diff --git a/OnLive/src/main/AndroidManifest.xml b/OnLive/src/main/AndroidManifest.xml index 4b6de5d..9bed52c 100644 --- a/OnLive/src/main/AndroidManifest.xml +++ b/OnLive/src/main/AndroidManifest.xml @@ -5,7 +5,6 @@ - + +lame_global_flags *glf; + +lame_global_flags *initializeDefault(JNIEnv *env) { + lame_global_flags *glf = lame_init(); + lame_init_params(glf); + return glf; +} + +lame_global_flags *initialize( + JNIEnv *env, + jint inSamplerate, jint outChannel, + jint outSamplerate, jint outBitrate, jfloat scaleInput, jint mode, jint vbrMode, + jint quality, jint vbrQuality, jint abrMeanBitrate, jint lowpassFreq, jint highpassFreq, + jstring id3tagTitle, jstring id3tagArtist, jstring id3tagAlbum, + jstring id3tagYear, jstring id3tagComment) { + + lame_global_flags *glf = lame_init(); + lame_set_in_samplerate(glf, inSamplerate); + lame_set_num_channels(glf, outChannel); + lame_set_out_samplerate(glf, outSamplerate); + lame_set_brate(glf, outBitrate); + lame_set_quality(glf, quality); + lame_set_scale(glf, scaleInput); + lame_set_VBR_q(glf, vbrQuality); + lame_set_VBR_mean_bitrate_kbps(glf, abrMeanBitrate); + lame_set_lowpassfreq(glf, lowpassFreq); + lame_set_highpassfreq(glf, highpassFreq); + + switch (mode) { + case 0: + lame_set_mode(glf, STEREO); + break; + case 1: + lame_set_mode(glf, JOINT_STEREO); + break; + case 3: + lame_set_mode(glf, MONO); + break; + case 4: + lame_set_mode(glf, NOT_SET); + break; + default: + lame_set_mode(glf, NOT_SET); + break; + } + + switch (vbrMode) { + case 0: + lame_set_VBR(glf, vbr_off); + break; + case 2: + lame_set_VBR(glf, vbr_rh); + break; + case 3: + lame_set_VBR(glf, vbr_abr); + break; + case 4: + lame_set_VBR(glf, vbr_mtrh); + break; + case 6: + lame_set_VBR(glf, vbr_default); + break; + default: + lame_set_VBR(glf, vbr_off); + break; + + } + + + const jchar *title = NULL; + const jchar *artist = NULL; + const jchar *album = NULL; + const jchar *year = NULL; + const jchar *comment = NULL; + if (id3tagTitle) { + title = (*env)->GetStringChars(env, id3tagTitle, NULL); + } + if (id3tagArtist) { + artist = (*env)->GetStringChars(env, id3tagArtist, NULL); + } + if (id3tagAlbum) { + album = (*env)->GetStringChars(env, id3tagAlbum, NULL); + } + if (id3tagYear) { + year = (*env)->GetStringChars(env, id3tagYear, NULL); + } + if (id3tagComment) { + comment = (*env)->GetStringChars(env, id3tagComment, NULL); + } + + if (title || artist || album || year || comment) { + id3tag_init(glf); + + if (title) { + id3tag_set_title(glf, (const char *) title); + (*env)->ReleaseStringChars(env, id3tagTitle, title); + } + if (artist) { + id3tag_set_artist(glf, (const char *) artist); + (*env)->ReleaseStringChars(env, id3tagArtist, artist); + } + if (album) { + id3tag_set_album(glf, (const char *) album); + (*env)->ReleaseStringChars(env, id3tagAlbum, album); + } + if (year) { + id3tag_set_year(glf, (const char *) year); + (*env)->ReleaseStringChars(env, id3tagYear, year); + } + if (comment) { + id3tag_set_comment(glf, (const char *) comment); + (*env)->ReleaseStringChars(env, id3tagComment, comment); + } + } + + lame_init_params(glf); + + + return glf; +} + +jint encode( + JNIEnv *env, lame_global_flags *glf, + jshortArray buffer_l, jshortArray buffer_r, + jint samples, jbyteArray mp3buf) { + jshort *j_buffer_l = (*env)->GetShortArrayElements(env, buffer_l, NULL); + + jshort *j_buffer_r = (*env)->GetShortArrayElements(env, buffer_r, NULL); + + const jsize mp3buf_size = (*env)->GetArrayLength(env, mp3buf); + jbyte *j_mp3buf = (*env)->GetByteArrayElements(env, mp3buf, NULL); + + int result = lame_encode_buffer(glf, j_buffer_l, j_buffer_r, + samples, j_mp3buf, mp3buf_size); + + (*env)->ReleaseShortArrayElements(env, buffer_l, j_buffer_l, 0); + (*env)->ReleaseShortArrayElements(env, buffer_r, j_buffer_r, 0); + (*env)->ReleaseByteArrayElements(env, mp3buf, j_mp3buf, 0); + + return result; +} + +jint encodeBufferInterleaved( + JNIEnv *env, lame_global_flags *glf, + jshortArray pcm, jint samples, jbyteArray mp3buf) { + jshort *j_pcm = (*env)->GetShortArrayElements(env, pcm, NULL); + + const jsize mp3buf_size = (*env)->GetArrayLength(env, mp3buf); + jbyte *j_mp3buf = (*env)->GetByteArrayElements(env, mp3buf, NULL); + + int