ref: d5e2fd9028687dfc18c0ac1b249ccec38ab075f6
dir: /sys/src/cmd/audio/mp3enc/lame.c/
/* -*- mode: C; mode: fold -*- */ /* * LAME MP3 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: lame.c,v 1.100 2001/03/25 23:14:45 markt Exp $ */ #ifdef HAVE_CONFIG_H # include <config.h> #endif #include <assert.h> #include "lame-analysis.h" #include "lame.h" #include "util.h" #include "bitstream.h" #include "version.h" #include "tables.h" #include "quantize_pvt.h" #include "VbrTag.h" #if defined(__FreeBSD__) && !defined(__alpha__) #include <floatingpoint.h> #endif #ifdef __riscos__ #include "asmstuff.h" #endif #ifdef WITH_DMALLOC #include <dmalloc.h> #endif static void lame_init_params_ppflt_lowpass(FLOAT8 amp_lowpass[32], FLOAT lowpass1, FLOAT lowpass2, int *lowpass_band, int *minband, int *maxband) { int band; FLOAT8 freq; for (band = 0; band <= 31; band++) { freq = band / 31.0; amp_lowpass[band] = 1; /* this band and above will be zeroed: */ if (freq >= lowpass2) { *lowpass_band = Min(*lowpass_band, band); amp_lowpass[band] = 0; } if (lowpass1 < freq && freq < lowpass2) { *minband = Min(*minband, band); *maxband = Max(*maxband, band); amp_lowpass[band] = cos((PI / 2) * (lowpass1 - freq) / (lowpass2 - lowpass1)); } /* * DEBUGF("lowpass band=%i amp=%f \n", * band, gfc->amp_lowpass[band]); */ } } /* lame_init_params_ppflt */ /*}}}*/ /* static void lame_init_params_ppflt (lame_internal_flags *gfc) *//*{{{ */ static void lame_init_params_ppflt(lame_global_flags * gfp) { lame_internal_flags *gfc = gfp->internal_flags; /**/ /* compute info needed for polyphase filter (filter type==0, default) */ /**/ int band, maxband, minband; FLOAT8 freq; if (gfc->lowpass1 > 0) { minband = 999; maxband = -1; lame_init_params_ppflt_lowpass(gfc->amp_lowpass, gfc->lowpass1, gfc->lowpass2, &gfc->lowpass_band, &minband, &maxband); /* compute the *actual* transition band implemented by * the polyphase filter */ if (minband == 999) { gfc->lowpass1 = (gfc->lowpass_band - .75) / 31.0; } else { gfc->lowpass1 = (minband - .75) / 31.0; } gfc->lowpass2 = gfc->lowpass_band / 31.0; gfc->lowpass_start_band = minband; gfc->lowpass_end_band = maxband; /* as the lowpass may have changed above * calculate the amplification here again */ for (band = minband; band <= maxband; band++) { freq = band / 31.0; gfc->amp_lowpass[band] = cos((PI / 2) * (gfc->lowpass1 - freq) / (gfc->lowpass2 - gfc->lowpass1)); } } else { gfc->lowpass_start_band = 0; gfc->lowpass_end_band = -1; /* do not to run into for-loops */ } /* make sure highpass filter is within 90% of what the effective * highpass frequency will be */ if (gfc->highpass2 > 0) { if (gfc->highpass2 < .9 * (.75 / 31.0)) { gfc->highpass1 = 0; gfc->highpass2 = 0; MSGF(gfc, "Warning: highpass filter disabled. " "highpass frequency too small\n"); } } if (gfc->highpass2 > 0) { minband = 999; maxband = -1; for (band = 0; band <= 31; band++) { freq = band / 31.0; gfc->amp_highpass[band] = 1; /* this band and below will be zereod */ if (freq <= gfc->highpass1) { gfc->highpass_band = Max(gfc->highpass_band, band); gfc->amp_highpass[band] = 0; } if (gfc->highpass1 < freq && freq < gfc->highpass2) { minband = Min(minband, band); maxband = Max(maxband, band); gfc->amp_highpass[band] = cos((PI / 2) * (gfc->highpass2 - freq) / (gfc->highpass2 - gfc->highpass1)); } /* DEBUGF("highpass band=%i amp=%f \n", band, gfc->amp_highpass[band]); */ } /* compute the *actual* transition band implemented by * the polyphase filter */ gfc->highpass1 = gfc->highpass_band / 31.0; if (maxband == -1) { gfc->highpass2 = (gfc->highpass_band + .75) / 31.0; } else { gfc->highpass2 = (maxband + .75) / 31.0; } gfc->highpass_start_band = minband; gfc->highpass_end_band = maxband; /* as the highpass may have changed above * calculate the amplification here again */ for (band = minband; band <= maxband; band++) { freq = band / 31.0; gfc->amp_highpass[band] = cos((PI / 2) * (gfc->highpass2 - freq) / (gfc->highpass2 - gfc->highpass1)); } } else { gfc->highpass_start_band = 0; gfc->highpass_end_band = -1; /* do not to run into for-loops */ } /* DEBUGF("lowpass band with amp=0: %i \n",gfc->lowpass_band); DEBUGF("highpass band with amp=0: %i \n",gfc->highpass_band); DEBUGF("lowpass band start: %i \n",gfc->lowpass_start_band); DEBUGF("lowpass band end: %i \n",gfc->lowpass_end_band); DEBUGF("highpass band start: %i \n",gfc->highpass_start_band); DEBUGF("highpass band end: %i \n",gfc->highpass_end_band); */ } /*}}}*/ static void optimum_bandwidth(double *const