From e53ea570b020fd3837347d78b8c7f970ea1abe27 Mon Sep 17 00:00:00 2001 From: Andrew Kelley Date: Wed, 15 Jul 2009 04:56:41 -0700 Subject: [PATCH] AudioFileMp3: provide support for mp3 export Add MP3 as one of the options to export the project as. Currently the lib path is hard-coded, need to add that to settings. It loads lame dynamically. --- TODO | 6 + include/audio_file_mp3.h | 182 +++++ include/lame.h | 1212 +++++++++++++++++++++++++++++ include/project_renderer.h | 3 + src/core/audio/audio_file_mp3.cpp | 312 ++++++++ src/core/main.cpp | 11 +- src/core/project_renderer.cpp | 6 +- 7 files changed, 1726 insertions(+), 6 deletions(-) create mode 100644 include/audio_file_mp3.h create mode 100644 include/lame.h create mode 100644 src/core/audio/audio_file_mp3.cpp diff --git a/TODO b/TODO index 27fa2d617..c88ef1954 100644 --- a/TODO +++ b/TODO @@ -52,6 +52,12 @@ - add FLAC as export-format? Andrew Kelley's todo: +* when you press down or up while browsing the resource browser, it should + play the sample +* select the output file format based on what you selected in file->save as +* segfault when you export an mp3 and press cancel + + - automation recording: * when you record and there is already an auto clip, it repeats it * it freezes when you try to do it with the Volume or Panning slider diff --git a/include/audio_file_mp3.h b/include/audio_file_mp3.h new file mode 100644 index 000000000..33b3c880b --- /dev/null +++ b/include/audio_file_mp3.h @@ -0,0 +1,182 @@ +/* + * audio_file_mp3.h - Audio-device which encodes mp3-stream and writes it + * into an mp3-file. This is used for song-export. + * + * Copyright (c) 2004-2008 Tobias Doerffel + * 2009 Andrew Kelley + * + * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net + * + * 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 of the License, 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 COPYING); if not, write to the + * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, + * Boston, MA 02110-1301 USA. + * + */ + + +#ifndef _AUDIO_FILE_MP3_H +#define _AUDIO_FILE_MP3_H + +#include +#include + +#include "lmmsconfig.h" +#include "audio_file_device.h" + +#include "lame.h" + +#include + + +class AudioFileMp3 : public audioFileDevice +{ +public: + AudioFileMp3( const sample_rate_t _sample_rate, + const ch_cnt_t _channels, + bool & _success_ful, + const QString & _file, + const bool _use_vbr, + const bitrate_t _nom_bitrate, + const bitrate_t _min_bitrate, + const bitrate_t _max_bitrate, + const int _depth, + mixer * _mixer ); + virtual ~AudioFileMp3(); + + static audioFileDevice * getInst( const sample_rate_t _sample_rate, + const ch_cnt_t _channels, + bool & _success_ful, + const QString & _file, + const bool _use_vbr, + const bitrate_t _nom_bitrate, + const bitrate_t _min_bitrate, + const bitrate_t _max_bitrate, + const int _depth, + mixer * _mixer ) + { + return( new AudioFileMp3( _sample_rate, _channels, + _success_ful, _file, _use_vbr, + _nom_bitrate, _min_bitrate, + _max_bitrate, _depth, + _mixer ) ); + } + + +private: + // functions we'll be importing from lame + typedef lame_global_flags *lame_init_t(void); + typedef int lame_init_params_t(lame_global_flags*); + typedef const char* get_lame_version_t(void); + + typedef int lame_encode_buffer_t ( + lame_global_flags* gf, + const short int buffer_l [], + const short int buffer_r [], + const int nsamples, + unsigned char * mp3buf, + const int mp3buf_size ); + + typedef int lame_encode_buffer_interleaved_t( + lame_global_flags* gf, + short int pcm[], + int num_samples, /* per channel */ + unsigned char* mp3buf, + int mp3buf_size ); + + typedef int lame_encode_flush_t( + lame_global_flags *gf, + unsigned char* mp3buf, + int size ); + + typedef int lame_close_t(lame_global_flags*); + + typedef int lame_set_in_samplerate_t(lame_global_flags*, int); + typedef int lame_set_out_samplerate_t(lame_global_flags*, int); + typedef int lame_set_num_channels_t(lame_global_flags*, int ); + typedef int lame_set_quality_t(lame_global_flags*, int); + typedef int lame_set_brate_t(lame_global_flags*, int); + typedef int lame_set_VBR_t(lame_global_flags *, vbr_mode); + typedef int lame_set_VBR_q_t(lame_global_flags *, int); + typedef int lame_set_VBR_min_bitrate_kbps_t(lame_global_flags *, int); + typedef int lame_set_mode_t(lame_global_flags *, MPEG_mode); + typedef int lame_set_preset_t(lame_global_flags *, int); + typedef int lame_set_error_protection_t(lame_global_flags *, int); + typedef int lame_set_disable_reservoir_t(lame_global_flags *, int); + typedef int lame_set_padding_type_t(lame_global_flags *, Padding_type); + typedef int lame_set_bWriteVbrTag_t(lame_global_flags *, int); + typedef size_t lame_get_lametag_frame_t(const lame_global_flags *, unsigned char* buffer, size_t size); + typedef void lame_mp3_tags_fid_t(lame_global_flags *, FILE *); + + typedef int lame_set_findReplayGain_t(lame_global_flags *, int); + typedef int lame_set_VBR_quality_t(lame_global_flags *, float); + typedef int lame_set_VBR_mean_bitrate_kbps_t(lame_global_flags *, int); + typedef int lame_set_VBR_max_bitrate_kbps_t(lame_global_flags *, int); + + + + bool initLame(QString libpath); + short int rescale(float sample); // convert float flame to short int frame + + // overloaded functions + virtual void writeBuffer( const surroundSampleFrame * _ab, + const fpp_t _frames, + float _master_gain ); + + bool startEncoding( void ); + void finishEncoding( void ); + + bool m_ok; // true if we need to close the handle in finishEncoding + + + // handle to lame + lame_global_flags *m_lgf; + QLibrary * m_lame; // lame .so file + QFile * m_outfile; + bool m_hq_mode; // true if we want really high quality + + + /* function pointers to the symbols we get from the library */ + lame_init_t* lame_init; + lame_init_params_t* lame_init_params; + lame_encode_buffer_t* lame_encode_buffer; + lame_encode_buffer_interleaved_t* lame_encode_buffer_interleaved; + lame_encode_flush_t* lame_encode_flush; + lame_close_t* lame_close; + get_lame_version_t* get_lame_version; + + lame_set_in_samplerate_t* lame_set_in_samplerate; + lame_set_out_samplerate_t* lame_set_out_samplerate; + lame_set_num_channels_t* lame_set_num_channels; + lame_set_quality_t* lame_set_quality; + lame_set_brate_t* lame_set_brate; + lame_set_VBR_t* lame_set_VBR; + lame_set_VBR_q_t* lame_set_VBR_q; + lame_set_VBR_min_bitrate_kbps_t* lame_set_VBR_min_bitrate_kbps; + lame_set_mode_t* lame_set_mode; + lame_set_preset_t* lame_set_preset; + lame_set_error_protection_t* lame_set_error_protection; + lame_set_disable_reservoir_t *lame_set_disable_reservoir; + lame_set_padding_type_t *lame_set_padding_type; + lame_set_bWriteVbrTag_t *lame_set_bWriteVbrTag; + lame_get_lametag_frame_t *lame_get_lametag_frame; + lame_mp3_tags_fid_t *lame_mp3_tags_fid; + lame_set_findReplayGain_t *lame_set_findReplayGain; + lame_set_VBR_quality_t *lame_set_VBR_quality; + lame_set_VBR_mean_bitrate_kbps_t *lame_set_VBR_mean_bitrate_kbps; + lame_set_VBR_max_bitrate_kbps_t *lame_set_VBR_max_bitrate_kbps; +} ; + + +#endif + diff --git a/include/lame.h