| /* |
| * Interface to MP3 LAME encoding engine |
| * |
| * Copyright (c) 1999 Mark Taylor |
| * |
| * This program is free software; you can redistribute it and/or modify |
| * it under the terms of the GNU General Public License as published by |
| * the Free Software Foundation; either version 2, or (at your option) |
| * any later version. |
| * |
| * This program 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 General Public License for more details. |
| * |
| * You should have received a copy of the GNU General Public License |
| * along with this program; see the file COPYING. If not, write to |
| * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. |
| */ |
| #ifndef LAME_H_INCLUDE |
| #define LAME_H_INCLUDE |
| #include <stdio.h> |
| |
| /* maximum size of mp3buffer needed if you encode at most 1152 samples for |
| each call to lame_encode_buffer. see lame_encode_buffer() below */ |
| #define LAME_MAXMP3BUFFER 16384 |
| |
| |
| typedef enum sound_file_format_e { |
| sf_unknown, sf_wave, sf_aiff, sf_mp3, sf_raw |
| } sound_file_format; |
| |
| |
| /*********************************************************************** |
| * |
| * Global Variables. |
| * |
| * substantiated in lame.c |
| * |
| * Initilized and default values set by gf=lame_init() |
| * gf is a pointer to this struct, which the user may use to |
| * override any of the default values |
| * |
| * a call to lame_set_params() is also needed |
| * |
| ***********************************************************************/ |
| typedef struct { |
| /* input file description */ |
| unsigned long num_samples; /* number of samples. default=2^32-1 */ |
| int num_channels; /* input number of channels. default=2 */ |
| int in_samplerate; /* input_samp_rate. default=44.1kHz */ |
| int out_samplerate; /* output_samp_rate. (usually determined automatically) */ |
| |
| |
| /* general control params */ |
| int gtkflag; /* run frame analyzer? */ |
| int bWriteVbrTag; /* add Xing VBR tag? */ |
| int quality; /* quality setting 0=best, 9=worst */ |
| int silent; /* disable some status output */ |
| int mode; /* 0,1,2,3 stereo,jstereo,dual channel,mono */ |
| int mode_fixed; /* use specified the mode, do not use lame's opinion of the best mode */ |
| int force_ms; /* force M/S mode. requires mode=1 */ |
| int brate; /* bitrate */ |
| |
| /* frame params */ |
| int copyright; /* mark as copyright. default=0 */ |
| int original; /* mark as original. default=1 */ |
| int error_protection; /* use 2 bytes per frame for a CRC checksum. default=0*/ |
| int padding_type; /* 0=no padding, 1=always pad, 2=adjust padding */ |
| int extension; /* the MP3 'private extension' bit. meaningless */ |
| |
| /* quantization/noise shaping */ |
| int disable_reservoir; /* use bit reservoir? */ |
| int experimentalX; |
| int experimentalY; |
| int experimentalZ; |
| |
| /* VBR control */ |
| int VBR; |
| int VBR_q; |
| int VBR_min_bitrate_kbps; |
| int VBR_max_bitrate_kbps; |
| |
| |
| /* resampling and filtering */ |
| int lowpassfreq; /* freq in Hz. 0=lame choses. -1=no filter */ |
| int highpassfreq; /* freq in Hz. 0=lame choses. -1=no filter */ |
| int lowpasswidth; /* freq width of filter, in Hz (default=15%)*/ |
| int highpasswidth; /* freq width of filter, in Hz (default=15%)*/ |
| |
| |
| /* input file reading - not used if calling program does the i/o */ |
| sound_file_format input_format; |
| int swapbytes; /* force byte swapping default=0*/ |
| char *inPath; /* name of input file */ |
| char *outPath; /* name of output file. */ |
| /* Note: outPath must be set if you want Xing VBR or id3 tags |
| * written */ |
| |
| |
| /* psycho acoustics and other aguments which you should not change |
| * unless you know what you are doing */ |
| int ATHonly; /* only use ATH */ |
| int noATH; /* disable ATH */ |
| float cwlimit; /* predictability limit */ |
| int allow_diff_short; /* allow blocktypes to differ between channels ? */ |
| int no_short_blocks; /* disable short blocks */ |
| int emphasis; /* obsolete */ |
| |
| |
| |
| /********************************************************************/ |
| /* internal variables NOT set by calling program, and should not be */ |
