmask
* Other channels described in documentation:
* Lc/Rc pair, Lrs/Rrs pair, Ts, Lsd/Rsd pair,
* Lw/Rw pair, Lvh/Rvh pair, Cvh, Reserved, LFE2
*/
enum CustomChannelMapLocation{
AC3_CHMAP_L= 1<<(15-0),
AC3_CHMAP_C= 1<<(15-1),
AC3_CHMAP_R= 1<<(15-2),
AC3_CHMAP_L_SUR= 1<<(15-3),
AC3_CHMAP_R_SUR = 1<<(15-4),
AC3_CHMAP_C_SUR= 1<<(15-7),
AC3_CHMAP_LFE = 1<<(15-15)
};
/* AC3 MDCT window */
/* MDCT window */
const int16_t ac3_window[256] = {
4, 7, 12, 16, 21, 28, 34, 42,
51, 61, 72, 84, 97, 111, 127, 145,
164, 184, 207, 231, 257, 285, 315, 347,
382, 419, 458, 500, 544, 591, 641, 694,
750, 810, 872, 937, 1007, 1079, 1155, 1235,
1318, 1406, 1497, 1593, 1692, 1796, 1903, 2016,
2132, 2253, 2379, 2509, 2644, 2783, 2927, 3076,
3230, 3389, 3552, 3721, 3894, 4072, 4255, 4444,
4637, 4835, 5038, 5246, 5459, 5677, 5899, 6127,
6359, 6596, 6837, 7083, 7334, 7589, 7848, 8112,
8380, 8652, 8927, 9207, 9491, 9778,10069,10363,
10660,10960,11264,11570,11879,12190,12504,12820,
13138,13458,13780,14103,14427,14753,15079,15407,
15735,16063,16392,16720,17049,17377,17705,18032,
18358,18683,19007,19330,19651,19970,20287,20602,
20914,21225,21532,21837,22139,22438,22733,23025,
23314,23599,23880,24157,24430,24699,24964,25225,
25481,25732,25979,26221,26459,26691,26919,27142,
27359,27572,27780,27983,28180,28373,28560,28742,
28919,29091,29258,29420,29577,29729,29876,30018,
30155,30288,30415,30538,30657,30771,30880,30985,
31086,31182,31274,31363,31447,31528,31605,31678,
31747,31814,31877,31936,31993,32046,32097,32145,
32190,32232,32272,32310,32345,32378,32409,32438,
32465,32490,32513,32535,32556,32574,32592,32608,
32623,32636,32649,32661,32671,32681,32690,32698,
32705,32712,32718,32724,32729,32733,32737,32741,
32744,32747,32750,32752,32754,32756,32757,32759,
32760,32761,32762,32763,32764,32764,32765,32765,
32766,32766,32766,32766,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
32767,32767,32767,32767,32767,32767,32767,32767,
};
static uint8_t masktab[253];
static const uint8_t latab[260]= {
0x0040,0x003f,0x003e,0x003d,0x003c,0x003b,0x003a,0x0039,0x0038,0x0037,
0x0036,0x0035,0x0034,0x0034,0x0033,0x0032,0x0031,0x0030,0x002f,0x002f,
0x002e,0x002d,0x002c,0x002c,0x002b,0x002a,0x0029,0x0029,0x0028,0x0027,
0x0026,0x0026,0x0025,0x0024,0x0024,0x0023,0x0023,0x0022,0x0021,0x0021,
0x0020,0x0020,0x001f,0x001e,0x001e,0x001d,0x001d,0x001c,0x001c,0x001b,
0x001b,0x001a,0x001a,0x0019,0x0019,0x0018,0x0018,0x0017,0x0017,0x0016,
0x0016,0x0015,0x0015,0x0015,0x0014,0x0014,0x0013,0x0013,0x0013,0x0012,
0x0012,0x0012,0x0011,0x0011,0x0011,0x0010,0x0010,0x0010,0x000f,0x000f,
0x000f,0x000e,0x000e,0x000e,0x000d,0x000d,0x000d,0x000d,0x000c,0x000c,
0x000c,0x000c,0x000b,0x000b,0x000b,0x000b,0x000a,0x000a,0x000a,0x000a,
0x000a,0x0009,0x0009,0x0009,0x0009,0x0009,0x0008,0x0008,0x0008,0x0008,
0x0008,0x0008,0x0007,0x0007,0x0007,0x0007,0x0007,0x0007,0x0006,0x0006,
0x0006,0x0006,0x0006,0x0006,0x0006,0x0006,0x0005,0x0005,0x0005,0x0005,
0x0005,0x0005,0x0005,0x0005,0x0004,0x0004,0x0004,0x0004,0x0004,0x0004,
0x0004,0x0004,0x0004,0x0004,0x0004,0x0003,0x0003,0x0003,0x0003,0x0003,
0x0003,0x0003,0x0003,0x0003,0x0003,0x0003,0x0003,0x0003,0x0003,0x0002,
0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,