result = lame_encode_buffer_interleaved(glf, j_pcm, + samples, j_mp3buf, mp3buf_size); + + (*env)->ReleaseShortArrayElements(env, pcm, j_pcm, 0); + (*env)->ReleaseByteArrayElements(env, mp3buf, j_mp3buf, 0); + + return result; +} + +jint flush( + JNIEnv *env, lame_global_flags *glf, + jbyteArray mp3buf) { + const jsize mp3buf_size = (*env)->GetArrayLength(env, mp3buf); + jbyte *j_mp3buf = (*env)->GetByteArrayElements(env, mp3buf, NULL); + + int result = lame_encode_flush(glf, j_mp3buf, mp3buf_size); + + (*env)->ReleaseByteArrayElements(env, mp3buf, j_mp3buf, 0); + + return result; +} + +void close_lame(lame_global_flags *glf) { + lame_close(glf); + glf = NULL; +} + + +JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_lameInitDefault + (JNIEnv *env, jclass jclazz) { + + glf = initializeDefault(env); +} + +JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_lameInit( + JNIEnv *env, jclass cls, jint inSampleRate, jint outChannel, + jint outSampleRate, jint outBitrate, jfloat scaleInput, jint mode, jint vbrMode, + jint quality, jint vbrQuality, jint abrMeanBitrate, jint lowPassFreq, jint highPassFreq, + jstring id3tagTitle, jstring id3tagArtist, jstring id3tagAlbum, + jstring id3tagYear, jstring id3tagComment) { + + glf = initialize(env, inSampleRate, outChannel, outSampleRate, outBitrate, scaleInput, mode, + vbrMode, + quality, vbrQuality, abrMeanBitrate, lowPassFreq, highPassFreq, id3tagTitle, + id3tagArtist, id3tagAlbum, + id3tagYear, + id3tagComment); +} + +JNIEXPORT jint JNICALL Java_com_frank_ffmpeg_AudioPlayer_lameEncode( + JNIEnv *env, jclass cls, jshortArray buffer_l, + jshortArray buffer_r, jint samples, jbyteArray mp3buf) { + return encode(env, glf, buffer_l, buffer_r, samples, mp3buf); +} + +JNIEXPORT jint JNICALL Java_com_frank_ffmpeg_AudioPlayer_encodeBufferInterleaved( + JNIEnv *env, jclass cls, jshortArray pcm, + jint samples, jbyteArray mp3buf) { + return encodeBufferInterleaved(env, glf, pcm, samples, mp3buf); +} + +JNIEXPORT jint JNICALL Java_com_frank_ffmpeg_AudioPlayer_lameFlush( + JNIEnv *env, jclass cls, jbyteArray mp3buf) { + return flush(env, glf, mp3buf); +} + +JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_lameClose( + JNIEnv *env, jclass cls) { + close_lame(glf); +} \ No newline at end of file diff --git a/app/src/main/cpp/include/lame/lame.h b/app/src/main/cpp/include/lame/lame.h new file mode 100644 index 0000000..21c8afd --- /dev/null +++ b/app/src/main/cpp/include/lame/lame.h @@ -0,0 +1,1323 @@ +/* + * 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 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. + */ + +/* $Id: include.lame.h,v 1.189.2.1 2012/01/08 23:49:58 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 + +typedef void (*lame_report_function)(const char *format, va_list ap); + +#if defined(WIN32) || 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; + + +/* buffer considerations */ +typedef enum buffer_constraint_e { + MDB_DEFAULT=0, + MDB_STRICT_ISO=1, + MDB_MAXIMUM=2 +} buffer_constraint; + + +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=1 + (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=1 + (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=1 + (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 include.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: include.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 + * include.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 *, lame_report_function); +int CDECL lame_set_debugf(lame_global_flags *, lame_report_function); +int CDECL lame_set_msgf (lame_global_flags *, lame_report_function); + + + +/* 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 = include.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 = include.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 *); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* 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 *); +#endif + +/* 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 *); + +/* + include.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 include.