/* * 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 #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 */ int num_channels; /* input number of channels */ int samplerate; /* input_samp_rate */ /* general control params */ int gtkflag; /* frame analyzer? */ int bWriteVbrTag; /* Xing VBR tag? */ int quality; /* quality setting 0=best, 9=worst */ int silent; /* disable some status output */ int mode; /* mono, stereo, jstereo */ int mode_fixed; /* use specified mode, not lame's opinion of the best mode */ int force_ms; /* force M/S mode */ int brate; /* bitrate */ /* frame params */ int copyright; /* mark as copyright */ int original; /* mark as original */ int error_protection; /* use 2 bytes per frame for a CRC checksum */ int padding; /* 0=no padding, 1=always pad, 2=adjust padding */ /* 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 resamplerate; /* output_samp_rate. */ 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 */ int sfb21; /* soon to be obsolete */ /*******************************************************************/ /* internal variables NOT set by calling program, but which might */ /* be of interest to the calling program */ /*******************************************************************/ long int frameNum; /* frame counter */ long totalframes; /* frames: 0..totalframes-1 (estimate)*/ int encoder_delay; int framesize; 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 */ float lowpass1,lowpass2; /* internal filter settings */ float highpass1,highpass2; int filter_type; /* 0=MDCT filter, 1= (expensive) filters */ 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 ms_masking; /* 0 = no, 1 = use mid/side maskings */ 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 */ lame_global_flags *lame_init(void); /* OPTIONAL: call this to print an error with a brief command line usage guide and quit */ void lame_usage(char *); /* OPTIONAL: call this to print a command line interface usage guide and quit */ void lame_help(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(int argc, char **argv); /* 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(void); /* REQUIRED: sets more internal configuration based on data provided * above */ void lame_init_params(); /* OPTONAL: print internal lame configuration on stderr*/ void lame_print_config(); /* 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(short int Buffer[2][1152]); /* input 1 pcm frame, output (maybe) 1 mp3 frame. */ /* return code = number of bytes output in mp3buffer. can be 0 */ int lame_encode(short int Buffer[2][1152],char *mp3buffer); /* here's a better interface (not yet written). It will do the 1152 frame * buffering for you, as well as downsampling. User is required to * provide the mp3buffer size (is there any good way around this?) * * return code = number of bytes output in mp3buffer. can be 0 */ int lame_encode_buffer(short int leftpcm[], short int rightpcm[],int num_samples, char *mp3buffer,int mp3buffer_size); /* REQUIRED: lame_cleanup will flush the buffers and may return a *final mp3 frame */ int lame_encode_finish(char *mp3buf); /* OPTIONAL: close the sound input file if lame_init_infile() was used */ void lame_close_infile(void); /* OPTIONAL: lame_mp3_tags will append id3 and Xing VBR tags to the mp3 file with name given by gf->outPath. These cals 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(void); /* a simple interface to mpglib, part of mpg123, is also included: * 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[][1152]); #endif