lowerlimit, double *const upperlimit, const unsigned bitrate, const int samplefreq, const double channels, lame_global_flags * gfp) { /* * Input: * bitrate total bitrate in bps * samplefreq output sampling frequency in Hz * channels 1 for mono, 2+epsilon for MS stereo, 3 for LR stereo * epsilon is the percentage of LR frames for typical audio * (I use 'Fade to Gray' by Metallica) * * Output: * lowerlimit: best lowpass frequency limit for input filter in Hz * upperlimit: best highpass frequency limit for input filter in Hz */ double f_low; double f_high; double br; assert(bitrate >= 8000 && bitrate <= 320000); assert(samplefreq >= 8000 && samplefreq <= 48000); assert(channels == 1 || (channels >= 2 && channels <= 3)); if (samplefreq >= 32000) br = bitrate - (channels == 1 ? (17 + 4) * 8 : (32 + 4) * 8) * samplefreq / 1152; else br = bitrate - (channels == 1 ? (9 + 4) * 8 : (17 + 4) * 8) * samplefreq / 576; if (channels >= 2.) br /= 1.75 + 0.25 * (channels - 2.); // MS needs 1.75x mono, LR needs 2.00x mono (experimental data of a lot of albums) br *= 0.5; // the sine and cosine term must share the bitrate /* * So, now we have the bitrate for every spectral line. * Let's look at the current settings: * * Bitrate limit bits/line * 8 kbps 0.34 kHz 4.76 * 16 kbps 1.9 kHz 2.06 * 24 kbps 2.8 kHz 2.21 * 32 kbps 3.85 kHz 2.14 * 40 kbps 5.1 kHz 2.06 * 48 kbps 5.6 kHz 2.21 * 56 kbps 7.0 kHz 2.10 * 64 kbps 7.7 kHz 2.14 * 80 kbps 10.1 kHz 2.08 * 96 kbps 11.2 kHz 2.24 * 112 kbps 14.0 kHz 2.12 * 128 kbps 15.4 kHz 2.17 * 160 kbps 18.2 kHz 2.05 * 192 kbps 21.1 kHz 2.14 * 224 kbps 22.0 kHz 2.41 * 256 kbps 22.0 kHz 2.78 * * What can we see? * Value for 8 kbps is nonsense (although 8 kbps and stereo is nonsense) * Values are between 2.05 and 2.24 for 16...192 kbps * Some bitrate lack the following bitrates have: 16, 40, 80, 160 kbps * A lot of bits per spectral line have: 24, 48, 96 kbps * * What I propose? * A slightly with the bitrate increasing bits/line function. It is * better to decrease NMR for low bitrates to get a little bit more * bandwidth. So we have a better trade off between twickling and * muffled sound. */ f_low = br / log10(br * 4.425e-3); // Tests with 8, 16, 32, 64, 112 and 160 kbps /* * What we get now? * * Bitrate limit bits/line difference * 8 kbps (8) 1.89 kHz 0.86 +1.6 kHz * 16 kbps (8) 3.16 kHz 1.24 +1.2 kHz * 32 kbps(16) 5.08 kHz 1.54 +1.2 kHz * 56 kbps(22) 7.88 kHz 1.80 +0.9 kHz * 64 kbps(22) 8.83 kHz 1.86 +1.1 kHz * 112 kbps(32) 14.02 kHz 2.12 0.0 kHz * 112 kbps(44) 13.70 kHz 2.11 -0.3 kHz * 128 kbps 15.40 kHz 2.17 0.0 kHz * 160 kbps 16.80 kHz 2.22 -1.4 kHz * 192 kbps 19.66 kHz 2.30 -1.4 kHz * 256 kbps 22.05 kHz 2.78 0.0 kHz */ #if 0 /* * Beginning at 128 kbps/jstereo, we can use the following additional * strategy: * * For every increase of f_low in a way that the ATH(f_low) * increases by 4 dB we force an additional NMR of 1.25 dB. * These are the setting of the VBR quality selecting scheme * for V <= 4. */ { double br_sw = (128000 - (32 + 4) * 8 * 44100 / 1152) / 1.75 * 0.5; double f_low_sw = br_sw / log10(br_sw * 4.425e-3); // printf ("br_sw=%f f_low_sw=%f\n", br_sw, f_low_sw ); // printf ("br =%f f_low =%f\n", br , f_low ); // fflush (stdout); while (f_low > f_low_sw) { double dATH = ATHformula(f_low, gfp) - ATHformula(f_low_sw, gfp); // [dB] double dNMR = br / f_low - br_sw / f_low_sw; // bit // printf ("br =%f f_low =%f\n", br , f_low ); // printf ("dATH =%f dNMR =%f\n", dATH , dNMR ); // fflush (stdout); if (dATH / 4.0 < dNMR * 6.0206 / 1.25) // 1 bit = 6.0206... dB break; f_low -= 25.; } } #endif /* * Now we try to choose a good high pass filtering frequency. * This value is currently not used. * For fu < 16 kHz: sqrt(fu*fl) = 560 Hz * For fu = 18 kHz: no high pass filtering * This gives: * * 2 kHz => 160 Hz * 3 kHz => 107 Hz * 4 kHz => 80 Hz * 8 kHz => 40 Hz * 16 kHz => 20 Hz * 17 kHz => 10 Hz * 18 kHz => 0 Hz * * These are ad hoc values and these can be optimized if a high pass is available. */ if (f_low <= 16000) f_high = 16000. * 20. / f_low; else if (f_low <= 18000) f_high = 180. - 0.01 * f_low; else f_high = 0.; /* * When we sometimes have a good highpass filter, we can add the highpass * frequency to the lowpass frequency */ if (lowerlimit != NULL) *lowerlimit = f_low /* + f_high */ ; if (upperlimit != NULL) *upperlimit = f_high; /* * Now the weak points: * * - the formula f_low=br/log10(br*4.425e-3) is an ad hoc formula * (but has a physical background and is easy to tune) * - the switch to the