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 pcm_l [], /* PCM data for left channel */ + const float pcm_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 'float's. + * !! NOTE: !! data must be scaled to +/- 1 full scale + */ +int CDECL lame_encode_buffer_ieee_float( + lame_t gfp, + const float pcm_l [], /* PCM data for left channel */ + const float pcm_r [], /* PCM data for right channel */ + const int nsamples, + unsigned char * mp3buf, + const int mp3buf_size); +int CDECL lame_encode_buffer_interleaved_ieee_float( + lame_t gfp, + const float pcm[], /* PCM data for left and right + channel, interleaved */ + const int nsamples, + unsigned char * mp3buf, + const int mp3buf_size); + +/* as lame_encode_buffer, but for 'double's. + * !! NOTE: !! data must be scaled to +/- 1 full scale + */ +int CDECL lame_encode_buffer_ieee_double( + lame_t gfp, + const double pcm_l [], /* PCM data for left channel */ + const double pcm_r [], /* PCM data for right channel */ + const int nsamples, + unsigned char * mp3buf, + const int mp3buf_size); +int CDECL lame_encode_buffer_interleaved_ieee_double( + lame_t gfp, + const double pcm[], /* PCM data for left and right + channel, interleaved */ + const int nsamples, + unsigned char * mp3buf, + const int mp3buf_size); + +/* 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 + * + *********************************************************************/ + +struct hip_global_struct; +typedef struct hip_global_struct hip_global_flags; +typedef hip_global_flags *hip_t; + + +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 */ +hip_t CDECL hip_decode_init(void); + +/* cleanup call to exit decoder */ +int CDECL hip_decode_exit(hip_t gfp); + +/* HIP reporting functions */ +void CDECL hip_set_errorf(hip_t gfp, lame_report_function f); +void CDECL hip_set_debugf(hip_t gfp, lame_report_function f); +void CDECL hip_set_msgf (hip_t gfp, lame_report_function f); + +/********************************************************************* + * input 1 mp3 frame, output (maybe) pcm data. + * + * nout = hip_decode(hip, 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 hip_decode( hip_t gfp + , unsigned char * mp3buf + , size_t len + , short pcm_l[] + , short pcm_r[] + ); + +/* same as hip_decode, and also returns mp3 header data */ +int CDECL hip_decode_headers( hip_t gfp + , unsigned char* mp3buf + , size_t len + , short pcm_l[] + , short pcm_r[] + , mp3data_struct* mp3data + ); + +/* same as hip_decode, but returns at most one frame */ +int CDECL hip_decode1( hip_t gfp + , unsigned char* mp3buf + , size_t len + , short pcm_l[] + , short pcm_r[] + ); + +/* same as hip_decode1, but returns at most one frame and mp3 header data */ +int CDECL hip_decode1_headers( hip_t gfp + , unsigned char* mp3buf + , size_t len + , short pcm_l[] + , short pcm_r[] + , mp3data_struct* mp3data + ); + +/* same as hip_decode1_headers, but also returns enc_delay and enc_padding + from VBR Info tag, (-1 if no info tag was found) */ +int CDECL hip_decode1_headersB( hip_t gfp + , unsigned char* mp3buf + , size_t len + , short pcm_l[] + , short pcm_r[] + , mp3data_struct* mp3data + , int *enc_delay + , int *enc_padding + ); + + + +/* OBSOLETE: + * lame_decode... functions are there to keep old code working + * but it is strongly recommended to replace calls by hip_decode... + * function calls, see above. + */ +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +int CDECL lame_decode_init(void); +int CDECL lame_decode( + unsigned char * mp3buf, + int len, + short pcm_l[], + short pcm_r[] ); +int CDECL lame_decode_headers( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[], + mp3data_struct* mp3data ); +int CDECL lame_decode1( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[] ); +int CDECL lame_decode1_headers( + unsigned char* mp3buf, + int len, + short pcm_l[], + short pcm_r[], + mp3data_struct* mp3data ); +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 ); +int CDECL lame_decode_exit(void); + +#endif /* obsolete lame_decode API calls */ + + +/********************************************************************* + * + * 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_t gfp); + +/* force addition of version 2 tag */ +void CDECL id3tag_add_v2 (lame_t gfp); + +/* add only a version 1 tag */ +void CDECL id3tag_v1_only (lame_t gfp); + +/* add only a version 