Mercurial > libavcodec.hg
view mpegaudio.h @ 12415:04b85031bf00 libavcodec
added/extended some comments in a64multi_encode_frame()
author | bindhammer |
---|---|
date | Wed, 25 Aug 2010 11:47:47 +0000 |
parents | b4888704c11e |
children |
line wrap: on
line source
/* * copyright (c) 2001 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 */ /** * @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 "fft.h" #define CONFIG_AUDIO_NONSHORT 0 /* max frame size, in samples */ #define MPA_FRAME_SIZE 1152 /* max compressed frame size */ #define MPA_MAX_CODED_FRAME_SIZE 1792 #define MPA_MAX_CHANNELS 2 #define SBLIMIT 32 /* number of subbands */ #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 #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 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 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 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 */