b/include/lame.h new file mode 100644 index 000000000..9863d1aab --- /dev/null +++ b/include/lame.h @@ -0,0 +1,1212 @@ +/* + * Interface to MP3 LAME encoding engine + * + * Copyright (c) 1999 Mark Taylor + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* $Id: lame.h,v 1.170.2.1 2008/09/14 11:51:49 robert Exp $ */ + +#ifndef LAME_LAME_H +#define LAME_LAME_H + +/* for size_t typedef */ +#include +/* for va_list typedef */ +#include +/* for FILE typedef, TODO: remove when removing lame_mp3_tags_fid */ +#include + +#if defined(__cplusplus) +extern "C" { +#endif + +#if defined(WIN32) +#undef CDECL +#define CDECL _cdecl +#else +#define CDECL +#endif + +#define DEPRECATED_OR_OBSOLETE_CODE_REMOVED 1 + +typedef enum vbr_mode_e { + vbr_off=0, + vbr_mt, /* obsolete, same as vbr_mtrh */ + vbr_rh, + vbr_abr, + vbr_mtrh, + vbr_max_indicator, /* Don't use this! It's used for sanity checks. */ + vbr_default=vbr_mtrh /* change this to change the default VBR mode of LAME */ +} vbr_mode; + + +/* MPEG modes */ +typedef enum MPEG_mode_e { + STEREO = 0, + JOINT_STEREO, + DUAL_CHANNEL, /* LAME doesn't supports this! */ + MONO, + NOT_SET, + MAX_INDICATOR /* Don't use this! It's used for sanity checks. */ +} MPEG_mode; + +/* Padding types */ +typedef enum Padding_type_e { + PAD_NO = 0, + PAD_ALL, + PAD_ADJUST, + PAD_MAX_INDICATOR /* Don't use this! It's used for sanity checks. */ +} Padding_type; + + + +/*presets*/ +typedef enum preset_mode_e { + /*values from 8 to 320 should be reserved for abr bitrates*/ + /*for abr I'd suggest to directly use the targeted bitrate as a value*/ + ABR_8 = 8, + ABR_320 = 320, + + V9 = 410, /*Vx to match Lame and VBR_xx to match FhG*/ + VBR_10 = 410, + V8 = 420, + VBR_20 = 420, + V7 = 430, + VBR_30 = 430, + V6 = 440, + VBR_40 = 440, + V5 = 450, + VBR_50 = 450, + V4 = 460, + VBR_60 = 460, + V3 = 470, + VBR_70 = 470, + V2 = 480, + VBR_80 = 480, + V1 = 490, + VBR_90 = 490, + V0 = 500, + VBR_100 = 500, + + + + /*still there for compatibility*/ + R3MIX = 1000, + STANDARD = 1001, + EXTREME = 1002, + INSANE = 1003, + STANDARD_FAST = 1004, + EXTREME_FAST = 1005, + MEDIUM = 1006, + MEDIUM_FAST = 1007 +} preset_mode; + + +/*asm optimizations*/ +typedef enum asm_optimizations_e { + MMX = 1, + AMD_3DNOW = 2, + SSE = 3 +} asm_optimizations; + + +/* psychoacoustic model */ +typedef enum Psy_model_e { + PSY_GPSYCHO = 1, + PSY_NSPSYTUNE = 2 +} Psy_model; + + +struct lame_global_struct; +typedef struct lame_global_struct lame_global_flags; +typedef lame_global_flags *lame_t; + + + + +/*********************************************************************** + * + * The LAME API + * These functions should be called, in this order, for each + * MP3 file to be encoded. See the file "API" for more documentation + * + ***********************************************************************/ + + +/* + * REQUIRED: + * initialize the encoder. sets default for all encoder parameters, + * returns NULL if some malloc()'s failed + * otherwise returns pointer to structure needed for all future + * API calls. + */ +lame_global_flags * CDECL lame_init(void); +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* obsolete version */ +int CDECL lame_init_old(lame_global_flags *); +#endif + +/* + * OPTIONAL: + * set as needed to override defaults + */ + +/******************************************************************** + * input stream description + ***********************************************************************/ +/* number of samples. default = 2^32-1 */ +int CDECL lame_set_num_samples(lame_global_flags *, unsigned long); +unsigned long CDECL lame_get_num_samples(const lame_global_flags *); + +/* input sample rate in Hz. default = 44100hz */ +int CDECL lame_set_in_samplerate(lame_global_flags *, int); +int CDECL lame_get_in_samplerate(const lame_global_flags *); + +/* number of channels in input stream. default=2 */ +int CDECL lame_set_num_channels(lame_global_flags *, int); +int CDECL lame_get_num_channels(const lame_global_flags *); + +/* + scale the input by this amount before encoding. default=0 (disabled) + (not used by decoding routines) +*/ +int CDECL lame_set_scale(lame_global_flags *, float); +float CDECL lame_get_scale(const lame_global_flags *); + +/* + scale the channel 0 (left) input by this amount before encoding. + default=0 (disabled) + (not used by decoding routines) +*/ +int CDECL lame_set_scale_left(lame_global_flags *, float); +float CDECL lame_get_scale_left(const lame_global_flags *); + +/* + scale the channel 1 (right) input by this amount before encoding. + default=0 (disabled) + (not used by decoding routines) +*/ +int CDECL lame_set_scale_right(lame_global_flags *, float); +float CDECL lame_get_scale_right(const lame_global_flags *); + +/* + output sample rate in Hz. default = 0, which means LAME picks best value + based on the amount of compression. MPEG only allows: + MPEG1 32, 44.1, 48khz + MPEG2 16, 22.05, 24 + MPEG2.5 8, 11.025, 12 + (not used by decoding routines) +*/ +int CDECL lame_set_out_samplerate(lame_global_flags *, int); +int CDECL lame_get_out_samplerate(const lame_global_flags *); + + +/******************************************************************** + * general control parameters + ***********************************************************************/ +/* 1=cause LAME to collect data for an MP3 frame analyzer. default=0 */ +int CDECL lame_set_analysis(lame_global_flags *, int); +int CDECL lame_get_analysis(const lame_global_flags *); + +/* + 1 = write a Xing VBR header frame. + default = 1 + this variable must have been added by a Hungarian notation Windows programmer :-) +*/ +int CDECL lame_set_bWriteVbrTag(lame_global_flags *, int); +int CDECL lame_get_bWriteVbrTag(const lame_global_flags *); + +/* 1=decode only. use lame/mpglib to convert mp3/ogg to wav. default=0 */ +int CDECL lame_set_decode_only(lame_global_flags *, int); +int CDECL lame_get_decode_only(const lame_global_flags *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* 1=encode a Vorbis .ogg file. default=0 */ +/* DEPRECATED */ +int CDECL lame_set_ogg(lame_global_flags *, int); +int CDECL lame_get_ogg(const lame_global_flags *); +#endif + +/* + internal algorithm selection. True quality is determined by the bitrate + but this variable will effect quality by selecting expensive or cheap algorithms. + quality=0..9. 0=best (very slow). 