2 tag */ +void CDECL id3tag_v2_only (lame_t gfp); + +/* pad version 1 tag with spaces instead of nulls */ +void CDECL id3tag_space_v1 (lame_t gfp); + +/* pad version 2 tag with extra 128 bytes */ +void CDECL id3tag_pad_v2 (lame_t gfp); + +/* pad version 2 tag with extra n bytes */ +void CDECL id3tag_set_pad (lame_t gfp, size_t n); + +void CDECL id3tag_set_title(lame_t gfp, const char* title); +void CDECL id3tag_set_artist(lame_t gfp, const char* artist); +void CDECL id3tag_set_album(lame_t gfp, const char* album); +void CDECL id3tag_set_year(lame_t gfp, const char* year); +void CDECL id3tag_set_comment(lame_t 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_t 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_t gfp, const char* genre); + +/* return non-zero result if field name is invalid */ +int CDECL id3tag_set_fieldvalue(lame_t gfp, const char* fieldvalue); + +/* return non-zero result if image type is invalid */ +int CDECL id3tag_set_albumart(lame_t gfp, const char* image, size_t 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_t 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_t 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); + +/* experimental */ +int CDECL id3tag_set_textinfo_latin1(lame_t gfp, char const *id, char const *text); + +/* experimental */ +int CDECL id3tag_set_comment_latin1(lame_t gfp, char const *lang, char const *desc, char const *text); + +#if DEPRECATED_OR_OBSOLETE_CODE_REMOVED +#else +/* experimental */ +int CDECL id3tag_set_textinfo_ucs2(lame_t gfp, char const *id, unsigned short const *text); + +/* experimental */ +int CDECL id3tag_set_comment_ucs2(lame_t gfp, char const *lang, + unsigned short const *desc, unsigned short const *text); + +/* experimental */ +int CDECL id3tag_set_fieldvalue_ucs2(lame_t gfp, const unsigned short *fieldvalue); +#endif + +/* experimental */ +int CDECL id3tag_set_fieldvalue_utf16(lame_t gfp, const unsigned short *fieldvalue); + +/* experimental */ +int CDECL id3tag_set_textinfo_utf16(lame_t gfp, char const *id, unsigned short const *text); + +/* experimental */ +int CDECL id3tag_set_comment_utf16(lame_t gfp, char const *lang, unsigned short const *desc, unsigned short const *text); + + +/*********************************************************************** +* +* 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]; + +/* access functions for use in DLL, global vars are not exported */ +int CDECL lame_get_bitrate(int mpeg_version, int table_index); +int CDECL lame_get_samplerate(int mpeg_version, int table_index); + + +/* 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/app/src/main/cpp/openSL_audio_player.c b/app/src/main/cpp/openSL_audio_player.c index f683df8..bc6bef4 100644 --- a/app/src/main/cpp/openSL_audio_player.c +++ b/app/src/main/cpp/openSL_audio_player.c @@ -271,7 +271,7 @@ int releaseAudioPlayer() { } JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_playAudio - (JNIEnv * env, jclass jobject, jstring filePath) { + (JNIEnv * env, jclass jclazz, jstring filePath) { int rate, channel; const char *file_name = (*env)->GetStringUTFChars(env, filePath, NULL); @@ -292,7 +292,7 @@ JNIEXPORT void JNICALL Java_com_frank_ffmpeg_AudioPlayer_playAudio //停止播放，释放相关资源 JNIEXPORT jint JNICALL Java_com_frank_ffmpeg_AudioPlayer_stop - (JNIEnv * env, jclass jobject) { + (JNIEnv * env, jclass jclazz) { if (bqPlayerObject != NULL) { (*bqPlayerObject)->Destroy(bqPlayerObject); bqPlayerObject = NULL; diff --git a/app/src/main/java/com/frank/ffmpeg/AudioPlayer.java b/app/src/main/java/com/frank/ffmpeg/AudioPlayer.java index f6bbae8..cb82279 100644 --- a/app/src/main/java/com/frank/ffmpeg/AudioPlayer.java +++ b/app/src/main/java/com/frank/ffmpeg/AudioPlayer.java @@ -18,7 +18,25 @@ public class AudioPlayer { //调用OpenSL ES播放 public native void playAudio(String audioPath); //调用OpenSL ES播放 - public native void stop(String audioPath); + public native int stop(); + + public native static void lameInitDefault(); + + public native static void lameInit(int inSamplerate, int outChannel, + int outSamplerate, int outBitrate, float scaleInput, int mode, int vbrMode, + int quality, int vbrQuality, int abrMeanBitrate, int lowpassFreq, int highpassFreq, String