X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudio.h;h=fbfddcc5d25058d7542af94e2b76c347b6229006;hb=88312a4de3708cdd8f0ca4121546ec882777b7fa;hp=072c41bda7fbee6d7b0613b0e2f6d26a5e81e1e5;hpb=a7a858996f8dc5807c696c344b347a0c314417ae;p=ffmpeg diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 072c41bda7f..fbfddcc5d25 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -1,10 +1,44 @@ +/* + * copyright (c) 2001 Fabrice Bellard + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + /** - * @file mpegaudio.h + * @file * mpeg audio declarations for both encoder and decoder. */ +#ifndef AVCODEC_MPEGAUDIO_H +#define AVCODEC_MPEGAUDIO_H + +#ifndef CONFIG_FLOAT +# define CONFIG_FLOAT 0 +#endif + +#include "avcodec.h" +#include "get_bits.h" +#include "dsputil.h" +#include "dct.h" + +#define CONFIG_AUDIO_NONSHORT 0 + /* max frame size, in samples */ -#define MPA_FRAME_SIZE 1152 +#define MPA_FRAME_SIZE 1152 /* max compressed frame size */ #define MPA_MAX_CODED_FRAME_SIZE 1792 @@ -22,17 +56,139 @@ #define SAME_HEADER_MASK \ (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19)) -int l2_select_table(int bitrate, int nb_channels, int freq, int lsf); -int mpa_decode_header(AVCodecContext *avctx, uint32_t head); +#define MP3_MASK 0xFFFE0CCF + +#if CONFIG_MPEGAUDIO_HP +#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */ +#define WFRAC_BITS 16 /* fractional bits for window */ +#else +#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ +#define WFRAC_BITS 14 /* fractional bits for window */ +#endif + +#define FRAC_ONE (1 << FRAC_BITS) + +#define FIX(a) ((int)((a) * FRAC_ONE)) + +#if CONFIG_FLOAT +typedef float OUT_INT; +#define OUT_FMT AV_SAMPLE_FMT_FLT +#elif CONFIG_MPEGAUDIO_HP && CONFIG_AUDIO_NONSHORT +typedef int32_t OUT_INT; +#define OUT_MAX INT32_MAX +#define OUT_MIN INT32_MIN +#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 31) +#define OUT_FMT AV_SAMPLE_FMT_S32 +#else +typedef int16_t OUT_INT; +#define OUT_MAX INT16_MAX +#define OUT_MIN INT16_MIN +#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15) +#define OUT_FMT AV_SAMPLE_FMT_S16 +#endif -extern const uint16_t mpa_bitrate_tab[2][3][15]; -extern const uint16_t mpa_freq_tab[3]; -extern const unsigned char *alloc_tables[5]; -extern const double enwindow[512]; -extern const int sblimit_table[5]; -extern const int quant_steps[17]; -extern const int quant_bits[17]; -extern const int32_t mpa_enwindow[257]; +#if CONFIG_FLOAT +# define INTFLOAT float +typedef float MPA_INT; +#elif FRAC_BITS <= 15 +# define INTFLOAT int +typedef int16_t MPA_INT; +#else +# define INTFLOAT int +typedef int32_t MPA_INT; +#endif + +#define BACKSTEP_SIZE 512 +#define EXTRABYTES 24 + +/* layer 3 "granule" */ +typedef struct GranuleDef { + uint8_t scfsi; + int part2_3_length; + int big_values; + int global_gain; + int scalefac_compress; + uint8_t block_type; + uint8_t switch_point; + int table_select[3]; + int subblock_gain[3]; + uint8_t scalefac_scale; + uint8_t count1table_select; + int region_size[3]; /* number of huffman codes in each region */ + int preflag; + int short_start, long_end; /* long/short band indexes */ + uint8_t scale_factors[40]; + INTFLOAT sb_hybrid[SBLIMIT * 18]; /* 576 samples */ +} GranuleDef; + +#define MPA_DECODE_HEADER \ + int frame_size; \ + int error_protection; \ + int layer; \ + int sample_rate; \ + int sample_rate_index; /* between 0 and 8 */ \ + int bit_rate; \ + int nb_channels; \ + int mode; \ + int mode_ext; \ + int lsf; + +typedef struct MPADecodeHeader { + MPA_DECODE_HEADER +} MPADecodeHeader; + +typedef struct MPADecodeContext { + MPA_DECODE_HEADER + uint8_t last_buf[2*BACKSTEP_SIZE + EXTRABYTES]; + int last_buf_size; + /* next header (used in free format parsing) */ + uint32_t free_format_next_header; + GetBitContext gb; + GetBitContext in_gb; + DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2]; + int synth_buf_offset[MPA_MAX_CHANNELS]; + DECLARE_ALIGNED(16, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT]; + INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */ + GranuleDef granules[2][2]; /* Used in Layer 3 */ +#ifdef DEBUG + int frame_count; +#endif + int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3 + int dither_state; + int error_recognition; + AVCodecContext* avctx; +#if CONFIG_FLOAT + DCTContext dct; +#endif + void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window, + int *dither_state, OUT_INT *samples, int incr); +} MPADecodeContext; + +/* layer 3 huffman tables */ +typedef struct HuffTable { + int xsize; + const uint8_t *bits; + const uint16_t *codes; +} HuffTable; + +int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf); +int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate); +extern MPA_INT ff_mpa_synth_window[]; +void ff_mpa_synth_init(MPA_INT *window); +void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, + MPA_INT *window, int *dither_state, + OUT_INT *samples, int incr, + INTFLOAT sb_samples[SBLIMIT]); + +void ff_mpa_synth_init_float(MPA_INT *window); +void ff_mpa_synth_filter_float(MPADecodeContext *s, + MPA_INT *synth_buf_ptr, int *synth_buf_offset, + MPA_INT *window, int *dither_state, + OUT_INT *samples, int incr, + INTFLOAT sb_samples[SBLIMIT]); + +void ff_mpegaudiodec_init_mmx(MPADecodeContext *s); +void ff_mpegaudiodec_init_altivec(MPADecodeContext *s); /* fast header check for resync */ static inline int ff_mpa_check_header(uint32_t header){ @@ -50,3 +206,5 @@ static inline int ff_mpa_check_header(uint32_t header){ return -1; return 0; } + +#endif /* AVCODEC_MPEGAUDIO_H */