mirror of https://git.ffmpeg.org/ffmpeg.git
211 lines
6.0 KiB
C
211 lines
6.0 KiB
C
/*
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* copyright (c) 2001 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* mpeg audio declarations for both encoder and decoder.
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*/
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#ifndef AVCODEC_MPEGAUDIO_H
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#define AVCODEC_MPEGAUDIO_H
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#ifndef CONFIG_FLOAT
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# define CONFIG_FLOAT 0
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#endif
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "fft.h"
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#define CONFIG_AUDIO_NONSHORT 0
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/* max frame size, in samples */
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#define MPA_FRAME_SIZE 1152
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/* max compressed frame size */
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#define MPA_MAX_CODED_FRAME_SIZE 1792
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#define MPA_MAX_CHANNELS 2
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#define SBLIMIT 32 /* number of subbands */
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#define MPA_STEREO 0
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#define MPA_JSTEREO 1
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#define MPA_DUAL 2
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#define MPA_MONO 3
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/* header + layer + bitrate + freq + lsf/mpeg25 */
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#define SAME_HEADER_MASK \
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(0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
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#define MP3_MASK 0xFFFE0CCF
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#if CONFIG_MPEGAUDIO_HP
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#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */
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#define WFRAC_BITS 16 /* fractional bits for window */
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#else
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#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
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#define WFRAC_BITS 14 /* fractional bits for window */
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#endif
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#define FRAC_ONE (1 << FRAC_BITS)
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#define FIX(a) ((int)((a) * FRAC_ONE))
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#if CONFIG_FLOAT
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typedef float OUT_INT;
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#define OUT_FMT AV_SAMPLE_FMT_FLT
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#elif CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT
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typedef int32_t OUT_INT;
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#define OUT_MAX INT32_MAX
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#define OUT_MIN INT32_MIN
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#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31)
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#define OUT_FMT AV_SAMPLE_FMT_S32
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#else
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typedef int16_t OUT_INT;
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#define OUT_MAX INT16_MAX
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#define OUT_MIN INT16_MIN
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#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
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#define OUT_FMT AV_SAMPLE_FMT_S16
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#endif
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#if CONFIG_FLOAT
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# define INTFLOAT float
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typedef float MPA_INT;
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#elif FRAC_BITS <= 15
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# define INTFLOAT int
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typedef int16_t MPA_INT;
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#else
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# define INTFLOAT int
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typedef int32_t MPA_INT;
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#endif
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#define BACKSTEP_SIZE 512
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#define EXTRABYTES 24
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/* layer 3 "granule" */
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typedef struct GranuleDef {
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uint8_t scfsi;
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int part2_3_length;
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int big_values;
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int global_gain;
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int scalefac_compress;
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uint8_t block_type;
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uint8_t switch_point;
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int table_select[3];
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int subblock_gain[3];
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uint8_t scalefac_scale;
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uint8_t count1table_select;
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int region_size[3]; /* number of huffman codes in each region */
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int preflag;
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int short_start, long_end; /* long/short band indexes */
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uint8_t scale_factors[40];
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INTFLOAT sb_hybrid[SBLIMIT * 18]; /* 576 samples */
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} GranuleDef;
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#define MPA_DECODE_HEADER \
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int frame_size; \
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int error_protection; \
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int layer; \
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int sample_rate; \
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int sample_rate_index; /* between 0 and 8 */ \
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int bit_rate; \
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int nb_channels; \
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int mode; \
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int mode_ext; \
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int lsf;
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typedef struct MPADecodeHeader {
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MPA_DECODE_HEADER
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} MPADecodeHeader;
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typedef struct MPADecodeContext {
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MPA_DECODE_HEADER
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uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES];
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int last_buf_size;
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/* next header (used in free format parsing) */
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uint32_t free_format_next_header;
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GetBitContext gb;
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GetBitContext in_gb;
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DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
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int synth_buf_offset[MPA_MAX_CHANNELS];
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DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
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INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
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GranuleDef granules[2][2]; /* Used in Layer 3 */
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#ifdef DEBUG
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int frame_count;
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#endif
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int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
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int dither_state;
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int error_recognition;
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AVCodecContext* avctx;
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#if CONFIG_FLOAT
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DCTContext dct;
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#endif
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void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window,
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int *dither_state, OUT_INT *samples, int incr);
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} MPADecodeContext;
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/* layer 3 huffman tables */
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typedef struct HuffTable {
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int xsize;
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const uint8_t *bits;
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const uint16_t *codes;
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} HuffTable;
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int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
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int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
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extern MPA_INT ff_mpa_synth_window[];
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void ff_mpa_synth_init(MPA_INT *window);
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void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
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MPA_INT *window, int *dither_state,
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OUT_INT *samples, int incr,
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INTFLOAT sb_samples[SBLIMIT]);
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void ff_mpa_synth_init_float(MPA_INT *window);
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void ff_mpa_synth_filter_float(MPADecodeContext *s,
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MPA_INT *synth_buf_ptr, int *synth_buf_offset,
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MPA_INT *window, int *dither_state,
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OUT_INT *samples, int incr,
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INTFLOAT sb_samples[SBLIMIT]);
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void ff_mpegaudiodec_init_mmx(MPADecodeContext *s);
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void ff_mpegaudiodec_init_altivec(MPADecodeContext *s);
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/* fast header check for resync */
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static inline int ff_mpa_check_header(uint32_t header){
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/* header */
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if ((header & 0xffe00000) != 0xffe00000)
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return -1;
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/* layer check */
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if ((header & (3<<17)) == 0)
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return -1;
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/* bit rate */
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if ((header & (0xf<<12)) == 0xf<<12)
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return -1;
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/* frequency */
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if ((header & (3<<10)) == 3<<10)
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return -1;
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return 0;
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}
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#endif /* AVCODEC_MPEGAUDIO_H */
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