0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0002,0x0001,0x0001,
0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,
0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,
0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,0x0001,
0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
};
static const uint16_t hth[50][3]= {
{ 0x04d0,0x04f0,0x0580 },
{ 0x04d0,0x04f0,0x0580 },
{ 0x0440,0x0460,0x04b0 },
{ 0x0400,0x0410,0x0450 },
{ 0x03e0,0x03e0,0x0420 },
{ 0x03c0,0x03d0,0x03f0 },
{ 0x03b0,0x03c0,0x03e0 },
{ 0x03b0,0x03b0,0x03d0 },
{ 0x03a0,0x03b0,0x03c0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03b0 },
{ 0x03a0,0x03a0,0x03a0 },
{ 0x0390,0x03a0,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0390,0x0390,0x03a0 },
{ 0x0380,0x0390,0x03a0 },
{ 0x0380,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0370,0x0380,0x03a0 },
{ 0x0360,0x0370,0x0390 },
{ 0x0360,0x0370,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0350,0x0360,0x0390 },
{ 0x0340,0x0350,0x0380 },
{ 0x0340,0x0350,0x0380 },
{ 0x0330,0x0340,0x0380 },
{ 0x0320,0x0340,0x0370 },
{ 0x0310,0x0320,0x0360 },
{ 0x0300,0x0310,0x0350 },
{ 0x02f0,0x0300,0x0340 },
{ 0x02f0,0x02f0,0x0330 },
{ 0x02f0,0x02f0,0x0320 },
{ 0x02f0,0x02f0,0x0310 },
{ 0x0300,0x02f0,0x0300 },
{ 0x0310,0x0300,0x02f0 },
{ 0x0340,0x0320,0x02f0 },
{ 0x0390,0x0350,0x02f0 },
{ 0x03e0,0x0390,0x0300 },
{ 0x0420,0x03e0,0x0310 },
{ 0x0460,0x0420,0x0330 },
{ 0x0490,0x0450,0x0350 },
{ 0x04a0,0x04a0,0x03c0 },
{ 0x0460,0x0490,0x0410 },
{ 0x0440,0x0460,0x0470 },
{ 0x0440,0x0440,0x04a0 },
{ 0x0520,0x0480,0x0460 },
{ 0x0800,0x0630,0x0440 },
{ 0x0840,0x0840,0x0450 },
{ 0x0840,0x0840,0x04e0 },
};
static const uint8_t baptab[64]= {
0, 1, 1, 1, 1, 1, 2, 2, 3, 3,
3, 4, 4, 5, 5, 6, 6, 6, 6, 7,
7, 7, 7, 8, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 10, 11, 11, 11, 11, 12,
12, 12, 12, 13, 13, 13, 13, 14, 14, 14,
14, 14, 14, 14, 14, 15, 15, 15, 15, 15,
15, 15, 15, 15,
};
const uint8_t sdecaytab[4]={
0x0f, 0x11, 0x13, 0x15,
};
const uint8_t fdecaytab[4]={
0x3f, 0x53, 0x67, 0x7b,
};
const uint16_t sgaintab[4]= {
0x540, 0x4d8, 0x478, 0x410,
};
const uint16_t dbkneetab[4]= {
0x000, 0x700, 0x900, 0xb00,
};
const uint16_t floortab[8]= {
0x2f0, 0x2b0, 0x270, 0x230, 0x1f0, 0x170, 0x0f0, 0xf800,
};
const uint16_t fgaintab[8]= {
0x080, 0x100, 0x180, 0x200, 0x280, 0x300, 0x380, 0x400,
};
static const uint8_t bndsz[50]={
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 3, 3, 3, 3, 3, 3,
3, 6, 6, 6, 6, 6, 6, 12, 12, 12, 12, 24, 24, 24, 24, 24
};
static uint8_t bndtab[51];
/* fft & mdct sin cos tables */
static int16_t costab[64];
static int16_t sintab[64];
static int16_t fft_rev[512];
static int16_t xcos1[128];
static int16_t xsin1[128];
static uint16_t crc_table[256];
#endif /* FFMPEG_AC3TAB_H */

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/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* 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
*/
#ifndef FFMPEG_ACELP_FILTERS_H
#define FFMPEG_ACELP_FILTERS_H
#include <stdint.h>
/**
* low-pass Finite Impulse Response filter coefficients.
*
* Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
* the coefficients are scaled by 2^15.
* This array only contains the right half of the filter.