ATH based bandwidth selecting is the ad hoc * value of 128 kbps */ } static int optimum_samplefreq(int lowpassfreq, int input_samplefreq) { /* * Rules: * * - output sample frequency should NOT be decreased by more than 3% if lowpass allows this * - if possible, sfb21 should NOT be used * * Problem: Switches to 32 kHz at 112 kbps */ if (input_samplefreq <= 8000 * 1.03 || lowpassfreq <= 3622) return 8000; if (input_samplefreq <= 11025 * 1.03 || lowpassfreq <= 4991) return 11025; if (input_samplefreq <= 12000 * 1.03 || lowpassfreq <= 5620) return 12000; if (input_samplefreq <= 16000 * 1.03 || lowpassfreq <= 7244) return 16000; if (input_samplefreq <= 22050 * 1.03 || lowpassfreq <= 9982) return 22050; if (input_samplefreq <= 24000 * 1.03 || lowpassfreq <= 11240) return 24000; if (input_samplefreq <= 32000 * 1.03 || lowpassfreq <= 15264) return 32000; if (input_samplefreq <= 44100 * 1.03) return 44100; return 48000; } /* set internal feature flags. USER should not access these since * some combinations will produce strange results */ void lame_init_qval(lame_global_flags * gfp) { lame_internal_flags *gfc = gfp->internal_flags; switch (gfp->quality) { case 9: /* no psymodel, no noise shaping */ gfc->filter_type = 0; gfc->psymodel = 0; gfc->quantization = 0; gfc->noise_shaping = 0; gfc->noise_shaping_amp = 0; gfc->noise_shaping_stop = 0; gfc->use_best_huffman = 0; break; case 8: gfp->quality = 7; case 7: /* use psymodel (for short block and m/s switching), but no noise shapping */ gfc->filter_type = 0; gfc->psymodel = 1; gfc->quantization = 0; gfc->noise_shaping = 0; gfc->noise_shaping_amp = 0; gfc->noise_shaping_stop = 0; gfc->use_best_huffman = 0; break; case 6: gfp->quality = 5; case 5: /* the default */ gfc->filter_type = 0; gfc->psymodel = 1; gfc->quantization = 0; gfc->noise_shaping = 1; /**/ gfc->noise_shaping_amp = 0; gfc->noise_shaping_stop = 0; gfc->use_best_huffman = 0; break; case 4: gfp->quality = 3; case 3: gfc->filter_type = 0; gfc->psymodel = 1; gfc->quantization = 1; gfc->noise_shaping = 1; gfc->noise_shaping_amp = 0; gfc->noise_shaping_stop = 0; gfc->use_best_huffman = 1; break; case 2: gfc->filter_type = 0; gfc->psymodel = 1; gfc->quantization = 1; gfc->noise_shaping = 1; gfc->noise_shaping_amp = 1; gfc->noise_shaping_stop = 1; gfc->use_best_huffman = 1; break; case 1: gfc->filter_type = 0; gfc->psymodel = 1; gfc->quantization = 1; gfc->noise_shaping = 1; gfc->noise_shaping_amp = 2; gfc->noise_shaping_stop = 1; gfc->use_best_huffman = 1; break; case 0: /* 0..1 quality */ gfc->filter_type = 0; /* 1 not yet coded */ gfc->psymodel = 1; gfc->quantization = 1; gfc->noise_shaping = 1; /* 2=usually lowers quality */ gfc->noise_shaping_amp = 2; gfc->noise_shaping_stop = 1; gfc->use_best_huffman = 1; /* 2 not yet coded */ } /* modifications to the above rules: */ /* -Z option enables scalefactor_scale: */ if (gfp->experimentalZ) { gfc->noise_shaping = 2; } if (gfp->exp_nspsytune & 1) { if (gfp->quality <= 2) gfc->noise_shaping = 2; /* use scalefac_scale */ } } /* int lame_init_params (lame_global_flags *gfp) *//*{{{ */ /* * initialize internal params based on data in gf * (globalflags struct filled in by calling program) * * OUTLINE: * * We first have some complex code to determine bitrate, * output samplerate and mode. It is complicated by the fact * that we allow the user to set some or all of these parameters, * and need to determine best possible values for the rest of them: * * 1. set some CPU related flags * 2. check if we are mono->mono, stereo->mono or stereo->stereo * 3. compute bitrate and output samplerate: * user may have set compression ratio * user may have set a bitrate * user may have set a output samplerate * 4. set some options which depend on output samplerate * 5. compute the actual compression ratio * 6. set mode based on compression ratio * * The remaining code is much simpler - it just sets options * based on the mode & compression ratio: * * set allow_diff_short based on mode * select lowpass filter based on compression ratio & mode * set the bitrate index, and min/max bitrates for VBR modes * disable VBR tag if it is not appropriate * initialize the bitstream * initialize scalefac_band data * set sideinfo_len (based on channels, CRC, out_samplerate) * write an id3v2 tag into the bitstream * write VBR tag into the bitstream * set mpeg1/2 flag * estimate the number of frames (based on a lot of data) * * now we set more flags: * nspsytune: * see code * VBR modes * see code * CBR/ABR * see