9=worst. + recommended: 2 near-best quality, not too slow + 5 good quality, fast + 7 ok quality, really fast +*/ +int CDECL lame_set_quality(lame_global_flags *, int); +int CDECL lame_get_quality(const lame_global_flags *); + +/* + mode = 0,1,2,3 = stereo, jstereo, dual channel (not supported), mono + default: lame picks based on compression ration and input channels +*/ +int CDECL lame_set_mode(lame_global_flags *, MPEG_mode); +MPEG_mode CDECL lame_get_mode(const lame_global_flags *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* + mode_automs. Use a M/S mode with a switching threshold based on + compression ratio + DEPRECATED +*/ +int CDECL lame_set_mode_automs(lame_global_flags *, int); +int CDECL lame_get_mode_automs(const lame_global_flags *); +#endif + +/* + force_ms. Force M/S for all frames. For testing only. + default = 0 (disabled) +*/ +int CDECL lame_set_force_ms(lame_global_flags *, int); +int CDECL lame_get_force_ms(const lame_global_flags *); + +/* use free_format? default = 0 (disabled) */ +int CDECL lame_set_free_format(lame_global_flags *, int); +int CDECL lame_get_free_format(const lame_global_flags *); + +/* perform ReplayGain analysis? default = 0 (disabled) */ +int CDECL lame_set_findReplayGain(lame_global_flags *, int); +int CDECL lame_get_findReplayGain(const lame_global_flags *); + +/* decode on the fly. Search for the peak sample. If the ReplayGain + * analysis is enabled then perform the analysis on the decoded data + * stream. default = 0 (disabled) + * NOTE: if this option is set the build-in decoder should not be used */ +int CDECL lame_set_decode_on_the_fly(lame_global_flags *, int); +int CDECL lame_get_decode_on_the_fly(const lame_global_flags *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* DEPRECATED: now does the same as lame_set_findReplayGain() + default = 0 (disabled) */ +int CDECL lame_set_ReplayGain_input(lame_global_flags *, int); +int CDECL lame_get_ReplayGain_input(const lame_global_flags *); + +/* DEPRECATED: now does the same as + lame_set_decode_on_the_fly() && lame_set_findReplayGain() + default = 0 (disabled) */ +int CDECL lame_set_ReplayGain_decode(lame_global_flags *, int); +int CDECL lame_get_ReplayGain_decode(const lame_global_flags *); + +/* DEPRECATED: now does the same as lame_set_decode_on_the_fly() + default = 0 (disabled) */ +int CDECL lame_set_findPeakSample(lame_global_flags *, int); +int CDECL lame_get_findPeakSample(const lame_global_flags *); +#endif + +/* counters for gapless encoding */ +int CDECL lame_set_nogap_total(lame_global_flags*, int); +int CDECL lame_get_nogap_total(const lame_global_flags*); + +int CDECL lame_set_nogap_currentindex(lame_global_flags* , int); +int CDECL lame_get_nogap_currentindex(const lame_global_flags*); + + +/* + * OPTIONAL: + * Set printf like error/debug/message reporting functions. + * The second argument has to be a pointer to a function which looks like + * void my_debugf(const char *format, va_list ap) + * { + * (void) vfprintf(stdout, format, ap); + * } + * If you use NULL as the value of the pointer in the set function, the + * lame buildin function will be used (prints to stderr). + * To quiet any output you have to replace the body of the example function + * with just "return;" and use it in the set function. + */ +int CDECL lame_set_errorf(lame_global_flags *, + void (*func)(const char *, va_list)); +int CDECL lame_set_debugf(lame_global_flags *, + void (*func)(const char *, va_list)); +int CDECL lame_set_msgf (lame_global_flags *, + void (*func)(const char *, va_list)); + + + +/* set one of brate compression ratio. default is compression ratio of 11. */ +int CDECL lame_set_brate(lame_global_flags *, int); +int CDECL lame_get_brate(const lame_global_flags *); +int CDECL lame_set_compression_ratio(lame_global_flags *, float); +float CDECL lame_get_compression_ratio(const lame_global_flags *); + + +int CDECL lame_set_preset( lame_global_flags* gfp, int ); +int CDECL lame_set_asm_optimizations( lame_global_flags* gfp, int, int ); + + + +/******************************************************************** + * frame params + ***********************************************************************/ +/* mark as copyright. default=0 */ +int CDECL lame_set_copyright(lame_global_flags *, int); +int CDECL lame_get_copyright(const lame_global_flags *); + +/* mark as original. default=1 */ +int CDECL lame_set_original(lame_global_flags *, int); +int CDECL lame_get_original(const lame_global_flags *); + +/* error_protection. Use 2 bytes from each frame for CRC checksum. default=0 */ +int CDECL lame_set_error_protection(lame_global_flags *, int); +int CDECL lame_get_error_protection(const lame_global_flags *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* padding_type. 0=pad no frames 1=pad all frames 2=adjust padding(default) */ +int CDECL lame_set_padding_type(lame_global_flags *, Padding_type); +Padding_type CDECL lame_get_padding_type(const lame_global_flags *); +#endif + +/* MP3 'private extension' bit Meaningless. default=0 */ +int CDECL lame_set_extension(lame_global_flags *, int); +int CDECL lame_get_extension(const lame_global_flags *); + +/* enforce strict ISO compliance. default=0 */ +int CDECL lame_set_strict_ISO(lame_global_flags *, int); +int CDECL lame_get_strict_ISO(const lame_global_flags *); + + +/******************************************************************** + * quantization/noise shaping + ***********************************************************************/ + +/* disable the bit reservoir. For testing only. default=0 */ +int CDECL lame_set_disable_reservoir(lame_global_flags *, int); +int CDECL lame_get_disable_reservoir(const lame_global_flags *); + +/* select a different "best quantization" function. default=0 */ +int CDECL lame_set_quant_comp(lame_global_flags *, int); +int CDECL lame_get_quant_comp(const lame_global_flags *); +int CDECL lame_set_quant_comp_short(lame_global_flags *, int); +int CDECL lame_get_quant_comp_short(const lame_global_flags *); + +int CDECL lame_set_experimentalX(lame_global_flags *, int); /* compatibility*/ +int CDECL lame_get_experimentalX(const lame_global_flags *); + +/* another experimental option. for testing only */ +int CDECL lame_set_experimentalY(lame_global_flags *, int); +int CDECL lame_get_experimentalY(const lame_global_flags *); + +/* another experimental option. for testing only */ +int CDECL lame_set_experimentalZ(lame_global_flags *, int); +int CDECL lame_get_experimentalZ(const lame_global_flags *); + +/* Naoki's psycho acoustic model. default=0 */ +int CDECL lame_set_exp_nspsytune(lame_global_flags *, int); +int CDECL lame_get_exp_nspsytune(const lame_global_flags *); + +void CDECL lame_set_msfix(lame_global_flags *, double); +float CDECL lame_get_msfix(const lame_global_flags *); + + +/******************************************************************** + * VBR control + ***********************************************************************/ +/* Types of VBR. default = vbr_off = CBR */ +int CDECL lame_set_VBR(lame_global_flags *, vbr_mode); +vbr_mode CDECL lame_get_VBR(const lame_global_flags *); + +/* VBR quality level. 0=highest 9=lowest */ +int CDECL lame_set_VBR_q(lame_global_flags *, int); +int CDECL lame_get_VBR_q(const lame_global_flags *); + +/* VBR quality level. 