X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudio.h;h=f12b897e238b8884fe2fe23ec625450b53c85981;hb=5705b02079449c685a3dd337fcc3a8b440dca4a0;hp=9d5fdfb0114823a11573a577771f6935ed2d97d0;hpb=1a56543279a6fd146c8616781b4160e207bb4f6d;p=ffmpeg diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 9d5fdfb0114..f12b897e238 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -1,34 +1,199 @@ +/* + * 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 + * 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" + +/* max frame size, in samples */ +#define MPA_FRAME_SIZE 1152 /* max compressed frame size */ -#define MPA_MAX_CODED_FRAME_SIZE 1200 +#define MPA_MAX_CODED_FRAME_SIZE 1792 -#define MPA_FRAME_SIZE 1152 #define MPA_MAX_CHANNELS 2 -#define SAMPLES_BUF_SIZE 4096 #define SBLIMIT 32 /* number of subbands */ -#define DCT_BITS 14 /* number of bits for the DCT */ -#define MUL(a,b) (((a) * (b)) >> DCT_BITS) -#define FIX(a) ((int)((a) * (1 << DCT_BITS))) - -typedef struct MpegAudioContext { - PutBitContext pb; - int nb_channels; - int freq, bit_rate; - int lsf; /* 1 if mpeg2 low bitrate selected */ - int bitrate_index; /* bit rate */ - int freq_index; - int frame_size; /* frame size, in bits, without padding */ - INT64 nb_samples; /* total number of samples encoded */ - /* padding computation */ - int frame_frac, frame_frac_incr, do_padding; - short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ - int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ - int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; - unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ - /* code to group 3 scale factors */ - unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; - int sblimit; /* number of used subbands */ - const unsigned char *alloc_table; -} MpegAudioContext; +#define MPA_STEREO 0 +#define MPA_JSTEREO 1 +#define MPA_DUAL 2 +#define MPA_MONO 3 + +/* header + layer + bitrate + freq + lsf/mpeg25 */ +#define SAME_HEADER_MASK \ + (0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19)) + +#define MP3_MASK 0xFFFE0CCF + +#ifndef FRAC_BITS +#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */ +#define WFRAC_BITS 16 /* 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 +#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 + +#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){ + /* header */ + if ((header & 0xffe00000) != 0xffe00000) + return -1; + /* layer check */ + if ((header & (3<<17)) == 0) + return -1; + /* bit rate */ + if ((header & (0xf<<12)) == 0xf<<12) + return -1; + /* frequency */ + if ((header & (3<<10)) == 3<<10) + return -1; + return 0; +} + +#endif /* AVCODEC_MPEGAUDIO_H */