| /* modified by the calling program */ |
| /********************************************************************/ |
| long int frameNum; /* frame counter */ |
| long totalframes; /* frames: 0..totalframes-1 (estimate)*/ |
| int encoder_delay; |
| int framesize; |
| int version; /* 0=MPEG2 1=MPEG1 */ |
| int padding; /* padding for the current frame? */ |
| int mode_gr; /* granules per frame */ |
| int stereo; /* number of channels */ |
| int VBR_min_bitrate; /* min bitrate index */ |
| int VBR_max_bitrate; /* max bitrate index */ |
| float resample_ratio; /* input_samp_rate/output_samp_rate */ |
| int bitrate_index; |
| int samplerate_index; |
| int mode_ext; |
| |
| /* lowpass and highpass filter control */ |
| float lowpass1,lowpass2; /* normalized frequency bounds of passband */ |
| float highpass1,highpass2; /* normalized frequency bounds of passband */ |
| |
| /* polyphase filter (filter_type=0) */ |
| int lowpass_band; /* zero bands >= lowpass_band in the polyphase filterbank */ |
| int highpass_band; /* zero bands <= highpass_band */ |
| |
| |
| |
| int filter_type; /* 0=polyphase filter, 1= FIR filter 2=MDCT filter(bad)*/ |
| int quantization; /* 0 = ISO formual, 1=best amplitude */ |
| int noise_shaping; /* 0 = none |
| 1 = ISO AAC model |
| 2 = allow scalefac_select=1 |
| */ |
| |
| int noise_shaping_stop; /* 0 = stop at over=0, all scalefacs amplified or |
| a scalefac has reached max value |
| 1 = stop when all scalefacs amplified or |
| a scalefac has reached max value |
| 2 = stop when all scalefacs amplified |
| */ |
| |
| int psymodel; /* 0 = none 1=gpsycho */ |
| int use_best_huffman; /* 0 = no. 1=outside loop 2=inside loop(slow) */ |
| |
| |
| } lame_global_flags; |
| |
| |
| |
| |
| |
| |
| /* |
| |
| The LAME API |
| |
| */ |
| |
| |
| /* REQUIRED: initialize the encoder. sets default for all encoder paramters, |
| * returns pointer to encoder parameters listed above |
| */ |
| void lame_init(lame_global_flags *); |
| |
| |
| |
| |
| /********************************************************************* |
| * command line argument parsing & option setting. Only supported |
| * if libmp3lame compiled with LAMEPARSE defined |
| *********************************************************************/ |
| /* OPTIONAL: call this to print an error with a brief command line usage guide and quit |
| * only supported if libmp3lame compiled with LAMEPARSE defined. |
| */ |
| void lame_usage(lame_global_flags *, char *); |
| |
| /* OPTIONAL: call this to print a command line interface usage guide and quit */ |
| void lame_help(lame_global_flags *, char *); |
| |
| /* OPTIONAL: get the version number, in a string. of the form: "3.63 (beta)" or |
| just "3.63". Max allows length is 20 characters */ |
| void lame_version(lame_global_flags *, char *); |
| |
| |
| /* OPTIONAL: set internal options via command line argument parsing |
| * You can skip this call if you like the default values, or if |
| * set the encoder parameters your self |
| */ |
| void lame_parse_args(lame_global_flags *, int argc, char **argv); |
| |
| |
| |
| |
| |
| /* REQUIRED: sets more internal configuration based on data provided |
| * above |
| */ |
| void lame_init_params(lame_global_flags *); |
| |
| |
| /* OPTONAL: print internal lame configuration on stderr*/ |
| void lame_print_config(lame_global_flags *); |
| |
| |
| |
| |
| /* input pcm data, output (maybe) mp3 frames. |
| * This routine handles all buffering, resampling and filtering for you. |
| * |
| * leftpcm[] array of 16bit pcm data, left channel |
| * rightpcm[] array of 16bit pcm data, right channel |
| * num_samples number of samples in leftpcm[] and rightpcm[] (if stereo) |
| * mp3buffer pointer to buffer where mp3 output is written |
| * mp3buffer_size size of mp3buffer, in bytes |
| * return code number of bytes output in mp3buffer. can be 0 |
| * if return code = -1: mp3buffer was too small |
| * |
| * The required mp3buffer_size can be computed from num_samples, |
| * samplerate and encoding rate, but here is a worst case estimate: |
| * |
| * mp3buffer_size in bytes = 1.25*num_samples + 7200 |