0=highest 9=lowest, Range [0,...,10[ */ +int CDECL lame_set_VBR_quality(lame_global_flags *, float); +float CDECL lame_get_VBR_quality(const lame_global_flags *); + +/* Ignored except for VBR=vbr_abr (ABR mode) */ +int CDECL lame_set_VBR_mean_bitrate_kbps(lame_global_flags *, int); +int CDECL lame_get_VBR_mean_bitrate_kbps(const lame_global_flags *); + +int CDECL lame_set_VBR_min_bitrate_kbps(lame_global_flags *, int); +int CDECL lame_get_VBR_min_bitrate_kbps(const lame_global_flags *); + +int CDECL lame_set_VBR_max_bitrate_kbps(lame_global_flags *, int); +int CDECL lame_get_VBR_max_bitrate_kbps(const lame_global_flags *); + +/* + 1=strictly enforce VBR_min_bitrate. Normally it will be violated for + analog silence +*/ +int CDECL lame_set_VBR_hard_min(lame_global_flags *, int); +int CDECL lame_get_VBR_hard_min(const lame_global_flags *); + +/* for preset */ +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +int CDECL lame_set_preset_expopts(lame_global_flags *, int); +#endif + +/******************************************************************** + * Filtering control + ***********************************************************************/ +/* freq in Hz to apply lowpass. Default = 0 = lame chooses. -1 = disabled */ +int CDECL lame_set_lowpassfreq(lame_global_flags *, int); +int CDECL lame_get_lowpassfreq(const lame_global_flags *); +/* width of transition band, in Hz. Default = one polyphase filter band */ +int CDECL lame_set_lowpasswidth(lame_global_flags *, int); +int CDECL lame_get_lowpasswidth(const lame_global_flags *); + +/* freq in Hz to apply highpass. Default = 0 = lame chooses. -1 = disabled */ +int CDECL lame_set_highpassfreq(lame_global_flags *, int); +int CDECL lame_get_highpassfreq(const lame_global_flags *); +/* width of transition band, in Hz. Default = one polyphase filter band */ +int CDECL lame_set_highpasswidth(lame_global_flags *, int); +int CDECL lame_get_highpasswidth(const lame_global_flags *); + + +/******************************************************************** + * psycho acoustics and other arguments which you should not change + * unless you know what you are doing + ***********************************************************************/ + +/* only use ATH for masking */ +int CDECL lame_set_ATHonly(lame_global_flags *, int); +int CDECL lame_get_ATHonly(const lame_global_flags *); + +/* only use ATH for short blocks */ +int CDECL lame_set_ATHshort(lame_global_flags *, int); +int CDECL lame_get_ATHshort(const lame_global_flags *); + +/* disable ATH */ +int CDECL lame_set_noATH(lame_global_flags *, int); +int CDECL lame_get_noATH(const lame_global_flags *); + +/* select ATH formula */ +int CDECL lame_set_ATHtype(lame_global_flags *, int); +int CDECL lame_get_ATHtype(const lame_global_flags *); + +/* lower ATH by this many db */ +int CDECL lame_set_ATHlower(lame_global_flags *, float); +float CDECL lame_get_ATHlower(const lame_global_flags *); + +/* select ATH adaptive adjustment type */ +int CDECL lame_set_athaa_type( lame_global_flags *, int); +int CDECL lame_get_athaa_type( const lame_global_flags *); + +/* select the loudness approximation used by the ATH adaptive auto-leveling */ +int CDECL lame_set_athaa_loudapprox( lame_global_flags *, int); +int CDECL lame_get_athaa_loudapprox( const lame_global_flags *); + +/* adjust (in dB) the point below which adaptive ATH level adjustment occurs */ +int CDECL lame_set_athaa_sensitivity( lame_global_flags *, float); +float CDECL lame_get_athaa_sensitivity( const lame_global_flags* ); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* OBSOLETE: predictability limit (ISO tonality formula) */ +int CDECL lame_set_cwlimit(lame_global_flags *, int); +int CDECL lame_get_cwlimit(const lame_global_flags *); +#endif + +/* + allow blocktypes to differ between channels? + default: 0 for jstereo, 1 for stereo +*/ +int CDECL lame_set_allow_diff_short(lame_global_flags *, int); +int CDECL lame_get_allow_diff_short(const lame_global_flags *); + +/* use temporal masking effect (default = 1) */ +int CDECL lame_set_useTemporal(lame_global_flags *, int); +int CDECL lame_get_useTemporal(const lame_global_flags *); + +/* use temporal masking effect (default = 1) */ +int CDECL lame_set_interChRatio(lame_global_flags *, float); +float CDECL lame_get_interChRatio(const lame_global_flags *); + +/* disable short blocks */ +int CDECL lame_set_no_short_blocks(lame_global_flags *, int); +int CDECL lame_get_no_short_blocks(const lame_global_flags *); + +/* force short blocks */ +int CDECL lame_set_force_short_blocks(lame_global_flags *, int); +int CDECL lame_get_force_short_blocks(const lame_global_flags *); + +/* Input PCM is emphased PCM (for instance from one of the rarely + emphased CDs), it is STRONGLY not recommended to use this, because + psycho does not take it into account, and last but not least many decoders + ignore these bits */ +int CDECL lame_set_emphasis(lame_global_flags *, int); +int CDECL lame_get_emphasis(const lame_global_flags *); + + + +/************************************************************************/ +/* internal variables, cannot be set... */ +/* provided because they may be of use to calling application */ +/************************************************************************/ +/* version 0=MPEG-2 1=MPEG-1 (2=MPEG-2.5) */ +int CDECL lame_get_version(const lame_global_flags *); + +/* encoder delay */ +int CDECL lame_get_encoder_delay(const lame_global_flags *); + +/* + padding appended to the input to make sure decoder can fully decode + all input. Note that this value can only be calculated during the + call to lame_encoder_flush(). Before lame_encoder_flush() has + been called, the value of encoder_padding = 0. +*/ +int CDECL lame_get_encoder_padding(const lame_global_flags *); + +/* size of MPEG frame */ +int CDECL lame_get_framesize(const lame_global_flags *); + +/* number of PCM samples buffered, but not yet encoded to mp3 data. */ +int CDECL lame_get_mf_samples_to_encode( const lame_global_flags* gfp ); + +/* + size (bytes) of mp3 data buffered, but not yet encoded. + this is the number of bytes which would be output by a call to + lame_encode_flush_nogap. NOTE: lame_encode_flush() will return + more bytes than this because it will encode the reamining buffered + PCM samples before flushing the mp3 buffers. +*/ +int CDECL lame_get_size_mp3buffer( const lame_global_flags* gfp ); + +/* number of frames encoded so far */ +int CDECL lame_get_frameNum(const lame_global_flags *); + +/* + lame's estimate of the total number of frames to be encoded + only valid if calling program set num_samples +*/ +int CDECL lame_get_totalframes(const lame_global_flags *); + +/* RadioGain value. Multiplied by 10 and rounded to the nearest. */ +int CDECL lame_get_RadioGain(const lame_global_flags *); + +/* AudiophileGain value. Multipled by 10 and rounded to the nearest. */ +int CDECL lame_get_AudiophileGain(const lame_global_flags *); + +/* the peak sample */ +float CDECL lame_get_PeakSample(const lame_global_flags *); + +/* Gain change required for preventing clipping. The value is correct only if + peak sample searching was enabled. If negative then the waveform + already does not clip. The value is multiplied by 10 and rounded up. */ +int CDECL lame_get_noclipGainChange(const lame_global_flags *); + +/* user-specified scale factor required for preventing clipping. Value is + correct only if peak sample searching was enabled and no user-specified + scaling was performed. If negative then either the waveform already does + not