code * * Finally, we set the algorithm flags based on the gfp->quality value * lame_init_qval(gfp); * */ int lame_init_params(lame_global_flags * const gfp) { int i; int j; lame_internal_flags *gfc = gfp->internal_flags; gfc->gfp = gfp; gfc->Class_ID = 0; /* report functions */ gfc->report.msgf = gfp->report.msgf; gfc->report.debugf = gfp->report.debugf; gfc->report.errorf = gfp->report.errorf; gfc->CPU_features.i387 = has_i387(); gfc->CPU_features.AMD_3DNow = has_3DNow(); gfc->CPU_features.MMX = has_MMX(); gfc->CPU_features.SIMD = has_SIMD(); gfc->CPU_features.SIMD2 = has_SIMD2(); if (NULL == gfc->ATH) gfc->ATH = calloc(1, sizeof(ATH_t)); if (NULL == gfc->ATH) return -2; // maybe error codes should be enumerated in lame.h ?? #ifdef KLEMM_44 /* Select the fastest functions for this CPU */ init_scalar_functions(gfc); #endif gfc->channels_in = gfp->num_channels; if (gfc->channels_in == 1) gfp->mode = MONO; gfc->channels_out = (gfp->mode == MONO) ? 1 : 2; gfc->mode_ext = MPG_MD_LR_LR; if (gfp->mode == MONO) gfp->force_ms = 0; // don't allow forced mid/side stereo for mono output if (gfp->VBR != vbr_off) { gfp->free_format = 0; /* VBR can't be mixed with free format */ } if (gfp->VBR == vbr_off && gfp->brate == 0) { /* no bitrate or compression ratio specified, use a compression ratio of 11.025 */ if (gfp->compression_ratio == 0) gfp->compression_ratio = 11.025; /* rate to compress a CD down to exactly 128000 bps */ } if (gfp->VBR == vbr_off && gfp->brate == 0) { /* no bitrate or compression ratio specified, use 11.025 */ if (gfp->compression_ratio == 0) gfp->compression_ratio = 11.025; /* rate to compress a CD down to exactly 128000 bps */ } /* find bitrate if user specify a compression ratio */ if (gfp->VBR == vbr_off && gfp->compression_ratio > 0) { if (gfp->out_samplerate == 0) gfp->out_samplerate = map2MP3Frequency(0.97 * gfp->in_samplerate); /* round up with a margin of 3% */ /* choose a bitrate for the output samplerate which achieves * specified compression ratio */ gfp->brate = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->compression_ratio); /* we need the version for the bitrate table look up */ gfc->samplerate_index = SmpFrqIndex(gfp->out_samplerate, &gfp->version); if (!gfp->free_format) /* for non Free Format find the nearest allowed bitrate */ gfp->brate = FindNearestBitrate(gfp->brate, gfp->version, gfp->out_samplerate); } /* * at 160 kbps (MPEG-2/2.5)/ 320 kbps (MPEG-1), only Free format * or CBR are possible, no VBR */ if (gfp->VBR != vbr_off && gfp->brate >= 320) gfp->VBR = vbr_off; if (gfp->out_samplerate == 0) { /* if output sample frequency is not given, find a useful value */ gfp->out_samplerate = map2MP3Frequency(0.97 * gfp->in_samplerate); /* check if user specified bitrate requires downsampling, if compression */ /* ratio is > 13, choose a new samplerate to get the ratio down to about 10 */ if (gfp->VBR == vbr_off && gfp->brate > 0) { gfp->compression_ratio = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->brate); if (gfp->compression_ratio > 13.) gfp->out_samplerate = map2MP3Frequency((10. * 1.e3 * gfp->brate) / (16 * gfc->channels_out)); } if (gfp->VBR == vbr_abr) { gfp->compression_ratio = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->VBR_mean_bitrate_kbps); if (gfp->compression_ratio > 13.) gfp->out_samplerate = map2MP3Frequency((10. * 1.e3 * gfp->VBR_mean_bitrate_kbps) / (16 * gfc->channels_out)); } } if (gfp->ogg) { gfp->framesize = 1024; gfp->encoder_delay = ENCDELAY; gfc->coding = coding_Ogg_Vorbis; } else { gfc->mode_gr = gfp->out_samplerate <= 24000 ? 1 : 2; // Number of granules per frame gfp->framesize = 576 * gfc->mode_gr; gfp->encoder_delay = ENCDELAY; gfc->coding = coding_MPEG_Layer_3; } gfc->frame_size = gfp->framesize; gfc->resample_ratio = (double) gfp->in_samplerate / gfp->out_samplerate; /* * sample freq bitrate compression ratio * [kHz] [kbps/channel] for 16 bit input * 44.1 56 12.6 * 44.1 64 11.025 * 44.1 80 8.82 * 22.05 24 14.7 * 22.05 32 11.025 * 22.05 40 8.82 * 16 16 16.0 * 16 24 10.667 * */ /* * For VBR, take a guess at the compression_ratio. * For example: * * VBR_q compression like * - 4.4 320 kbps/44 kHz * 0...1 5.5 256 kbps/44 kHz * 2 7.3 192 kbps/44 kHz * 4 8.8 160 kbps/44 kHz * 6 11 128 kbps/44 kHz * 9 14.7 96 kbps * * for lower bitrates, downsample with --resample */ switch (gfp->VBR) { case vbr_mt: case vbr_rh: case vbr_mtrh: { FLOAT8 cmp[] = { 5, 6, 7, 8, 9, 10, 11, 12, 13, 14 }; gfp->compression_ratio = cmp[gfp->VBR_q]; } break; case