* This filter is likely identical to the one used in G.729, though this
* could not be determined from the original comments with certainity.
*/
extern const int16_t ff_acelp_interp_filter[61];
/**
* Generic FIR interpolation routine.
* @param out [out] buffer for interpolated data
* @param in input data
* @param filter_coeffs interpolation filter coefficients (0.15)
* @param precision sub sample factor, that is the precision of the position
* @param frac_pos fractional part of position [0..precision-1]
* @param filter_length filter length
* @param length length of output
*
* filter_coeffs contains coefficients of the right half of the symmetric
* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
* See ff_acelp_interp_filter for an example.
*
*/
void ff_acelp_interpolate(
int16_t* out,
const int16_t* in,
const int16_t* filter_coeffs,
int precision,
int frac_pos,
int filter_length,
int length);
/**
* Circularly convolve fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
* @param fc_out vector with filter applied
* @param fc_in source vector
* @param filter phase filter coefficients
*
* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
*
* \note fc_in and fc_out should not overlap!
*/
void ff_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len);
/**
* LP synthesis filter.
* @param out [out] pointer to output buffer
* @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
* @param in input signal
* @param buffer_length amount of data to process
* @param filter_length filter length (10 for 10th order LP filter)
* @param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
* @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
*
* @return 1 if overflow occurred, 0 - otherwise
*
* @note Output buffer must contain 10 samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
int ff_acelp_lp_synthesis_filter(
int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow,
int rounder);
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
* @param out [out] output buffer for filtered speech data
* @param hpf_f [in/out] past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* @param in speech data to process
* @param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
* The filter has a cut-off frequency of 1/80 of the sampling freq
*
* @note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
*
* @remark It is safe to pass the same array in in and out parameters.
*
* @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
void ff_acelp_high_pass_filter(
int16_t* out,
int hpf_f[2],
const int16_t* in,
int length);
#endif /* FFMPEG_ACELP_FILTERS_H */

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/*
* Various fixed-point math operations
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* 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
*/
#ifndef FFMPEG_ACELP_MATH_H
#define FFMPEG_ACELP_MATH_H
#include <stdint.h>
/**
* fixed-point implementation of cosine in [0; PI) domain.
* @param arg fixed-point cosine argument, 0 <= arg < 0x4000
*
* @return value of (1<<15) * cos(arg * PI / (1<<14)), -0x8000 <= result <= 0x7fff
*/
int16_t ff_cos(uint16_t arg);
/**
* fixed-point implementation of exp2(x) in [0; 1] domain.