clip or the value cannot be determined */ +float CDECL lame_get_noclipScale(const lame_global_flags *); + + + + + + + +/* + * REQUIRED: + * sets more internal configuration based on data provided above. + * returns -1 if something failed. + */ +int CDECL lame_init_params(lame_global_flags *); + + +/* + * OPTIONAL: + * get the version number, in a string. of the form: + * "3.63 (beta)" or just "3.63". + */ +const char* CDECL get_lame_version ( void ); +const char* CDECL get_lame_short_version ( void ); +const char* CDECL get_lame_very_short_version ( void ); +const char* CDECL get_psy_version ( void ); +const char* CDECL get_lame_url ( void ); +const char* CDECL get_lame_os_bitness ( void ); + +/* + * OPTIONAL: + * get the version numbers in numerical form. + */ +typedef struct { + /* generic LAME version */ + int major; + int minor; + int alpha; /* 0 if not an alpha version */ + int beta; /* 0 if not a beta version */ + + /* version of the psy model */ + int psy_major; + int psy_minor; + int psy_alpha; /* 0 if not an alpha version */ + int psy_beta; /* 0 if not a beta version */ + + /* compile time features */ + const char *features; /* Don't make assumptions about the contents! */ +} lame_version_t; +void CDECL get_lame_version_numerical(lame_version_t *); + + +/* + * OPTIONAL: + * print internal lame configuration to message handler + */ +void CDECL lame_print_config(const lame_global_flags* gfp); + +void CDECL lame_print_internals( const lame_global_flags *gfp); + + +/* + * input pcm data, output (maybe) mp3 frames. + * This routine handles all buffering, resampling and filtering for you. + * + * return code number of bytes output in mp3buf. Can be 0 + * -1: mp3buf was too small + * -2: malloc() problem + * -3: lame_init_params() not called + * -4: psycho acoustic problems + * + * The required mp3buf_size can be computed from num_samples, + * samplerate and encoding rate, but here is a worst case estimate: + * + * mp3buf_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 mp3buf_size = 0 and LAME will not check if mp3buf_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 buffer_l[] and buffer_r[]. + * +*/ +int CDECL lame_encode_buffer ( + lame_global_flags* gfp, /* global context handle */ + const short int buffer_l [], /* PCM data for left channel */ + const short int buffer_r [], /* PCM data for right channel */ + const int nsamples, /* number of samples per channel */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + const int mp3buf_size ); /* number of valid octets in this + stream */ + +/* + * 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 CDECL lame_encode_buffer_interleaved( + lame_global_flags* gfp, /* global context handlei */ + short int pcm[], /* PCM data for left and right + channel, interleaved */ + int num_samples, /* number of samples per channel, + _not_ number of samples in + pcm[] */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + int mp3buf_size ); /* number of valid octets in this + stream */ + + +/* as lame_encode_buffer, but for 'float's. + * !! NOTE: !! data must still be scaled to be in the same range as + * short int, +/- 32768 + */ +int CDECL lame_encode_buffer_float( + lame_global_flags* gfp, /* global context handle */ + const float buffer_l [], /* PCM data for left channel */ + const float buffer_r [], /* PCM data for right channel */ + const int nsamples, /* number of samples per channel */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + const int mp3buf_size ); /* number of valid octets in this + stream */ + + +/* as lame_encode_buffer, but for long's + * !! NOTE: !! data must still be scaled to be in the same range as + * short int, +/- 32768 + * + * This scaling was a mistake (doesn't allow one to exploit full + * precision of type 'long'. Use lame_encode_buffer_long2() instead. + * + */ +int CDECL lame_encode_buffer_long( + lame_global_flags* gfp, /* global context handle */ + const long buffer_l [], /* PCM data for left channel */ + const long buffer_r [], /* PCM data for right channel */ + const int nsamples, /* number of samples per channel */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + const int mp3buf_size ); /* number of valid octets in this + stream */ + +/* Same as lame_encode_buffer_long(), but with correct scaling. + * !! NOTE: !! data must still be scaled to be in the same range as + * type 'long'. Data should be in the range: +/- 2^(8*size(long)-1) + * + */ +int CDECL lame_encode_buffer_long2( + lame_global_flags* gfp, /* global context handle */ + const long buffer_l [], /* PCM data for left channel */ + const long buffer_r [], /* PCM data for right channel */ + const int nsamples, /* number of samples per channel */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + const int mp3buf_size ); /* number of valid octets in this + stream */ + +/* as lame_encode_buffer, but for int's + * !! NOTE: !! input should be scaled to the maximum range of 'int' + * If int is 4 bytes, then the values should range from + * +/- 2147483648. + * + * This routine does not (and cannot, without loosing precision) use + * the same scaling as the rest of the lame_encode_buffer() routines. + * + */ +int CDECL lame_encode_buffer_int( + lame_global_flags* gfp, /* global context handle */ + const int buffer_l [], /* PCM data for left channel */ + const int buffer_r [], /* PCM data for right channel */ + const int nsamples, /* number of samples per channel */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + const int mp3buf_size ); /* number of valid octets in this + stream */ + + + + + +/* + * REQUIRED: + * lame_encode_flush will flush the intenal PCM buffers, padding with + * 0's to make sure the final frame is complete, and then flush + * the internal MP3 buffers, and thus may return a + * final few mp3 frames. 'mp3buf' should be at least 7200 bytes long + * to hold all possible emitted data. + * + * will also write id3v1 tags (if any) into the bitstream + * + * return code = number of bytes output to mp3buf. Can be 0 + */ +int CDECL lame_encode_flush( + lame_global_flags * gfp, /* global context handle */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + int size); /* number of valid octets in this stream */ + +/* + * OPTIONAL: + * lame_encode_flush_nogap will flush the internal mp3 buffers and pad + * the last frame with ancillary data so it is a complete mp3 frame. + * + * 'mp3buf' should be at least 7200 bytes long + * to hold all possible emitted data. + * + * After a call to this routine, the outputed mp3 data is complete, but + * you may continue to encode new PCM samples and write future mp3 data + * to a different file. The two mp3 files will play back with no gaps + * if they are concatenated together. + * + * This routine will NOT write id3v1 tags into the bitstream. + * + * return code = number of bytes output to mp3buf. Can be 0 + */ +int CDECL lame_encode_flush_nogap( + lame_global_flags * gfp, /* global context handle */ + unsigned char* mp3buf, /* pointer to encoded MP3 stream */ + int size); /* number of valid octets in this stream */ + +/* + * OPTIONAL: + * Normally, this is called by lame_init_params(). It writes id3v2 and + * Xing headers into the front of the bitstream, and sets frame counters + * and bitrate histogram data to 0. You can also call this after + * lame_encode_flush_nogap(). + */ +int CDECL lame_init_bitstream( + lame_global_flags * gfp); /* global context handle */ + + + +/* + * OPTIONAL: some simple statistics + * a bitrate histogram to visualize the distribution of used frame sizes + * a stereo mode histogram to visualize the distribution of used stereo + * modes, useful in joint-stereo mode only + * 0: LR left-right encoded + * 1: LR-I left-right and intensity encoded (currently not supported) + * 2: MS mid-side encoded + * 3: MS-I mid-side and intensity encoded (currently not supported) + * + * attention: don't call them after lame_encode_finish + * suggested: lame_encode_flush -> lame_*_hist -> lame_close + */ + +void CDECL lame_bitrate_hist( + const lame_global_flags * gfp, + int bitrate_count[14] ); +void CDECL lame_bitrate_kbps( + const lame_global_flags * gfp, + int bitrate_kbps [14] ); +void CDECL lame_stereo_mode_hist( + const lame_global_flags * gfp, + int stereo_mode_count[4] ); + +void CDECL lame_bitrate_stereo_mode_hist ( + const lame_global_flags * gfp, + int bitrate_stmode_count[14][4] ); + +void CDECL lame_block_type_hist ( + const lame_global_flags * gfp, + int btype_count[6] ); + +void CDECL lame_bitrate_block_type_hist ( + const lame_global_flags * gfp, + int bitrate_btype_count[14][6] ); + +#if (DEPRECATED_OR_OBSOLETE_CODE_REMOVED && 0) +#else +/* + * OPTIONAL: + * lame_mp3_tags_fid will rewrite 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. Make sure lame_encode_flush has been called, + * and all mp3 data has been written to the file before calling this + * function. + * 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 + * NOTE: + * LAME wants to read from the file to skip an optional ID3v2 tag, so + * make sure you opened the file for writing and reading. + * NOTE: + * You can call lame_get_lametag_frame instead, if you want to insert + * the lametag yourself. +*/ +void CDECL lame_mp3_tags_fid(lame_global_flags *, FILE* fid); +#endif + +/* + * OPTIONAL: + * lame_get_lametag_frame copies the final LAME-tag into 'buffer'. + * The function returns the number of bytes copied into buffer, or + * the required buffer size, if the provided buffer is too small. + * Function failed, if the return value is larger than 'size'! + * Make sure lame_encode flush has been called before calling this function. + * NOTE: + * if VBR tags are turned off by the user, or turned off by LAME, + * this call does nothing and returns 0. + * NOTE: + * LAME inserted an empty frame in the beginning of mp3 audio data, + * which you have to replace by the final LAME-tag frame after encoding. + * In case there is no ID3v2 tag, usually this frame will be the very first + * data in your mp3 file. If you put some other leading data into your + * file, you'll have to do some bookkeeping about where to write this buffer. + */ +size_t CDECL lame_get_lametag_frame( + const lame_global_flags *, unsigned char* buffer, size_t size); + +/* + * REQUIRED: + * final call to free all remaining buffers + */ +int CDECL lame_close (lame_global_flags *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* + * OBSOLETE: + * lame_encode_finish combines lame_encode_flush() and lame_close() in + * one call. However, once this call is made, the statistics routines + * will no longer work because the data will have been cleared, and + * lame_mp3_tags_fid() cannot be called to add data to the VBR header + */ +int CDECL lame_encode_finish( + lame_global_flags* gfp, + unsigned char* mp3buf, + int size ); +#endif + + + + + + +/********************************************************************* + * + * decoding + * + * a simple interface to mpglib, part of mpg123, is also included if + * libmp3lame is compiled with HAVE_MPGLIB + * + *********************************************************************/ +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 only computed if mpglib detects a Xing VBR header */ + unsigned long nsamp; /* number of samples in mp3 file. */ + int totalframes; /* total number of frames in mp3 file */ + + /* this data is not currently computed by the mpglib routines */ + int framenum; /* frames decoded counter */ +} mp3data_struct; + + +/* required call to initialize decoder + * NOTE: the decoder should not be used when encoding is performed + * with decoding on the fly */ +int CDECL lame_decode_init(void); + +/********************************************************************* + * input 1 mp3 frame, output (maybe) pcm data. + * + * nout = lame_decode(mp3buf,len,pcm_l,pcm_r); + * + * input: + * len : number of bytes of mp3 data in mp3buf + * mp3buf[len] : mp3 data to be decoded + * + * output: + * nout: -1 : decoding error + * 0 : need more data before we can complete the decode + * >0 : returned 'nout' samples worth of data in pcm_l,pcm_r + * pcm_l[nout] : left channel data + * pcm_r[nout] : right channel data + * + *********************************************************************/ +int CDECL lame_decode( + unsigned char * mp3buf, + int len, + short pcm_l[], + short pcm_r[] ); + +/* same as lame_decode, and also returns mp3 header data */ +int CDECL lame_decode_headers( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[], + mp3data_struct* mp3data ); + +/* same as lame_decode, but returns at most one frame */ +int CDECL lame_decode1( + unsigned 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 CDECL lame_decode1_headers( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[], + mp3data_struct* mp3data ); + +/* same as lame_decode1_headers, but also returns enc_delay and enc_padding + from VBR Info tag, (-1 if no info tag was found) */ +int CDECL lame_decode1_headersB( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[], + mp3data_struct* mp3data, + int *enc_delay, + int *enc_padding ); + + +/* cleanup call to exit decoder */ +int CDECL lame_decode_exit(void); + + + +/********************************************************************* + * + * id3tag stuff + * + *********************************************************************/ + +/* + * id3tag.h -- Interface to write ID3 version 1 and 2 tags. + * + * Copyright (C) 2000 Don Melton. + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library 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 + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA. + */ + +/* utility to obtain alphabetically sorted list of genre names with numbers */ +void CDECL id3tag_genre_list( + void (*handler)(int, const char *, void *), + void* cookie); + +void CDECL id3tag_init (lame_global_flags *gfp); + +/* force addition of version 2 tag */ +void CDECL id3tag_add_v2 (lame_global_flags *gfp); + +/* add only a version 1 tag */ +void CDECL id3tag_v1_only (lame_global_flags *gfp); + +/* add only a version 2 tag */ +void CDECL id3tag_v2_only (lame_global_flags *gfp); + +/* pad version 1 tag with spaces instead