blob: 20a94f85d7eb50dfacf2cbc5d6e6c7c6c5759b20 [file] [log] [blame]
* 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
* 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.
#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;
/* 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 */
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[]);
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[]);