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FFmpeg/libavcodec/mlp.h
Diego Biurrun dc8a7c93d3 Add missing void keyword to parameterless function declarations.
Originally committed as revision 16860 to svn://svn.ffmpeg.org/ffmpeg/trunk
2009-01-30 18:38:25 +00:00

120 lines
4.1 KiB
C

/*
* MLP codec common header file
* Copyright (c) 2007-2008 Ian Caulfield
*
* 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 AVCODEC_MLP_H
#define AVCODEC_MLP_H
#include <stdint.h>
#include "avcodec.h"
/** Maximum number of channels that can be decoded. */
#define MAX_CHANNELS 16
/** Maximum number of matrices used in decoding; most streams have one matrix
* per output channel, but some rematrix a channel (usually 0) more than once.
*/
#define MAX_MATRICES 15
/** Maximum number of substreams that can be decoded. This could also be set
* higher, but I haven't seen any examples with more than two.
*/
#define MAX_SUBSTREAMS 2
/** maximum sample frequency seen in files */
#define MAX_SAMPLERATE 192000
/** maximum number of audio samples within one access unit */
#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
/** next power of two greater than MAX_BLOCKSIZE */
#define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000))
/** number of allowed filters */
#define NUM_FILTERS 2
/** The maximum number of taps in either the IIR or FIR filter;
* I believe MLP actually specifies the maximum order for IIR filters as four,
* and that the sum of the orders of both filters must be <= 8.
*/
#define MAX_FILTER_ORDER 8
/** Code that signals end of a stream. */
#define END_OF_STREAM 0xd234d234
#define FIR 0
#define IIR 1
/** filter data */
typedef struct {
uint8_t order; ///< number of taps in filter
uint8_t shift; ///< Right shift to apply to output of filter.
int32_t coeff[MAX_FILTER_ORDER];
int32_t state[MAX_FILTER_ORDER];
} FilterParams;
/** sample data coding information */
typedef struct {
FilterParams filter_params[NUM_FILTERS];
int16_t huff_offset; ///< Offset to apply to residual values.
int32_t sign_huff_offset; ///< sign/rounding-corrected version of huff_offset
uint8_t codebook; ///< Which VLC codebook to use to read residuals.
uint8_t huff_lsbs; ///< Size of residual suffix not encoded using VLC.
} ChannelParams;
/** Tables defining the Huffman codes.
* There are three entropy coding methods used in MLP (four if you count
* "none" as a method). These use the same sequences for codes starting with
* 00 or 01, but have different codes starting with 1.
*/
extern const uint8_t ff_mlp_huffman_tables[3][18][2];
/** MLP uses checksums that seem to be based on the standard CRC algorithm, but
* are not (in implementation terms, the table lookup and XOR are reversed).
* We can implement this behavior using a standard av_crc on all but the
* last element, then XOR that with the last element.
*/
uint8_t ff_mlp_checksum8 (const uint8_t *buf, unsigned int buf_size);
uint16_t ff_mlp_checksum16(const uint8_t *buf, unsigned int buf_size);
/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
* number of bits, starting two bits into the first byte of buf.
*/
uint8_t ff_mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size);
/** XOR together all the bytes of a buffer.
* Does this belong in dspcontext?
*/
uint8_t ff_mlp_calculate_parity(const uint8_t *buf, unsigned int buf_size);
void ff_mlp_init_crc(void);
/** XOR four bytes into one. */
static inline uint8_t xor_32_to_8(uint32_t value)
{
value ^= value >> 16;
value ^= value >> 8;
return value;
}
#endif /* AVCODEC_MLP_H */