vbr_abr: gfp->compression_ratio = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->VBR_mean_bitrate_kbps); break; default: gfp->compression_ratio = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->brate); break; } /* mode = -1 (not set by user) or * mode = MONO (because of only 1 input channel). * If mode has been set, then select between STEREO or J-STEREO * At higher quality (lower compression) use STEREO instead of J-STEREO. * (unless the user explicitly specified a mode) * * The threshold to switch to STEREO is: * 48 kHz: 171 kbps (used at 192+) * 44.1 kHz: 160 kbps (used at 160+) * 32 kHz: 119 kbps (used at 128+) * * Note, that for 32 kHz/128 kbps J-STEREO FM recordings sound much * better than STEREO, so I'm not so very happy with that. * fs < 32 kHz I have not tested. */ if (gfp->mode == NOT_SET) { if (gfp->compression_ratio < 8) gfp->mode = STEREO; else gfp->mode = JOINT_STEREO; } /* KLEMM's jstereo with ms threshold adjusted via compression ratio */ if (gfp->mode_automs) { if (gfp->mode != MONO && gfp->compression_ratio < 6.6) gfp->mode = STEREO; } if (gfp->allow_diff_short == -1) { if (gfp->mode == STEREO) gfp->allow_diff_short = 1; } /**/ /* if a filter has not been enabled, see if we should add one: */ /**/ if (gfp->lowpassfreq == 0) { double lowpass; double highpass; double channels; switch (gfp->mode) { case MONO: channels = 1.; break; case JOINT_STEREO: channels = 2. + 0.00; break; case DUAL_CHANNEL: case STEREO: channels = 3.; break; default: channels = 1.; // just to make data flow analysis happy :-) assert(0); break; } optimum_bandwidth(&lowpass, &highpass, gfp->out_samplerate * 16 * gfc->channels_out / gfp->compression_ratio, gfp->out_samplerate, channels, gfp); if (lowpass < 0.5 * gfp->out_samplerate) { //MSGF(gfc,"Lowpass @ %7.1f Hz\n", lowpass); gfc->lowpass1 = gfc->lowpass2 = lowpass / (0.5 * gfp->out_samplerate); } if (0 && gfp->out_samplerate != optimum_samplefreq(lowpass, gfp->in_samplerate)) { MSGF(gfc, "I would suggest to use %u Hz instead of %u Hz sample frequency\n", optimum_samplefreq(lowpass, gfp->in_samplerate), gfp->out_samplerate); } fflush(stderr); } /* apply user driven high pass filter */ if (gfp->highpassfreq > 0) { gfc->highpass1 = 2. * gfp->highpassfreq / gfp->out_samplerate; /* will always be >=0 */ if (gfp->highpasswidth >= 0) gfc->highpass2 = 2. * (gfp->highpassfreq + gfp->highpasswidth) / gfp->out_samplerate; else /* 0% above on default */ gfc->highpass2 = (1 + 0.00) * 2. * gfp->highpassfreq / gfp->out_samplerate; } /* apply user driven low pass filter */ if (gfp->lowpassfreq > 0) { gfc->lowpass2 = 2. * gfp->lowpassfreq / gfp->out_samplerate; /* will always be >=0 */ if (gfp->lowpasswidth >= 0) { gfc->lowpass1 = 2. * (gfp->lowpassfreq - gfp->lowpasswidth) / gfp->out_samplerate; if (gfc->lowpass1 < 0) /* has to be >= 0 */ gfc->lowpass1 = 0; } else { /* 0% below on default */ gfc->lowpass1 = (1 - 0.00) * 2. * gfp->lowpassfreq / gfp->out_samplerate; } } /**/ /* compute info needed for polyphase filter (filter type==0, default) */ /**/ lame_init_params_ppflt(gfp); /* * compute info needed for FIR filter (filter_type==1) */ /* not yet coded */ /* * samplerate and bitrate index */ gfc->samplerate_index = SmpFrqIndex(gfp->out_samplerate, &gfp->version); if (gfc->samplerate_index < 0) return -1; if (gfp->VBR == vbr_off) { if (gfp->free_format) gfc->bitrate_index = 0; else { gfc->bitrate_index = BitrateIndex(gfp->brate, gfp->version, gfp->out_samplerate); if (gfc->bitrate_index < 0) return -1; } } else { /* choose a min/max bitrate for VBR */ /* if the user didn't specify VBR_max_bitrate: */ gfc->VBR_min_bitrate = 1; /* default: allow 8 kbps (MPEG-2) or 32 kbps (MPEG-1) */ gfc->VBR_max_bitrate = 14; /* default: allow 160 kbps (MPEG-2) or 320 kbps (MPEG-1) */ if (gfp->VBR_min_bitrate_kbps) if ( (gfc->VBR_min_bitrate = BitrateIndex(gfp->VBR_min_bitrate_kbps, gfp->version, gfp->out_samplerate)) < 0) return -1; if (gfp->VBR_max_bitrate_kbps) if ( (gfc->VBR_max_bitrate = BitrateIndex(gfp->VBR_max_bitrate_kbps, gfp->version, gfp->out_samplerate)) < 0) return -1; gfp->VBR_min_bitrate_kbps = bitrate_table[gfp->version][gfc->VBR_min_bitrate]; gfp->VBR_max_bitrate_kbps = bitrate_table[gfp->version][gfc->VBR_max_bitrate]; gfp->VBR_mean_bitrate_kbps = Min(bitrate_table[gfp->version][gfc->VBR_max_bitrate], gfp->VBR_mean_bitrate_kbps); gfp->VBR_mean_bitrate_kbps = Max(bitrate_table[gfp->version][gfc->VBR_min_bitrate], gfp->VBR_mean_bitrate_kbps); } /* Do not write VBR tag if VBR flag