* @param power argument to exp2, 0 <= power <= 0x7fff
*
* @return value of (1<<20) * exp2(power / (1<<15))
* 0x8000c <= result <= 0xfffea
*/
int ff_exp2(uint16_t power);
/**
* Calculates log2(x).
* @param value function argument, 0 < value <= 7fff ffff
*
* @return value of (1<<15) * log2(value)
*/
int ff_log2(uint32_t value);
/**
* returns the dot product.
* @param a input data array
* @param b input data array
* @param length number of elements
* @param shift right shift by this value will be done after multiplication
*
* @return dot product = sum of elementwise products
*/
static int dot_product(const int16_t* a, const int16_t* b, int length, int shift)
{
int sum = 0;
int i;
for(i=0; i<length; i++)
sum += (a[i] * b[i]) >> shift;
return sum;
}
/**
* Shift value left or right depending on sign of offset parameter.
* @param value value to shift
* @param offset shift offset
*
* @return value << offset, if offset>=0; value >> -offset - otherwise
*/
static inline int bidir_sal(int value, int offset)
{
if(offset < 0) return value >> -offset;
else return value << offset;
}
#endif /* FFMPEG_ACELP_MATH_H */

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/*
* gain code, gain pitch and pitch delay decoding
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* 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
*/
#ifndef FFMPEG_ACELP_PITCH_DELAY_H
#define FFMPEG_ACELP_PITCH_DELAY_H
#include <stdint.h>
#define PITCH_DELAY_MIN 20
#define PITCH_DELAY_MAX 143
/**
* \brief Decode pitch delay of the first subframe encoded by 8 bits with 1/3
* resolution.
* \param ac_index adaptive codebook index (8 bits)
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* with 1/3 resolution, 19 < pitch_delay < 85
* integers only, 85 <= pitch_delay <= 143
*/
int ff_acelp_decode_8bit_to_1st_delay3(int ac_index);
/**
* \brief Decode pitch delay of the second subframe encoded by 5 or 6 bits
* with 1/3 precision.
* \param ac_index adaptive codebook index (5 or 6 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval
* for second subframe
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* with 1/3 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in G.729 @8k, AMR @10.2k, AMR @7.95k,
* AMR @7.4k for the second subframe.
*/
int ff_acelp_decode_5_6_bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min);
/**
* \brief Decode pitch delay with 1/3 precision.
* \param ac_index adaptive codebook index (4 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/3 units
*
* Pitch delay is coded:
* integers only, -6 < pitch_delay - int(prev_pitch_delay) <= -2
* with 1/3 resolution, -2 < pitch_delay - int(prev_pitch_delay) < 1
* integers only, 1 <= pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in G.729 @6.4k, AMR @6.7k, AMR @5.9k,
* AMR @5.15k, AMR @4.75k for the second subframe.
*/
int ff_acelp_decode_4bit_to_2nd_delay3(
int ac_index,
int pitch_delay_min);
/**
* \brief Decode pitch delay of the first subframe encoded by 9 bits
* with 1/6 precision.
* \param ac_index adaptive codebook index (9 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/6 units
*
* Pitch delay is coded:
* with 1/6 resolution, 17 < pitch_delay < 95
* integers only, 95 <= pitch_delay <= 143
*
* \remark The routine is used in AMR @12.2k for the first and third subframes.
*/
int ff_acelp_decode_9bit_to_1st_delay6(int ac_index);
/**
* \brief Decode pitch delay of the second subframe encoded by 6 bits
* with 1/6 precision.
* \param ac_index adaptive codebook index (6 bits)
* \param pitch_delay_min lower bound (integer) of pitch delay interval for
* second subframe
*
* \return pitch delay in 1/6 units
*
* Pitch delay is coded:
* with 1/6 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
*
* \remark The routine is used in AMR @12.2k for the second and fourth subframes.