id3tagTitle, + String id3tagArtist, String id3tagAlbum, String id3tagYear, + String id3tagComment); + + public native static int lameEncode(short[] buffer_l, short[] buffer_r, + int samples, byte[] mp3buf); + + public native static int encodeBufferInterleaved(short[] pcm, int samples, + byte[] mp3buf); + + public native static int lameFlush(byte[] mp3buf); + + public native static void lameClose(); /** * 创建一个AudioTrack对象 diff --git a/app/src/main/java/com/frank/ffmpeg/activity/AudioHandleActivity.java b/app/src/main/java/com/frank/ffmpeg/activity/AudioHandleActivity.java index 790442b..c291159 100644 --- a/app/src/main/java/com/frank/ffmpeg/activity/AudioHandleActivity.java +++ b/app/src/main/java/com/frank/ffmpeg/activity/AudioHandleActivity.java @@ -14,6 +14,7 @@ import java.io.File; import com.frank.ffmpeg.AudioPlayer; import com.frank.ffmpeg.FFmpegCmd; import com.frank.ffmpeg.R; +import com.frank.ffmpeg.mp3.Mp3Converter; import com.frank.ffmpeg.util.FFmpegUtil; import com.frank.ffmpeg.util.FileUtil; @@ -140,8 +141,13 @@ public class AudioHandleActivity extends AppCompatActivity implements View.OnCli } switch (handleType){ case 0://转码 - String transformFile = PATH + File.separator + "transform.aac"; - commandLine = FFmpegUtil.transformAudio(srcFile, transformFile); +// String transformFile = PATH + File.separator + "transform.aac"; +// commandLine = FFmpegUtil.transformAudio(srcFile, transformFile); + //使用mp3lame进行转码 + String inputFile = PATH + File.separator + "hello.aac"; + String transformFile = PATH + File.separator + "transform.mp3"; + Mp3Converter mp3Converter = new Mp3Converter(); + mp3Converter.convertToMp3(inputFile, transformFile); break; case 1://剪切 String cutFile = PATH + File.separator + "cut.mp3"; diff --git a/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Converter.java b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Converter.java new file mode 100644 index 0000000..8f4ac13 --- /dev/null +++ b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Converter.java @@ -0,0 +1,340 @@ +package com.frank.ffmpeg.mp3; + +import android.annotation.TargetApi; +import android.media.MediaCodec; +import android.media.MediaCodec.BufferInfo; +import android.media.MediaExtractor; +import android.media.MediaFormat; +import android.os.Build; +import android.util.Log; + +import java.io.BufferedOutputStream; +import java.io.File; +import java.io.FileOutputStream; +import java.io.IOException; +import java.nio.ByteBuffer; +import java.nio.ByteOrder; +import java.nio.ShortBuffer; +import java.util.concurrent.BlockingDeque; +import java.util.concurrent.LinkedBlockingDeque; + +@TargetApi(Build.VERSION_CODES.JELLY_BEAN) +public class Mp3Converter { + + private final static String TAG = Mp3Converter.class.getSimpleName(); + + private MediaCodec mMediaCodec; + private MediaExtractor mediaExtractor; + private BufferInfo bufferInfo; + private ByteBuffer[] rawInputBuffers; + private ByteBuffer[] encodedOutputBuffers; + private int inSampleRate; + private int channels; + + private final static int DEFAULT_QUEUE_SIZE = 512; + private BlockingDeque writeQueue = new LinkedBlockingDeque<>(DEFAULT_QUEUE_SIZE); + private BlockingDeque encodeQueue = new LinkedBlockingDeque<>(DEFAULT_QUEUE_SIZE); + private byte[] mp3buf; + + private boolean decodeFinished; + private boolean encodeFinished; + + private long readSize; + private long decodeSize; + private long encodeSize; + + private Mp3Lame mp3Lame; + private WriteThread writeThread; + private EncodeThread encodeThread; + private long lastPts; + + private class BufferDecoded { + int channels; + short[] leftBuffer; + short[] rightBuffer; + short[] pcm; + long pts; + } + + private class BufferEncoded { + byte[] buffer; + int length; + } + + private class WriteThread extends Thread { + + private String mp3Path; + private FileOutputStream fos; + private BufferedOutputStream bos; + + WriteThread(String path) { + super(); + mp3Path = path; + } + + @Override + public void run() { + try { + Log.i(TAG, "WriteThread start"); + + fos = new FileOutputStream(new File(mp3Path)); + bos = new BufferedOutputStream(fos, 200 * 1024); + + while (true) { + if (encodeFinished && writeQueue.size() == 0) { + break; + } + + BufferEncoded