is not specified */ if (gfp->VBR == vbr_off) gfp->bWriteVbrTag = 0; if (gfp->ogg) gfp->bWriteVbrTag = 0; if (gfp->analysis) gfp->bWriteVbrTag = 0; /* some file options not allowed if output is: not specified or stdout */ if (gfc->pinfo != NULL) gfp->bWriteVbrTag = 0; /* disable Xing VBR tag */ init_bit_stream_w(gfc); j = gfc->samplerate_index + (3 * gfp->version) + 6 * (gfp->out_samplerate < 16000); for (i = 0; i < SBMAX_l + 1; i++) gfc->scalefac_band.l[i] = sfBandIndex[j].l[i]; for (i = 0; i < SBMAX_s + 1; i++) gfc->scalefac_band.s[i] = sfBandIndex[j].s[i]; /* determine the mean bitrate for main data */ if (gfp->version == 1) /* MPEG 1 */ gfc->sideinfo_len = (gfc->channels_out == 1) ? 4 + 17 : 4 + 32; else /* MPEG 2 */ gfc->sideinfo_len = (gfc->channels_out == 1) ? 4 + 9 : 4 + 17; if (gfp->error_protection) gfc->sideinfo_len += 2; /* * Write id3v2 tag into the bitstream. * This tag must be before the Xing VBR header. */ if (!gfp->ogg) id3tag_write_v2(gfp); /* Write initial VBR Header to bitstream */ if (gfp->bWriteVbrTag) InitVbrTag(gfp); if (gfp->version == 1) /* 0 indicates use lower sample freqs algorithm */ gfc->is_mpeg1 = 1; /* yes */ else gfc->is_mpeg1 = 0; /* no */ /* estimate total frames. */ gfp->totalframes = 2 + gfp->num_samples / (gfc->resample_ratio * gfp->framesize); gfc->Class_ID = LAME_ID; if (gfp->exp_nspsytune & 1) { int i; gfc->nsPsy.use = 1; gfc->nsPsy.safejoint = (gfp->exp_nspsytune & 2) != 0; for (i = 0; i < 19; i++) gfc->nsPsy.pefirbuf[i] = 700; if (gfp->VBR == vbr_mtrh || gfp->VBR == vbr_mt) { ERRORF(gfc, "\n**** nspsytune doesn't support --vbr-new **** \n\n"); gfp->VBR = vbr_rh; } if (gfp->ATHtype == -1) gfp->ATHtype = 0; gfc->nsPsy.bass = gfc->nsPsy.alto = gfc->nsPsy.treble = 0; i = (gfp->exp_nspsytune >> 2) & 63; if (i >= 32) i -= 64; gfc->nsPsy.bass = pow(10, i / 4.0 / 10.0); i = (gfp->exp_nspsytune >> 8) & 63; if (i >= 32) i -= 64; gfc->nsPsy.alto = pow(10, i / 4.0 / 10.0); i = (gfp->exp_nspsytune >> 14) & 63; if (i >= 32) i -= 64; gfc->nsPsy.treble = pow(10, i / 4.0 / 10.0); } switch (gfp->VBR) { case vbr_mtrh: /* default quality for --vbr-mtrh is 1 */ if (gfp->quality < 0) gfp->quality = 1; /* tonality */ if (gfp->cwlimit <= 0) gfp->cwlimit = 0.454 * gfp->out_samplerate; /* fall through */ case vbr_mt: /* use Gaby's ATH for vbr-mtrh by default */ if (gfp->ATHtype == -1) gfp->ATHtype = 2; /* fall through */ case vbr_rh: /* use Roel's tweaked Gaby-ATH for VBR by default */ if (gfp->ATHtype == -1) gfp->ATHtype = 2; /* automatic ATH adjustment on, VBR modes need it */ gfc->ATH->use_adjust = 1; /* sfb21 extra only with MPEG-1 at higher sampling rates */ gfc->sfb21_extra = (gfp->out_samplerate > 44000); /* VBR needs at least the output of GPSYCHO, * so we have to garantee that by setting a minimum * quality level, actually level 5 does it. * the -v and -V x settings switch the quality to level 2 * you would have to add a -q 5 to reduce the quality * down to level 5 */ if (gfp->quality > 5) gfp->quality = 5; /* default quality setting is 2 */ if (gfp->quality < 0) gfp->quality = 2; /* allow left and right channels to have different block types */ gfp->allow_diff_short = 1; break; default: /* automatic ATH adjustment off, not so important for CBR code */ gfc->ATH->use_adjust = 0; /* use Frank's ATH for CBR/ABR by default */ if (gfp->ATHtype == -1) gfp->ATHtype = 2; /* no sfb21 extra with CBR code */ gfc->sfb21_extra = 0; /* default quality setting for CBR/ABR is 5 */ if (gfp->quality < 0) gfp->quality = 5; break; } /* initialize internal qval settings */ lame_init_qval(gfp); #ifdef KLEMM_44 gfc->mfbuf[0] = (sample_t *) calloc(sizeof(sample_t), MFSIZE); gfc->mfbuf[1] = (sample_t *) calloc(sizeof(sample_t), MFSIZE); gfc->sampfreq_in = unround_samplefrequency(gfp->in_samplerate); gfc->sampfreq_out = gfp->out_samplerate; gfc->resample_in = resample_open(gfc->sampfreq_in, gfc->sampfreq_out, -1 .0 /* Auto */ , 32); #endif return 0; } /*}}}*/ /* void lame_print_config (lame_global_flags *gfp) *//*{{{ */ /* * print_config * * Prints some selected information about the coding parameters via * the macro command MSGF(), which is currently mapped to lame_errorf * (reports via a error function?), which is a printf-like function * for <stderr>. */ void lame_print_config(const lame_global_flags * gfp) { lame_internal_flags *gfc = gfp->internal_flags; double out_samplerate = gfp->out_samplerate; double in_samplerate = gfp->out_samplerate * gfc->resample_ratio; MSGF(gfc, "mp3enc (from lame version %s (%s))\n", get_lame_version(), get_lame_url()); if (gfc->CPU_features.MMX || gfc->CPU_features.AMD_3DNow || gfc->CPU_features.SIMD || gfc->CPU_features.SIMD2) { MSGF(gfc, "CPU features:"); if (gfc->CPU_features.i387) MSGF(gfc, " i387"); if (gfc->CPU_features.MMX) #ifdef MMX_choose_table MSGF(gfc, ", MMX (ASM used)"); #else MSGF(gfc, ", MMX"); #endif if (gfc->CPU_features.AMD_3DNow) MSGF(gfc, ", 3DNow!"); if (gfc->CPU_features.SIMD) MSGF(gfc, ", SIMD"); if (gfc->CPU_features.SIMD2) MSGF(gfc, ", SIMD2"); MSGF(gfc, "\n"); } if (gfp->num_channels == 2 && gfc->channels_out == 1 /* mono */ ) { MSGF (gfc, "Autoconverting from stereo to mono. Setting encoding to mono mode.