*/
int ff_acelp_decode_6bit_to_2nd_delay6(
int ac_index,
int pitch_delay_min);
/**
* \brief Update past quantized energies
* \param quant_energy [in/out] past quantized energies (5.10)
* \param gain_corr_factor gain correction factor
* \param log2_ma_pred_order log2() of MA prediction order
* \param erasure frame erasure flag
*
* If frame erasure flag is not equal to zero, memory is updated with
* averaged energy, attenuated by 4dB:
* max(avg(quant_energy[i])-4, -14), i=0,ma_pred_order
*
* In normal mode memory is updated with
* Er - Ep = 20 * log10(gain_corr_factor)
*
* \remark The routine is used in G.729 and AMR (all modes).
*/
void ff_acelp_update_past_gain(
int16_t* quant_energy,
int gain_corr_factor,
int log2_ma_pred_order,
int erasure);
/**
* \brief Decode the adaptive codebook gain and add
* correction (4.1.5 and 3.9.1 of G.729).
* \param gain_corr_factor gain correction factor (2.13)
* \param fc_v fixed-codebook vector (2.13)
* \param mr_energy mean innovation energy and fixed-point correction (7.13)
* \param quant_energy [in/out] past quantized energies (5.10)
* \param subframe_size length of subframe
* \param ma_pred_order MA prediction order
*
* \return quantized fixed-codebook gain (14.1)
*
* The routine implements equations 69, 66 and 71 of the G.729 specification (3.9.1)
*
* Em - mean innovation energy (dB, constant, depends on decoding algorithm)
* Ep - mean-removed predicted energy (dB)
* Er - mean-removed innovation energy (dB)
* Ei - mean energy of the fixed-codebook contribution (dB)
* N - subframe_size
* M - MA (Moving Average) prediction order
* gc - fixed-codebook gain
* gc_p - predicted fixed-codebook gain
*
* Fixed codebook gain is computed using predicted gain gc_p and
* correction factor gain_corr_factor as shown below:
*
* gc = gc_p * gain_corr_factor
*
* The predicted fixed codebook gain gc_p is found by predicting
* the energy of the fixed-codebook contribution from the energy
* of previous fixed-codebook contributions.
*
* mean = 1/N * sum(i,0,N){ fc_v[i] * fc_v[i] }
*
* Ei = 10log(mean)
*
* Er = 10log(1/N * gc^2 * mean) - Em = 20log(gc) + Ei - Em
*
* Replacing Er with Ep and gc with gc_p we will receive:
*
* Ep = 10log(1/N * gc_p^2 * mean) - Em = 20log(gc_p) + Ei - Em
*
* and from above:
*
* gc_p = 10^((Ep - Ei + Em) / 20)
*
* Ep is predicted using past energies and prediction coefficients:
*
* Ep = sum(i,0,M){ ma_prediction_coeff[i] * quant_energy[i] }
*
* gc_p in fixed-point arithmetic is calculated as following:
*
* mean = 1/N * sum(i,0,N){ (fc_v[i] / 2^13) * (fc_v[i] / 2^13) } =
* = 1/N * sum(i,0,N) { fc_v[i] * fc_v[i] } / 2^26
*
* Ei = 10log(mean) = -10log(N) - 10log(2^26) +
* + 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
*
* Ep - Ei + Em = Ep + Em + 10log(N) + 10log(2^26) -
* - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] }) =
* = Ep + mr_energy - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
*
* gc_p = 10 ^ ((Ep - Ei + Em) / 20) =
* = 2 ^ (3.3219 * (Ep - Ei + Em) / 20) = 2 ^ (0.166 * (Ep - Ei + Em))
*
* where
*
* mr_energy = Em + 10log(N) + 10log(2^26)
*
* \remark The routine is used in G.729 and AMR (all modes).