buffer = null; + try { + buffer = writeQueue.take(); + } catch (InterruptedException e) { + Log.e(TAG, "WriteThread InterruptedException=" + e.toString()); + } + + if(buffer != null) { + bos.write(buffer.buffer, 0, buffer.length); + } + } + bos.flush(); + bos.close(); + } catch (Exception e) { + e.printStackTrace(); + } finally { + try { + bos.flush(); + bos.close(); + } catch (IOException e) { + e.printStackTrace(); + } + } + + Log.i(TAG, "WriteThread end"); + } + } + + private class EncodeThread extends Thread { + @Override + public void run() { + Log.i(TAG, "EncodeThread start"); + + while (true) { + if (decodeFinished && encodeQueue.size() == 0) { + break; + } + BufferDecoded buffer = null; + try { + buffer = encodeQueue.take(); + } catch (InterruptedException e) { + Log.e(TAG, "EncodeThread InterruptedException=" + e.toString()); + } + if (buffer != null) { + encodeToMp3(buffer); + } + } + encodeFinished = true; + + writeThread.interrupt(); + + Log.i(TAG, "EncodeThread end"); + } + } + + private class DecodeThread extends Thread { + @Override + public void run() { + long startTime = System.currentTimeMillis(); + try { + Log.i(TAG, "DecodeThread start"); + + while (true) { + int outputBufIndex = mMediaCodec.dequeueOutputBuffer(bufferInfo, -1); + if (outputBufIndex >= 0) { + ByteBuffer buffer = encodedOutputBuffers[outputBufIndex]; + decodeSize += bufferInfo.size; + ShortBuffer shortBuffer = buffer.order(ByteOrder.nativeOrder()).asShortBuffer(); + + short[] leftBuffer = null; + short[] rightBuffer = null; + short[] pcm = null; + + if (channels == 2) { + pcm = new short[shortBuffer.remaining()]; + shortBuffer.get(pcm); + } else { + leftBuffer = new short[shortBuffer.remaining()]; + rightBuffer = leftBuffer; + shortBuffer.get(leftBuffer); + Log.e(TAG, "single channel leftBuffer.length = " + leftBuffer.length); + } + + buffer.clear(); + + BufferDecoded bufferDecoded = new BufferDecoded(); + bufferDecoded.leftBuffer = leftBuffer; + bufferDecoded.rightBuffer = rightBuffer; + bufferDecoded.pcm = pcm; + bufferDecoded.channels = channels; + bufferDecoded.pts = bufferInfo.presentationTimeUs; + encodeQueue.put(bufferDecoded); + + mMediaCodec.releaseOutputBuffer(outputBufIndex, false); + + if ((bufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0) { + Log.e(TAG, "DecodeThread get BUFFER_FLAG_END_OF_STREAM"); + decodeFinished = true; + break; + } + } else if (outputBufIndex == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) { + encodedOutputBuffers = mMediaCodec.getOutputBuffers(); + } else if (outputBufIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) { + final MediaFormat oformat = mMediaCodec.getOutputFormat(); + Log.d(TAG, "Output format has changed to " + oformat); + } + } + } catch (Exception e) { + e.printStackTrace(); + } + + encodeThread.interrupt(); + long endTime = System.currentTimeMillis(); + Log.i(TAG, "DecodeThread finished time=" + (endTime - startTime) / 1000); + } + } + + public void convertToMp3(String srcPath, String mp3Path) { + + long startTime = System.currentTimeMillis(); + long endTime; + + encodeThread = new EncodeThread(); + writeThread = new WriteThread(mp3Path); + DecodeThread decodeThread = new DecodeThread(); + + encodeThread.start(); + writeThread.start(); + prepareDecode(srcPath); + decodeThread.start(); + readSampleData(); + + try { + writeThread.join(); + } catch (InterruptedException e) { + Log.e(TAG, "convertToMp3 InterruptedException=" + e.toString()); + } + + double mReadSize = readSize / 1024.0 / 1024.0; + double mDecodeSize = decodeSize / 1024.0 / 1024.0; + double mEncodeSize = encodeSize / 1024.0 / 1024.0; + Log.i(TAG, "readSize=" + mReadSize + ", decodeSize=" + mDecodeSize + ",encodeSize=" + mEncodeSize); + + endTime = System.currentTimeMillis(); + Log.i(TAG, "convertToMp3 finish time=" + (endTime - startTime) / 1000); + } + + private void prepareDecode(String path) { + try { + mediaExtractor = new MediaExtractor(); + mediaExtractor.setDataSource(path); + for (int i = 0; i < mediaExtractor.getTrackCount(); i++) { + MediaFormat mMediaFormat = mediaExtractor.getTrackFormat(i); + Log.i(TAG, "prepareDecode get mMediaFormat=" + mMediaFormat); + + String mime = mMediaFormat.getString(MediaFormat.KEY_MIME); + if (mime.startsWith("audio")) { + mMediaCodec = MediaCodec.createDecoderByType(mime); + mMediaCodec.configure(mMediaFormat, null, null, 0); + mediaExtractor.selectTrack(i); + inSampleRate = mMediaFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE); + channels = mMediaFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT); + break; + } + } + mMediaCodec.start(); + + bufferInfo = new BufferInfo(); + rawInputBuffers = mMediaCodec.getInputBuffers(); + encodedOutputBuffers = mMediaCodec.getOutputBuffers(); + Log.i(TAG, "--channel=" + channels + "--sampleRate=" + inSampleRate); + + mp3Lame = new Mp3LameBuilder() + .setInSampleRate(inSampleRate) + .setOutChannels(channels) + .setOutBitrate(128) + .setOutSampleRate(inSampleRate) + .setQuality(9) + .setVbrMode(Mp3LameBuilder.VbrMode.VBR_MTRH) + .build(); + + } catch (IOException e) { + e.printStackTrace(); + } + } + + private void readSampleData() { + boolean rawInputEOS = false; + + while (!rawInputEOS) { + for (int i = 0; i < rawInputBuffers.length; i++) { + int inputBufIndex = mMediaCodec.dequeueInputBuffer(-1); + if (inputBufIndex >= 0) { + ByteBuffer buffer = rawInputBuffers[inputBufIndex]; + int sampleSize = mediaExtractor.readSampleData(buffer, 0); + long presentationTimeUs = 0; + if (sampleSize < 0) { + rawInputEOS = true; + sampleSize = 0; + } else { + readSize += sampleSize; + presentationTimeUs = mediaExtractor.getSampleTime(); + } + mMediaCodec.queueInputBuffer(inputBufIndex, 0, + sampleSize, presentationTimeUs, + rawInputEOS ? MediaCodec.BUFFER_FLAG_END_OF_STREAM : 0); + if (!rawInputEOS) { + mediaExtractor.advance(); + } else { + break; + } + } else { + Log.e(TAG, "wrong inputBufIndex=" + inputBufIndex); + } + } + } + } + + private void encodeToMp3(BufferDecoded buffer) { + + if (buffer == null || buffer.pts == lastPts) { + return; + } + lastPts = buffer.pts; + + int bufferLength = buffer.pcm.length / 2; + if (mp3buf == null) { + mp3buf = new byte[(int) (bufferLength * 1.25 + 7200)]; + } + if (bufferLength > 0) { + int bytesEncoded; + if (channels == 2) { + bytesEncoded = mp3Lame.encodeBufferInterLeaved(buffer.pcm, bufferLength, mp3buf); + }else { + bytesEncoded = mp3Lame.encode(buffer.leftBuffer, buffer.leftBuffer, bufferLength, mp3buf); + } + Log.d(TAG, "mp3Lame encodeSize=" + bytesEncoded); + + if (bytesEncoded > 0) { + BufferEncoded be = new BufferEncoded(); + be.buffer = mp3buf; + be.length = bytesEncoded; + try { + writeQueue.put(be); + } catch (InterruptedException e) { + e.printStackTrace(); + } + encodeSize += bytesEncoded; + } + } + } + +} diff --git a/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Lame.java b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Lame.java new file mode 100644 index 0000000..4dd3730 --- /dev/null +++ b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3Lame.java @@ -0,0 +1,71 @@ +package com.frank.ffmpeg.mp3; + +import com.frank.ffmpeg.AudioPlayer; + +public class Mp3Lame { + + public Mp3Lame() { + AudioPlayer.lameInitDefault(); + } + + Mp3Lame(Mp3LameBuilder builder) { + initialize(builder); + } + + private void initialize(Mp3LameBuilder builder) { + AudioPlayer.lameInit(builder.inSampleRate, builder.outChannel, builder.outSampleRate, + builder.outBitrate, builder.scaleInput, getIntForMode(builder.mode), getIntForVbrMode(builder.vbrMode), builder.quality, builder.vbrQuality, builder.abrMeanBitrate, + builder.lowPassFreq, builder.highPassFreq, builder.id3tagTitle, builder.id3tagArtist, + builder.id3tagAlbum, builder.id3tagYear, builder.id3tagComment); + } + + public int encode(short[] buffer_l, short[] buffer_r, + int samples, byte[] mp3buf) { + + return AudioPlayer.lameEncode(buffer_l, buffer_r, samples, mp3buf); + } + + int encodeBufferInterLeaved(short[] pcm, int samples, + byte[] mp3buf) { + return AudioPlayer.encodeBufferInterleaved(pcm, samples, mp3buf); + } + + public int flush(byte[] mp3buf) { + return AudioPlayer.lameFlush(mp3buf); + } + + public void close() { + AudioPlayer.lameClose(); + } + + private static int getIntForMode(Mp3LameBuilder.Mode mode) { + switch (mode) { + case STEREO: + return 0; + case JSTEREO: + return 1; + case MONO: + return 3; + case DEFAULT: + return 4; + } + return -1; + } + + private static int getIntForVbrMode(Mp3LameBuilder.VbrMode mode) { + switch (mode) { + case VBR_OFF: + return 0; + case VBR_RH: + return 2; + case VBR_ABR: + return 3; + case VBR_MTRH: + return 4; + case VBR_DEFAUT: + return 6; + } + return -1; + } + +} diff --git a/app/src/main/java/com/frank/ffmpeg/mp3/Mp3LameBuilder.java b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3LameBuilder.java new file mode 100644 index 0000000..5ff5ef2 --- /dev/null +++ b/app/src/main/java/com/frank/ffmpeg/mp3/Mp3LameBuilder.java @@ -0,0 +1,147 @@ +package com.frank.ffmpeg.mp3; + +public class Mp3LameBuilder { + + + public enum Mode { + STEREO, JSTEREO, MONO, DEFAULT + } + + public enum VbrMode { + VBR_OFF, VBR_RH, VBR_MTRH, VBR_ABR, VBR_DEFAUT + } + + + int inSampleRate; + int outSampleRate; + int outBitrate; + int outChannel; + public int quality; + int vbrQuality; + int abrMeanBitrate; + int lowPassFreq; + int highPassFreq; + float scaleInput; + Mode mode; + VbrMode vbrMode; + + String id3tagTitle; + String id3tagArtist; + String id3tagAlbum; + String id3tagComment; + String id3tagYear; + + Mp3LameBuilder() { + + this.id3tagTitle = null; + this.id3tagAlbum = null; + this.id3tagArtist = null; + this.id3tagComment = null; + this.id3tagYear = null; + + this.inSampleRate = 44100; + this.outSampleRate = 0; + this.outChannel = 2; + this.outBitrate = 128; + this.scaleInput = 1; + + this.quality = 5; + this.mode = Mode.DEFAULT; + this.vbrMode = VbrMode.VBR_OFF; + this.vbrQuality = 5; + this.abrMeanBitrate = 128; + + this.lowPassFreq = 0; + this.highPassFreq = 0; + } + + Mp3LameBuilder setQuality(int quality) { + this.quality = quality; + return this; + } + + Mp3LameBuilder setInSampleRate(int inSampleRate) { + this.inSampleRate = inSampleRate; + return this; + } + + Mp3LameBuilder setOutSampleRate(int outSampleRate) { + this.outSampleRate = outSampleRate; + return this; + } + + Mp3LameBuilder setOutBitrate(int bitrate) { + this.outBitrate = bitrate; + return this; + } + + Mp3LameBuilder setOutChannels(int channels) { + this.outChannel = channels; + return this; + } + + Mp3LameBuilder setId3tagTitle(String title) { + this.id3tagTitle = title; + return this; + } + + Mp3LameBuilder setId3tagArtist(String artist) { + this.id3tagArtist = artist; + return this; + } + + Mp3LameBuilder setId3tagAlbum(String album) { + this.id3tagAlbum = album; + return this; + } + + Mp3LameBuilder setId3tagComment(String comment) { + this.id3tagComment = comment; + return this; + } + + Mp3LameBuilder setId3tagYear(String year) { + this.id3tagYear = year; + return this; + } + + Mp3LameBuilder setScaleInput(float scaleAmount) { + this.scaleInput = scaleAmount; + return this; + } + + Mp3LameBuilder setMode(Mode mode) { + this.mode = mode; + return this; + } + + Mp3LameBuilder setVbrMode(VbrMode mode) { + this.vbrMode = mode; + return this; + } + + Mp3LameBuilder setVbrQuality(int quality) { + this.vbrQuality = quality; + return this; + } + + Mp3LameBuilder setAbrMeanBitrate(int bitrate) { + this.abrMeanBitrate = bitrate; + return this; + } + + Mp3LameBuilder setLowpassFreqency(int freq) { + this.lowPassFreq = freq; + return this; + } + + Mp3LameBuilder setHighpassFreqency(int freq) { + this.highPassFreq = freq; + return this; + } + + Mp3Lame build() { + return new Mp3Lame(this); + } + +} diff --git a/build.gradle b/build.gradle index f209f27..0366ab0 100644 --- a/build.gradle +++ b/build.gradle @@ -9,7 +9,7 @@ buildscript { } } dependencies { - classpath 'com.android.tools.build:gradle:3.0.1' + classpath 'com.android.tools.build:gradle:3.2.0' // NOTE: Do not place your application dependencies here; they belong // in the individual module build.gradle files diff --git a/gradle/wrapper/gradle-wrapper.properties b/gradle/wrapper/gradle-wrapper.properties index d5e16da..4ca64ca 100644 --- a/gradle/wrapper/gradle-wrapper.properties +++ b/gradle/wrapper/gradle-wrapper.properties @@ -3,4 +3,4 @@ distributionBase=GRADLE_USER_HOME distributionPath=wrapper/dists zipStoreBase=GRADLE_USER_HOME zipStorePath=wrapper/dists -distributionUrl=https\://services.gradle.org/distributions/gradle-4.1-all.zip +distributionUrl=https\://services.gradle.org/distributions/gradle-4.6-all.zip