/* * 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_LAME_H #define LAME_LAME_H #include #if defined(__cplusplus) extern "C" { #endif /* maximum size of mp3buffer needed if you encode at most 1152 samples for each call to lame_encode_buffer. see lame_encode_buffer() below (LAME_MAXMP3BUFFER is now obsolete) */ #define LAME_MAXMP3BUFFER 16384 typedef enum sound_file_format_e { sf_unknown, sf_raw, sf_wave, sf_aiff, sf_mp1, /* MPEG Layer 1, aka mpg */ sf_mp2, /* MPEG Layer 2 */ sf_mp3, /* MPEG Layer 3 */ sf_ogg } sound_file_format; typedef enum vbr_mode_e { vbr_off=0, vbr_mt, vbr_rh, vbr_abr, vbr_mtrh, vbr_default=vbr_rh /* change this to change the default VBR mode of LAME */ } vbr_mode; struct id3tag_spec { /* private data members */ int flags; const char *title; const char *artist; const char *album; int year; const char *comment; int track; int genre; }; #ifndef BRHIST_BARMAX # define BRHIST_BARMAX 50 #endif typedef struct BRHST_TAG { long count[15]; long count_max; int vbrmin; int vbrmax; char kbps[15][4]; char bar[BRHIST_BARMAX + 1]; } BRHST, *PBRHST; /*********************************************************************** * * Control Parameters set by User * * substantiated by calling program * * Initilized and default values set by lame_init(&gf) * * ***********************************************************************/ 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) */ float scale; /* scale input by this amount */ /* general control params */ int gtkflag; /* run frame analyzer? */ int bWriteVbrTag; /* add Xing VBR tag? */ int disable_waveheader; /* disable writing of .wav header, when *decoding* */ int decode_only; /* use lame/mpglib to convert mp3 to wav */ int ogg; /* encode to Vorbis .ogg file */ int quality; /* quality setting 0=best, 9=worst */ int silent; /* disable some status output */ float update_interval; /* to use Frank's time status display */ int brhist_disp; /* enable VBR bitrate histogram display */ 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 */ float compression_ratio; /* user specified compression ratio, instead of brate */ int free_format; /* use free format? */ /* 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 */ int strict_ISO; /* enforce ISO spec as much as possible */ /* quantization/noise shaping */ int disable_reservoir; /* use bit reservoir? */ int experimentalX; int experimentalY; int experimentalZ; int exp_nspsytune; /* VBR control */ vbr_mode VBR; int VBR_q; int VBR_mean_bitrate_kbps; int VBR_min_bitrate_kbps; int VBR_max_bitrate_kbps; int VBR_hard_min; /* strictly enforce VBR_min_bitrate*/ /* normaly, it will be violated for analog silence */ /* 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%)*/ /* I/O - not used if calling program does the i/o */ sound_file_format input_format; FILE * musicin; /* file pointer to input file */ int swapbytes; /* force byte swapping default=0*/ #define MAX_NAME_SIZE 1000 char inPath[MAX_NAME_SIZE]; /* Note: outPath must be set if you want Xing VBR or ID3 version 1 tags written */ char outPath[MAX_NAME_SIZE]; /* optional ID3 tags */ int id3v1_enabled; struct id3tag_spec tag_spec; /* psycho acoustics and other aguments which you should not change * unless you know what you are doing */ int ATHonly; /* only use ATH */ int ATHshort; /* only use ATH for short blocks */ int noATH; /* disable ATH */ int ATHlower; /* lower ATH by this many db */ int cwlimit; /* predictability limit */ int allow_diff_short; /* allow blocktypes to differ between channels ? */ int no_short_blocks; /* disable short blocks */ int emphasis; /* obsolete */ float raiseSMR; /* 0...1 : 0 leave SMR, 1 maximize SMR */ /************************************************************************/ /* internal variables, do not set... */ /* provided because they may be of use to some applications */ /************************************************************************/ BRHST brhist; /* histogramming data */ int version; /* 0=MPEG2 1=MPEG1 */ int frameNum; /* frame counter */ int totalframes; /* frames: 0..totalframes-1 (estimate)*/ int encoder_delay; int framesize; /* VBR tags */ int nZeroStreamSize; int TotalFrameSize; int* pVbrFrames; int nVbrNumFrames; int nVbrFrameBufferSize; /************************************************************************/ /* more internal variables, which will not exist after lame_encode_finish() */ /************************************************************************/ void *internal_flags; } lame_global_flags; /* The LAME API */ /* REQUIRED: initialize the encoder. sets default for all encoder paramters, * returns -1 if some malloc()'s failed * otherwise returns 0 * */ int 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 *, const char *); /* OPTIONAL: call this to print a command line interface usage guide and quit */ void lame_help(lame_global_flags *, const 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. returns -1 if something failed. */ int lame_init_params(lame_global_flags *); /* OPTIONAL: 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 * -1: mp3buffer was too small * -2: malloc() problem * -3: lame_init_params() not called * -4: psycho acoustic problems * -5: ogg cleanup encoding error * -6: ogg frame encoding error * * 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); /* REQUIRED: lame_encode_finish will flush the buffers and may return a * final few mp3 frames. mp3buffer should be at least 7200 bytes. * * will also write id3 tags (if any) into the bitstream * * 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_fid will append a Xing VBR tag to the mp3 file with file pointer fid. These calls perform forward and backwards seeks, so make sure fid is a real file. Note: if VBR 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_fid(lame_global_flags *,FILE* fid); /********************************************************************* * 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 *********************************************************************/ typedef struct { int header_parsed; /* 1 if header was parsed and following data was computed: */ int stereo; /* number of channels */ int samplerate; /* sample rate */ int bitrate; /* bitrate */ int mode; /* mp3 frame type */ int mode_ext; /* mp3 frame type */ int framesize; /* number of samples per mp3 frame */ /* this data is not currently computed by the mpglib routines: */ unsigned long nsamp; /* number of samples in mp3 file. */ int framenum; /* frames decoded counter */ int totalframes; /* total number of frames in mp3 file */ } mp3data_struct; /* required call to initialize decoder */ int lame_decode_init(void); /********************************************************************* * input 1 mp3 frame, output (maybe) pcm data. * lame_decode return code: -1: error. 0: need more data. n>0: size of pcm output *********************************************************************/ int lame_decode(char *mp3buf,int len,short pcm_l[],short pcm_r[]); /* same as lame_decode, but returns mp3 header data */ int lame_decode_headers(char *mp3buf,int len,short pcm_l[],short pcm_r[], mp3data_struct *mp3data); /* same as lame_decode, but returns at most one frame */ int lame_decode1(char *mp3buf,int len,short pcm_l[],short pcm_r[]); /* same as lame_decode1, but returns at most one frame and mp3 header data */ int lame_decode1_headers(char *mp3buf,int len,short pcm_l[],short pcm_r[], mp3data_struct *mp3data); #if defined(__cplusplus) } #endif #endif