*/
int16_t ff_acelp_decode_gain_code(
int gain_corr_factor,
const int16_t* fc_v,
int mr_energy,
const int16_t* quant_energy,
const int16_t* ma_prediction_coeff,
int subframe_size,
int max_pred_order);
#endif /* FFMPEG_ACELP_PITCH_DELAY_H */

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/*
* adaptive and fixed codebook vector operations for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* 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
*/
#ifndef FFMPEG_ACELP_VECTORS_H
#define FFMPEG_ACELP_VECTORS_H
#include <stdint.h>
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75
* -------------------------------------------------------------------------
* 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76
* -------------------------------------------------------------------------
* 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
* -------------------------------------------------------------------------
*
* Table contains only first the pulse indexes.
*
* Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_tracks_13[16];
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78
* | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
* -------------------------------------------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* Used in G.729 @8k, G.729 @4.4k, AMR @7.95k, AMR @7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_track_4[32];
/**
* Track|Pulse| Positions
* -----------------------------------------
* 1 | 0 | 1, 6, 11, 16, 21, 26, 31, 36
* | | 3, 8, 13, 18, 23, 28, 33, 38
* -----------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* @note (EE) Reference G.729D code also uses gray decoding for each
* pulse index before looking up the value in the table.
*
* Used in G.729 @6.4k (with gray coding), AMR @5.9k (without gray coding)
*/
extern const uint8_t ff_fc_2pulses_9bits_track1[16];
extern const uint8_t ff_fc_2pulses_9bits_track1_gray[16];
/**
* Track|Pulse| Positions
* -----------------------------------------
* 2 | 1 | 0, 7, 14, 20, 27, 34, 1, 21
* | | 2, 9, 15, 22, 29, 35, 6, 26
* | | 4,10, 17, 24, 30, 37, 11, 31
* | | 5,12, 19, 25, 32, 39, 16, 36
* -----------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* @note (EE.1) This table (from the reference code) does not comply with
* the specification.
* The specification contains the following table:
*
* Track|Pulse| Positions
* -----------------------------------------
* 2 | 1 | 0, 5, 10, 15, 20, 25, 30, 35
* | | 1, 6, 11, 16, 21, 26, 31, 36
* | | 2, 7, 12, 17, 22, 27, 32, 37
* | | 4, 9, 14, 19, 24, 29, 34, 39
*
* -----------------------------------------
*
* @note (EE.2) Reference G.729D code also uses gray decoding for each
* pulse index before looking up the value in the table.
*
* Used in G.729 @6.4k (with gray coding)
*/
extern const uint8_t ff_fc_2pulses_9bits_track2_gray[32];
/**
* Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
* @param fc_v [out] decoded fixed codebook vector (2.13)
* @param tab1 table used for first pulse_count pulses
* @param tab2 table used for last pulse
* @param pulse_indexes fixed codebook indexes
* @param pulse_signs signs of the excitation pulses (0 bit value
* means negative sign)
* @param bits number of bits per one pulse index
* @param pulse_count number of pulses decoded using first table
* @param bits length of one pulse index in bits
*
* Used in G.729 @8k, G.729 @4.4k, G.729 @6.4k, AMR @7.95k, AMR @7.40k
*/
void ff_acelp_fc_pulse_per_track(
int16_t* fc_v,
const uint8_t *tab1,
const uint8_t *tab2,
int pulse_indexes,
int pulse_signs,
int pulse_count,
int bits);
/**
* weighted sum of two vectors with rounding.
* @param out [out] result of addition
* @param in_a first vector
* @param in_b second vector
* @param weight_coeff_a first vector weight coefficient
* @param weight_coeff_a second vector weight coefficient
* @param rounder this value will be added to the sum of the two vectors
* @param shift result will be shifted to right by this value
* @param length vectors length
*
* @note It is safe to pass the same buffer for out and in_a or in_b.