| * |
| * I think a tighter bound could be: (mt, March 2000) |
| * MPEG1: |
| * num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512 |
| * MPEG2: |
| * num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256 |
| * |
| * but test first if you use that! |
| * |
| * set mp3buffer_size = 0 and LAME will not check if mp3buffer_size is |
| * large enough. |
| * |
| * NOTE: if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels |
| * will be averaged into the L channel before encoding only the L channel |
| * This will overwrite the data in leftpcm[] and rightpcm[]. |
| * |
| */ |
| int lame_encode_buffer(lame_global_flags *,short int leftpcm[], short int rightpcm[],int num_samples, |
| char *mp3buffer,int mp3buffer_size); |
| |
| /* as above, but input has L & R channel data interleaved. Note: |
| * num_samples = number of samples in the L (or R) |
| * channel, not the total number of samples in pcm[] |
| */ |
| int lame_encode_buffer_interleaved(lame_global_flags *,short int pcm[], |
| int num_samples, char *mp3buffer,int mp3buffer_size); |
| |
| |
| |
| /* input 1 pcm frame, output (maybe) 1 mp3 frame. |
| * return code = number of bytes output in mp3buffer. can be 0 |
| * NOTE: this interface is outdated, please use lame_encode_buffer() instead |
| * declair mp3buffer with: char mp3buffer[LAME_MAXMP3BUFFER] |
| * if return code = -1: mp3buffer was too small |
| */ |
| int lame_encode(lame_global_flags *,short int Buffer[2][1152],char *mp3buffer,int mp3buffer_size); |
| |
| |
| |
| /* REQUIRED: lame_encode_finish will flush the buffers and may return a |
| * final few mp3 frames. mp3buffer should be at least 7200 bytes. |
| * |
| * return code = number of bytes output to mp3buffer. can be 0 |
| */ |
| int lame_encode_finish(lame_global_flags *,char *mp3buffer, int size); |
| |
| |
| /* OPTIONAL: lame_mp3_tags will append id3 and Xing VBR tags to |
| the mp3 file with name given by gf->outPath. These calls open the file, |
| write tags, and close the file, so make sure the the encoding is finished |
| before calling these routines. |
| Note: if VBR and id3 tags are turned off by the user, or turned off |
| by LAME because the output is not a regular file, this call does nothing |
| */ |
| void lame_mp3_tags(lame_global_flags *); |
| |
| |
| |
| |
| /********************************************************************* |
| * lame file i/o. Only supported |
| * if libmp3lame compiled with LAMESNDFILE or LIBSNDFILE |
| *********************************************************************/ |
| /* OPTIONAL: open the input file, and parse headers if possible |
| * you can skip this call if you will do your own PCM input |
| */ |
| void lame_init_infile(lame_global_flags *); |
| |
| /* OPTIONAL: read one frame of PCM data from audio input file opened by |
| * lame_init_infile. Input file can be wav, aiff, raw pcm, anything |
| * supported by libsndfile, or an mp3 file |
| */ |
| int lame_readframe(lame_global_flags *,short int Buffer[2][1152]); |
| |
| /* OPTIONAL: close the sound input file if lame_init_infile() was used */ |
| void lame_close_infile(lame_global_flags *); |
| |
| |
| |
| |
| |
| /********************************************************************* |
| * a simple interface to mpglib, part of mpg123, is also included if |
| * libmp3lame is compiled with HAVEMPGLIB |
| * input 1 mp3 frame, output (maybe) 1 pcm frame. |
| * lame_decode return code: -1: error. 0: need more data. n>0: size of pcm output |
| *********************************************************************/ |
| int lame_decode_init(void); |
| int lame_decode(char *mp3buf,int len,short pcm_l[],short pcm_r[]); |
| /* read mp3 file until mpglib returns one frame of PCM data */ |
| #ifdef AMIGA_MPEGA |
| int lame_decode_initfile(const char *fullname,int *stereo,int *samp,int *bitrate, unsigned long *nsamp); |
| int lame_decode_fromfile(FILE *fd,short int pcm_l[], short int pcm_r[]); |
| #else |
| int lame_decode_initfile(FILE *fd,int *stereo,int *samp,int *bitrate, unsigned long *nsamp); |
| int lame_decode_fromfile(FILE *fd,short int pcm_l[],short int pcm_r[]); |
| #endif |
| |
| |
| |
| |
| #endif |