\n"); } if (gfc->resample_ratio != 1.) { MSGF(gfc, "Resampling: input %g kHz output %g kHz\n", 1.e-3 * in_samplerate, 1.e-3 * out_samplerate); } if (gfc->filter_type == 0) { if (gfc->highpass2 > 0.) MSGF (gfc, "Using polyphase highpass filter, transition band: %5.0f Hz - %5.0f Hz\n", 0.5 * gfc->highpass1 * out_samplerate, 0.5 * gfc->highpass2 * out_samplerate); if (gfc->lowpass1 > 0.) { MSGF (gfc, "Using polyphase lowpass filter, transition band: %5.0f Hz - %5.0f Hz\n", 0.5 * gfc->lowpass1 * out_samplerate, 0.5 * gfc->lowpass2 * out_samplerate); } else { MSGF(gfc, "polyphase lowpass filter disabled\n"); } } else { MSGF(gfc, "polyphase filters disabled\n"); } if (gfp->free_format) { MSGF(gfc, "Warning: many decoders cannot handle free format bitstreams\n"); if (gfp->brate > 320) { MSGF (gfc, "Warning: many decoders cannot handle free format bitrates >320 kbps (see documentation)\n"); } } } /* int lame_encode_frame (lame_global_flags *gfp, sample_t inbuf_l[],sample_t inbuf_r[], char *mp3buf, int mp3buf_size) *//*{{{ */ /* routine to feed exactly one frame (gfp->framesize) worth of data to the encoding engine. All buffering, resampling, etc, handled by calling program. */ int lame_encode_frame(lame_global_flags * gfp, sample_t inbuf_l[], sample_t inbuf_r[], unsigned char *mp3buf, int mp3buf_size) { int ret; if (gfp->ogg) { #ifdef HAVE_VORBIS ret = lame_encode_ogg_frame(gfp, inbuf_l, inbuf_r, mp3buf, mp3buf_size); #else return -5; /* wanna encode ogg without vorbis */ #endif } else { ret = lame_encode_mp3_frame(gfp, inbuf_l, inbuf_r, mp3buf, mp3buf_size); } /* check to see if we underestimated totalframes */ gfp->frameNum++; if (gfp->totalframes < gfp->frameNum) gfp->totalframes = gfp->frameNum; return ret; } /*}}}*/ /* int lame_encode_buffer (lame_global_flags* gfp, short int buffer_l[], short int buffer_r[], int nsamples, char* mp3buf, int mp3buf_size )*//*{{{ */ /* * THE MAIN LAME ENCODING INTERFACE * mt 3/00 * * input pcm data, output (maybe) mp3 frames. * This routine handles all buffering, resampling and filtering for you. * The required mp3buffer_size can be computed from num_samples, * samplerate and encoding rate, but here is a worst case estimate: * * mp3buffer_size in bytes = 1.25*num_samples + 7200 * * return code = number of bytes output in mp3buffer. can be 0 */ int lame_encode_buffer_sample_t(lame_global_flags * gfp, sample_t buffer_l[], sample_t buffer_r[], int nsamples, unsigned char *mp3buf, const int mp3buf_size) { lame_internal_flags *gfc = gfp->internal_flags; int mp3size = 0, ret, i, ch, mf_needed; sample_t *mfbuf[2]; sample_t *in_buffer[2]; if (gfc->Class_ID != LAME_ID) return -3; if (nsamples == 0) return 0; in_buffer[0]=buffer_l; in_buffer[1]=buffer_r; /* some sanity checks */ #if ENCDELAY < MDCTDELAY # error ENCDELAY is less than MDCTDELAY, see encoder.h #endif #if FFTOFFSET > BLKSIZE # error FFTOFFSET is greater than BLKSIZE, see encoder.h #endif mf_needed = BLKSIZE + gfp->framesize - FFTOFFSET; /* amount needed for FFT */ mf_needed = Max(mf_needed, 286 + 576 * (1 + gfc->mode_gr)); /* amount needed for MDCT/filterbank */ assert(MFSIZE >= mf_needed); mfbuf[0] = gfc->mfbuf[0]; mfbuf[1] = gfc->mfbuf[1]; if (gfp->num_channels == 2 && gfc->channels_out == 1) { /* downsample to mono */ for (i = 0; i < nsamples; ++i) { in_buffer[0][i] = 0.5 * ((FLOAT8) in_buffer[0][i] + in_buffer[1][i]); in_buffer[1][i] = 0.0; } } while (nsamples > 0) { int n_in = 0; /* number of input samples processed with fill_buffer */ int n_out = 0; /* number of samples output with fill_buffer */ /* n_in <> n_out if we are resampling */ /* copy in new samples into mfbuf, with resampling & scaling if necessary */ fill_buffer(gfp, mfbuf, in_buffer, nsamples, &n_in, &n_out); /* update in_buffer counters */ nsamples -= n_in; in_buffer[0] += n_in; if (gfc->channels_out == 