*
* out[i] = (in_a[i]*weight_a + in_b[i]*weight_b + rounder) >> shift
*/
void ff_acelp_weighted_vector_sum(
int16_t* out,
const int16_t *in_a,
const int16_t *in_b,
int16_t weight_coeff_a,
int16_t weight_coeff_b,
int16_t rounder,
int shift,
int length);
#endif /* FFMPEG_ACELP_VECTORS_H */

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/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* 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 adx.h
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
#ifndef FFMPEG_ADX_H
#define FFMPEG_ADX_H
typedef struct {
int s1,s2;
} PREV;
typedef struct {
PREV prev[2];
int header_parsed;
unsigned char dec_temp[18*2];
int in_temp;
} ADXContext;
#define BASEVOL 0x4000
#define SCALE1 0x7298
#define SCALE2 0x3350
#endif /* FFMPEG_ADX_H */

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/*
* Atrac 3 compatible decoder data
* Copyright (c) 2006-2007 Maxim Poliakovski
* Copyright (c) 2006-2007 Benjamin Larsson
*
* 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 atrac3data.h
* Atrac 3 AKA RealAudio 8 compatible decoder data
*/
#ifndef FFMPEG_ATRAC3DATA_H
#define FFMPEG_ATRAC3DATA_H
#include <stdint.h>
/* VLC tables */
static const uint8_t huffcode1[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
};
static const uint8_t huffbits1[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffcode2[5] = {
0x0,0x4,0x5,0x6,0x7,
};
static const uint8_t huffbits2[5] = {
1,3,3,3,3,
};
static const uint8_t huffcode3[7] = {
0x0,0x4,0x5,0xC,0xD,0xE,0xF,
};
static const uint8_t huffbits3[7] = {
1,3,3,4,4,4,4,
};
static const uint8_t huffcode4[9] = {
0x0,0x4,0x5,0xC,0xD,0x1C,0x1D,0x1E,0x1F,
};
static const uint8_t huffbits4[9] = {
1,3,3,4,4,5,5,5,5,
};
static const uint8_t huffcode5[15] = {
0x0,0x2,0x3,0x8,0x9,0xA,0xB,0x1C,0x1D,0x3C,0x3D,0x3E,0x3F,0xC,0xD,
};
static const uint8_t huffbits5[15] = {
2,3,3,4,4,4,4,5,5,6,6,6,6,4,4
};
static const uint8_t huffcode6[31] = {
0x0,0x2,0x3,0x4,0x5,0x6,0x7,0x14,0x15,0x16,0x17,0x18,0x19,0x34,0x35,
0x36,0x37,0x38,0x39,0x3A,0x3B,0x78,0x79,0x7A,0x7B,0x7C,0x7D,0x7E,0x7F,0x8,0x9,
};
static const uint8_t huffbits6[31] = {
3,4,4,4,4,4,4,5,5,5,5,5,5,6,6,6,6,6,6,6,6,7,7,7,7,7,7,7,7,4,4
};
static const uint8_t huffcode7[63] = {
0x0,0x8,0x9,0xA,0xB,0xC,0xD,0xE,0xF,0x10,0x11,0x24,0x25,0x26,0x27,0x28,
0x29,0x2A,0x2B,0x2C,0x2D,0x2E,0x2F,0x30,0x31,0x32,0x33,0x68,0x69,0x6A,0x6B,0x6C,
0x6D,0x6E,0x6F,0x70,0x71,0x72,0x73,0x74,0x75,0xEC,0xED,0xEE,0xEF,0xF0,0xF1,0xF2,
0xF3,0xF4,0xF5,0xF6,0xF7,0xF8,0xF9,0xFA,0xFB,0xFC,0xFD,0xFE,0xFF,0x2,0x3,
};
static const uint8_t huffbits7[63] = {
3,5,5,5,5,5,5,5,5,5,5,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,7,7,7,7,7,
7,7,7,7,7,7,7,7,7,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,8,4,4
};
static const uint8_t huff_tab_sizes[7] = {
9, 5, 7, 9, 15, 31, 63,
};
static const uint8_t* const huff_codes[7] = {
huffcode1,huffcode2,huffcode3,huffcode4,huffcode5,huffcode6,huffcode7,
};
static const uint8_t* const huff_bits[7] = {
huffbits1,huffbits2,huffbits3,huffbits4,huffbits5,huffbits6,huffbits7,
};
/* selector tables */
static const uint8_t CLCLengthTab[8] = {0, 4, 3, 3, 4, 4, 5, 6};
static const int8_t seTab_0[4] = {0, 1, -2, -1};
static const int8_t decTable1[18] = {0,0, 0,1, 0,-1, 1,0, -1,0, 1,1, 1,-1, -1,1, -1,-1};