of nulls */ +void CDECL id3tag_space_v1 (lame_global_flags *gfp); + +/* pad version 2 tag with extra 128 bytes */ +void CDECL id3tag_pad_v2 (lame_global_flags *gfp); +/* pad version 2 tag with extra n bytes */ +void CDECL id3tag_set_pad (lame_global_flags *gfp, size_t n); + +void CDECL id3tag_set_title( + lame_global_flags* gfp, + const char* title ); +void CDECL id3tag_set_artist( + lame_global_flags* gfp, + const char* artist ); +void CDECL id3tag_set_album( + lame_global_flags* gfp, + const char* album ); +void CDECL id3tag_set_year( + lame_global_flags* gfp, + const char* year ); +void CDECL id3tag_set_comment( + lame_global_flags* gfp, + const char* comment ); + +/* return -1 result if track number is out of ID3v1 range + and ignored for ID3v1 */ +int CDECL id3tag_set_track( + lame_global_flags* gfp, + const char* track ); + +/* return non-zero result if genre name or number is invalid + result 0: OK + result -1: genre number out of range + result -2: no valid ID3v1 genre name, mapped to ID3v1 'Other' + but taken as-is for ID3v2 genre tag */ +int CDECL id3tag_set_genre( + lame_global_flags* gfp, + const char* genre ); + +/* return non-zero result if field name is invalid */ +int CDECL id3tag_set_fieldvalue( + lame_global_flags* gfp, + const char* fieldvalue); + +/* return non-zero result if image type is invalid */ +int CDECL id3tag_set_albumart( + lame_global_flags* gfp, + const char* image, + unsigned long size ); + +/* lame_get_id3v1_tag copies ID3v1 tag into buffer. + * Function returns number of bytes copied into buffer, or number + * of bytes rquired if buffer 'size' is too small. + * Function fails, if returned value is larger than 'size'. + * NOTE: + * This functions does nothing, if user/LAME disabled ID3v1 tag. + */ +size_t CDECL lame_get_id3v1_tag( + lame_global_flags * gfp, unsigned char* buffer, size_t size); + +/* lame_get_id3v2_tag copies ID3v2 tag into buffer. + * Function returns number of bytes copied into buffer, or number + * of bytes rquired if buffer 'size' is too small. + * Function fails, if returned value is larger than 'size'. + * NOTE: + * This functions does nothing, if user/LAME disabled ID3v2 tag. + */ +size_t CDECL lame_get_id3v2_tag( + lame_global_flags * gfp, unsigned char* buffer, size_t size); + +/* normaly lame_init_param writes ID3v2 tags into the audio stream + * Call lame_set_write_id3tag_automatic(gfp, 0) before lame_init_param + * to turn off this behaviour and get ID3v2 tag with above function + * write it yourself into your file. + */ +void CDECL lame_set_write_id3tag_automatic(lame_global_flags * gfp, int); +int CDECL lame_get_write_id3tag_automatic(lame_global_flags const* gfp); + +/*********************************************************************** +* +* list of valid bitrates [kbps] & sample frequencies [Hz]. +* first index: 0: MPEG-2 values (sample frequencies 16...24 kHz) +* 1: MPEG-1 values (sample frequencies 32...48 kHz) +* 2: MPEG-2.5 values (sample frequencies 8...12 kHz) +***********************************************************************/ +extern const int bitrate_table [3] [16]; +extern const int samplerate_table [3] [ 4]; + + + +/* maximum size of albumart image (128KB), which affects LAME_MAXMP3BUFFER + as well since lame_encode_buffer() also returns ID3v2 tag data */ +#define LAME_MAXALBUMART (128 * 1024) + +/* 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 + LAME_MAXALBUMART) + + +typedef enum { + LAME_OKAY = 0, + LAME_NOERROR = 0, + LAME_GENERICERROR = -1, + LAME_NOMEM = -10, + LAME_BADBITRATE = -11, + LAME_BADSAMPFREQ = -12, + LAME_INTERNALERROR = -13, + + FRONTEND_READERROR = -80, + FRONTEND_WRITEERROR = -81, + FRONTEND_FILETOOLARGE = -82 + +} lame_errorcodes_t; + +#if defined(__cplusplus) +} +#endif +#endif /* LAME_LAME_H */ + diff --git a/include/project_renderer.h b/include/project_renderer.h index a46cbc7b6..bddb5a534 100644 --- a/include/project_renderer.h +++ b/include/project_renderer.h @@ -37,9 +37,12 @@ public: { WaveFile, OggFile, + Mp3File, NumFileFormats } ; + static const char * EFF_ext[]; + enum Depths { Depth_16Bit, diff --git a/src/core/audio/audio_file_mp3.cpp b/src/core/audio/audio_file_mp3.cpp new file mode 100644 index 000000000..c1b63db24 --- /dev/null +++ b/src/core/audio/audio_file_mp3.cpp @@ -0,0 +1,312 @@ +/* + * audo_file_mp3.cpp - audio-device which encodes mp3-stream and writes it + * into an mp3-file. This is used for song-export. + * + * Copyright (c) 2004-2009 Tobias Doerffel + * 2009 Andrew Kelley + * + * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net + * + * 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 of the License, 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 COPYING); if not, write to the + * Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, + * Boston, MA 02110-1301 USA. + * + */ + +#include "lame.h" +#include "audio_file_mp3.h" + +#include +#include +using namespace std; + + + +AudioFileMp3::AudioFileMp3( const sample_rate_t _sample_rate, + const ch_cnt_t _channels, bool & _success_ful, + const QString & _file, + const bool _use_vbr, + const bitrate_t _nom_bitrate, + const bitrate_t _min_bitrate, + const bitrate_t _max_bitrate, + const int _depth, + mixer * _mixer ) : + audioFileDevice( _sample_rate, _channels, _file, _use_vbr, + _nom_bitrate, _min_bitrate, _max_bitrate, + _depth, _mixer ), + m_lgf( NULL ), + m_lame( NULL ), + m_outfile( NULL ), + m_hq_mode( false ) +{ + // connect to lame + m_ok = initLame("/usr/lib/libmp3lame.so.0"); + m_ok = _success_ful = m_ok && startEncoding(); +} + + +bool AudioFileMp3::initLame(QString libpath) +{ + // dynamically load the .so file + m_lame = new QLibrary(libpath); + + if( ! m_lame->load() ) return false; + + // grab the functions and stuff we need + lame_init = (lame_init_t *) + m_lame->resolve("lame_init"); + get_lame_version = (get_lame_version_t *) + m_lame->resolve("get_lame_version"); + lame_init_params = (lame_init_params_t *) + m_lame->resolve("lame_init_params"); + lame_encode_buffer = (lame_encode_buffer_t *) + m_lame->resolve("lame_encode_buffer"); + lame_encode_buffer_interleaved = (lame_encode_buffer_interleaved_t *) + m_lame->resolve("lame_encode_buffer_interleaved"); + lame_encode_flush = (lame_encode_flush_t *) + m_lame->resolve("lame_encode_flush"); + lame_close = (lame_close_t *) + m_lame->resolve("lame_close"); + + lame_set_in_samplerate = (lame_set_in_samplerate_t *) + m_lame->resolve("lame_set_in_samplerate"); + lame_set_out_samplerate = (lame_set_out_samplerate_t *) + m_lame->resolve("lame_set_out_samplerate"); + lame_set_num_channels = (lame_set_num_channels_t *) + m_lame->resolve("lame_set_num_channels"); + lame_set_quality = (lame_set_quality_t *) + m_lame->resolve("lame_set_quality"); + lame_set_brate = (lame_set_brate_t *) + m_lame->resolve("lame_set_brate"); + lame_set_VBR = (lame_set_VBR_t *) + m_lame->resolve("lame_set_VBR"); + lame_set_VBR_q = (lame_set_VBR_q_t *) + m_lame->resolve("lame_set_VBR_q"); + lame_set_VBR_min_bitrate_kbps = (lame_set_VBR_min_bitrate_kbps_t *) + m_lame->resolve("lame_set_VBR_min_bitrate_kbps"); + lame_set_mode = (lame_set_mode_t *) + m_lame->resolve("lame_set_mode"); + lame_set_preset = (lame_set_preset_t *) + m_lame->resolve("lame_set_preset"); + lame_set_error_protection = (lame_set_error_protection_t *) + m_lame->resolve("lame_set_error_protection"); + lame_set_disable_reservoir = (lame_set_disable_reservoir_t *) + m_lame->resolve("lame_set_disable_reservoir"); + lame_set_padding_type = (lame_set_padding_type_t *) + m_lame->resolve("lame_set_padding_type"); + lame_set_bWriteVbrTag = (lame_set_bWriteVbrTag_t *) + m_lame->resolve("lame_set_bWriteVbrTag"); + + lame_set_findReplayGain = (lame_set_findReplayGain_t *) + m_lame->resolve("lame_set_findReplayGain"); + lame_set_VBR_quality = (lame_set_VBR_quality_t *) + m_lame->resolve("lame_set_VBR_quality"); + lame_set_VBR_mean_bitrate_kbps = (lame_set_VBR_mean_bitrate_kbps_t *) + m_lame->resolve("lame_set_VBR_mean_bitrate_kbps"); + lame_set_VBR_max_bitrate_kbps = (lame_set_VBR_max_bitrate_kbps_t *) + m_lame->resolve("lame_set_VBR_max_bitrate_kbps"); + + // These are optional + lame_get_lametag_frame = (lame_get_lametag_frame_t *) + m_lame->resolve("lame_get_lametag_frame"); + lame_mp3_tags_fid = (lame_mp3_tags_fid_t *) + m_lame->resolve("lame_mp3_tags_fid"); + + if (!lame_init || + !get_lame_version || + !lame_init_params || + !lame_encode_buffer || + !lame_encode_buffer_interleaved || + !lame_encode_flush || + !lame_close || + !lame_set_in_samplerate || + !lame_set_out_samplerate || + !lame_set_num_channels || + !lame_set_quality || + !lame_set_brate || + !lame_set_VBR || + !lame_set_VBR_q || + !lame_set_mode || + !lame_set_preset || + !lame_set_error_protection || + !lame_set_disable_reservoir || + !lame_set_padding_type || + !lame_set_bWriteVbrTag) + { + // some symbols are missing + printf("AudioFileMp3: some symbols are missing from the lame library\n"); + m_lame->unload(); + return false; + } + + return true; +} + + + +AudioFileMp3::~AudioFileMp3() +{ + finishEncoding(); + + if( m_lame ) + { + if( m_lame->isLoaded() ) + m_lame->unload(); + + delete m_lame; + m_lame = NULL; + } + + if( m_outfile ) + { + if( m_outfile->isOpen() ) + m_outfile->close(); + + delete m_outfile; + } +} + + +void AudioFileMp3::finishEncoding( void ) +{ + if( m_lgf ) + { + // flush + int bufSize = 7200; + unsigned char * out = new unsigned char[bufSize]; + int rc = lame_encode_flush(m_lgf, out, bufSize); + + if( m_outfile && m_outfile->isOpen() ){ + m_outfile->write( (const char *) out, rc ); + + m_outfile->close(); + delete m_outfile; + m_outfile = NULL; + } + + // cleanup + delete[] out; + + // close any open handles we may have + lame_close(m_lgf); + m_lgf = NULL; + } +} + + +bool AudioFileMp3::startEncoding( void ) +{ + // open any handles, files, etc + m_lgf = lame_init(); + if( m_lgf == NULL ){ + printf("AudioFileMp3: Unable to initialize lame\n"); + return false; + } + + if( channels() > 2 ) + printf("I don't think lame can do more than 2 channels\n"); + + lame_set_in_samplerate(m_lgf, sampleRate() ); + lame_set_num_channels(m_lgf, channels() ); + + if( m_hq_mode ) + lame_set_quality(m_lgf, 0); // best, very slow + else + lame_set_quality(m_lgf, 2); // near-best, not too slow + + lame_set_mode(m_lgf, STEREO); + lame_set_findReplayGain(m_lgf, 1); // perform ReplayGain analysis + + if( useVBR() == 0 ) + { + lame_set_VBR(m_lgf, vbr_off); + lame_set_brate(m_lgf, nominalBitrate() ); + } + else + { + lame_set_VBR(m_lgf, vbr_abr); + lame_set_VBR_quality(m_lgf, 2); + lame_set_VBR_mean_bitrate_kbps(m_lgf, nominalBitrate() ); + lame_set_VBR_min_bitrate_kbps(m_lgf, minBitrate() ); + lame_set_VBR_max_bitrate_kbps(m_lgf, maxBitrate() ); + } + + if( lame_init_params( m_lgf ) < 0 ) + return false; + + // open the file + m_outfile = new QFile( outputFile() ); + if( ! m_outfile->open( QIODevice::WriteOnly ) ) + { + printf("AudioFileMp3: unable to open file for output\n"); + return false; + } + + // write the headers and such + + return true; + +} + + +short int AudioFileMp3::rescale(float sample) { + return (sample / 1) * std::numeric_limits::max(); +} + +// encode data and write to file +void AudioFileMp3::writeBuffer( const surroundSampleFrame * _ab, + const fpp_t _frames, const float _master_gain ) +{ + // encode with lame + int bufSize = 1.25 * _frames + 7200; + short int * in = new short int[_frames*2]; + unsigned char * out = new unsigned char[bufSize]; + + // scale to short int instead of float + for(int i=0; i < _frames; ++i) + { + in[i*2] = rescale( _ab[i][0] ); + in[i*2+1] = rescale( _ab[i][1] ); + } + + int rc = lame_encode_buffer_interleaved( m_lgf, in, _frames, out, bufSize); + + switch(rc){ + case -1: + printf("AudioFileMp3: encode error: buffer too small.\n"); + return; + case -2: + printf("AudioFileMp3: encode error: out of memory\n"); + return; + case -3: + printf("AudioFileMp3: encode error: lame_init_params not called\n"); + return; + case -4: + printf("AudioFileMp3: encode error: psycho acoustic problems\n"); + return; + } + + // write to file + m_outfile->write( (const char *) out, rc ); + + // clean up + delete[] out; + delete[] in; +} + + + + + + diff --git a/src/core/main.cpp b/src/core/main.cpp index 9406d2c48..80969fdb1 100644 --- a/src/core/main.cpp +++ b/src/core/main.cpp @@ -162,7 +162,7 @@ int main( int argc, char * * argv ) "-r, --render render given project file\n" "-o, --output render into \n" "-f, --output-format specify format of render-output where\n" - " format is either 'wav' or 'ogg'.\n" + " format is either 'wav', 'ogg', or 'mp3'.\n" "-s, --samplerate specify output samplerate in Hz\n" " range: 44100 (default) to 192000\n" "-b, --bitrate specify output bitrate in kHz\n" @@ -227,6 +227,10 @@ int main( int argc, char * * argv ) eff = projectRenderer::OggFile; } #endif + else if( ext == "mp3" ) + { + eff = projectRenderer::Mp3File; + } else { printf( "\nInvalid output format %s.\n\n" @@ -491,10 +495,7 @@ int main( int argc, char * * argv ) { // create renderer projectRenderer * r = new projectRenderer( qs, os, eff, - render_out + - QString( ( eff == - projectRenderer::WaveFile ) ? - "wav" : "ogg" ) ); + render_out + QString(projectRenderer::EFF_ext[eff])); QCoreApplication::instance()->connect( r, SIGNAL( finished() ), SLOT( quit() ) ); diff --git a/src/core/project_renderer.cpp b/src/core/project_renderer.cpp index 1a3a8b8dc..69084bb84 100644 --- a/src/core/project_renderer.cpp +++ b/src/core/project_renderer.cpp @@ -31,6 +31,7 @@ #include "audio_file_wave.h" #include "audio_file_ogg.h" +#include "audio_file_mp3.h" #ifdef LMMS_HAVE_PTHREAD_H #include @@ -53,6 +54,9 @@ fileEncodeDevice __fileEncodeDevices[] = NULL #endif }, + { projectRenderer::Mp3File, + QT_TRANSLATE_NOOP( "projectRenderer", "MP3 File (*.mp3)" ), + ".mp3", &AudioFileMp3::getInst }, // ... insert your own file-encoder-infos here... may be one day the // user can add own encoders inside the program... @@ -60,7 +64,7 @@ fileEncodeDevice __fileEncodeDevices[] = } ; - +const char * projectRenderer::EFF_ext[] = {"wav", "ogg", "mp3"}; projectRenderer::projectRenderer( const mixer::qualitySettings & _qs,