/* tables for the scalefactor decoding */
static const float iMaxQuant[8] = {
0.0, 1.0/1.5, 1.0/2.5, 1.0/3.5, 1.0/4.5, 1.0/7.5, 1.0/15.5, 1.0/31.5
};
static const uint16_t subbandTab[33] = {
0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224,
256, 288, 320, 352, 384, 416, 448, 480, 512, 576, 640, 704, 768, 896, 1024
};
/* transform data */
static const float qmf_48tap_half[24] = {
-0.00001461907, -0.00009205479, -0.000056157569, 0.00030117269,
0.0002422519,-0.00085293897, -0.0005205574, 0.0020340169,
0.00078333891, -0.0042153862, -0.00075614988, 0.0078402944,
-0.000061169922, -0.01344162, 0.0024626821, 0.021736089,
-0.007801671, -0.034090221, 0.01880949, 0.054326009,
-0.043596379, -0.099384367, 0.13207909, 0.46424159
};
/* joint stereo related tables */
static const float matrixCoeffs[8] = {0.0, 2.0, 2.0, 2.0, 0.0, 0.0, 1.0, 1.0};
#endif /* FFMPEG_ATRAC3DATA_H */

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/*
* audio conversion
* Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2008 Peter Ross
*
* 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
*/
#ifndef FFMPEG_AUDIOCONVERT_H
#define FFMPEG_AUDIOCONVERT_H
/**
* @file audioconvert.h
* Audio format conversion routines
*/
#include "avcodec.h"
/**
* Generate string corresponding to the sample format with
* number sample_fmt, or a header if sample_fmt is negative.
*
* @param[in] buf the buffer where to write the string
* @param[in] buf_size the size of buf
* @param[in] sample_fmt the number of the sample format to print the corresponding info string, or
* a negative value to print the corresponding header.
* Meaningful values for obtaining a sample format info vary from 0 to SAMPLE_FMT_NB -1.
*/
void avcodec_sample_fmt_string(char *buf, int buf_size, int sample_fmt);
/**
* @return NULL on error
*/
const char *avcodec_get_sample_fmt_name(int sample_fmt);
/**
* @return SAMPLE_FMT_NONE on error
*/
enum SampleFormat avcodec_get_sample_fmt(const char* name);
struct AVAudioConvert;
typedef struct AVAudioConvert AVAudioConvert;
/**
* Create an audio sample format converter context
* @param out_fmt Output sample format
* @param out_channels Number of output channels
* @param in_fmt Input sample format
* @param in_channels Number of input channels
* @param[in] matrix Channel mixing matrix (of dimension in_channel*out_channels). Set to NULL to ignore.
* @param flags See FF_MM_xx
* @return NULL on error
*/
AVAudioConvert *av_audio_convert_alloc(enum SampleFormat out_fmt, int out_channels,
enum SampleFormat in_fmt, int in_channels,
const float *matrix, int flags);
/**
* Free audio sample format converter context
*/
void av_audio_convert_free(AVAudioConvert *ctx);
/**
* Convert between audio sample formats
* @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel.
* @param[in] out_stride distance between consecutive input samples (measured in bytes)
* @param[in] in array of input buffers for each channel
* @param[in] in_stride distance between consecutive output samples (measured in bytes)
* @param len length of audio frame size (measured in samples)
*/
int av_audio_convert(AVAudioConvert *ctx,
void * const out[6], const int out_stride[6],
const void * const in[6], const int in_stride[6], int len);
#endif /* FFMPEG_AUDIOCONVERT_H */

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