2) in_buffer[1] += n_in; /* update mfbuf[] counters */ gfc->mf_size += n_out; assert(gfc->mf_size <= MFSIZE); gfc->mf_samples_to_encode += n_out; if (gfc->mf_size >= mf_needed) { /* encode the frame. */ ret = lame_encode_frame(gfp, mfbuf[0], mfbuf[1], mp3buf, mp3buf_size); if (ret < 0) goto retr; mp3buf += ret; mp3size += ret; /* shift out old samples */ gfc->mf_size -= gfp->framesize; gfc->mf_samples_to_encode -= gfp->framesize; for (ch = 0; ch < gfc->channels_out; ch++) for (i = 0; i < gfc->mf_size; i++) mfbuf[ch][i] = mfbuf[ch][i + gfp->framesize]; } } assert(nsamples == 0); ret = mp3size; retr: return ret; } int lame_encode_buffer(lame_global_flags * gfp, const short int buffer_l[], const short int buffer_r[], int nsamples, unsigned char *mp3buf, const int mp3buf_size) { lame_internal_flags *gfc = gfp->internal_flags; int ret, i; sample_t *in_buffer[2]; if (gfc->Class_ID != LAME_ID) return -3; if (nsamples == 0) return 0; in_buffer[0] = calloc(sizeof(sample_t), nsamples); in_buffer[1] = calloc(sizeof(sample_t), nsamples); if (in_buffer[0] == NULL || in_buffer[1] == NULL) { ERRORF(gfc, "Error: can't allocate in_buffer buffer\n"); return -2; } /* make a copy of input buffer, changing type to sample_t */ for (i = 0; i < nsamples; i++) { in_buffer[0][i] = buffer_l[i]; in_buffer[1][i] = buffer_r[i]; } ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1], nsamples, mp3buf, mp3buf_size); free(in_buffer[0]); free(in_buffer[1]); return ret; } int lame_encode_buffer_float(lame_global_flags * gfp, const float buffer_l[], const float buffer_r[], int nsamples, unsigned char *mp3buf, const int mp3buf_size) { lame_internal_flags *gfc = gfp->internal_flags; int ret, i; sample_t *in_buffer[2]; if (gfc->Class_ID != LAME_ID) return -3; if (nsamples == 0) return 0; in_buffer[0] = calloc(sizeof(sample_t), nsamples); in_buffer[1] = calloc(sizeof(sample_t), nsamples); if (in_buffer[0] == NULL || in_buffer[1] == NULL) { ERRORF(gfc, "Error: can't allocate in_buffer buffer\n"); return -2; } /* make a copy of input buffer, changing type to sample_t */ for (i = 0; i < nsamples; i++) { in_buffer[0][i] = buffer_l[i]; in_buffer[1][i] = buffer_r[i]; } ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1], nsamples, mp3buf, mp3buf_size); free(in_buffer[0]); free(in_buffer[1]); return ret; } int lame_encode_buffer_long(lame_global_flags * gfp, const long buffer_l[], const long buffer_r[], int nsamples, unsigned char *mp3buf, const int mp3buf_size) { lame_internal_flags *gfc = gfp->internal_flags; int ret, i; sample_t *in_buffer[2]; if (gfc->Class_ID != LAME_ID) return -3; if (nsamples == 0) return 0; in_buffer[0] = calloc(sizeof(sample_t), nsamples); in_buffer[1] = calloc(sizeof(sample_t), nsamples); if (in_buffer[0] == NULL || in_buffer[1] == NULL) { ERRORF(gfc, "Error: can't allocate in_buffer buffer\n"); return -2; } /* make a copy of input buffer, changing type to sample_t */ for (i = 0; i < nsamples; i++) { in_buffer[0][i] = buffer_l[i]; in_buffer[1][i] = buffer_r[i]; } ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1], nsamples, mp3buf, mp3buf_size); free(in_buffer[0]); free(in_buffer[1]); return ret; } int lame_encode_buffer_interleaved(lame_global_flags * gfp, short int buffer[], int nsamples, unsigned char *mp3buf, int mp3buf_size) { int ret, i; short int *buffer_l; short int *buffer_r; buffer_l = malloc(sizeof(short int) * nsamples); buffer_r = malloc(sizeof(short int) * nsamples); if (buffer_l == NULL || buffer_r == NULL) { return -2; } for (i = 0; i < nsamples; i++) { buffer_l[i] = buffer[2 * i]; buffer_r[i] = buffer[2 * i + 1]; } ret = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size); free(buffer_l); free(buffer_r); return ret; } /*}}}*/ /* int lame_encode (lame_global_flags* gfp, short int in_buffer[2][1152], char* mp3buf, int size ) *//*{{{ */ /* old LAME interface. use lame_encode_buffer instead */ int lame_encode(lame_global_flags * const gfp, const short int in_buffer[2][1152], unsigned char *const mp3buf, const int size) { lame_internal_flags *gfc = gfp->internal_flags; if (gfc->Class_ID != LAME_ID) return -3; return lame_encode_buffer(gfp, in_buffer[0], in_buffer[1], gfp->framesize, mp3buf, size); } /*}}}*/ /* int lame_encode_flush (lame_global_flags* gfp, char* mp3buffer, int mp3buffer_size ) *//*{{{ */ /**/ /* flush internal mp3 buffers, */ /**/ int lame_encode_flush(lame_global_flags * gfp, unsigned char *mp3buffer, int mp3buffer_size) { short int buffer[2][1152]; int imp3 = 0, mp3count, mp3buffer_size_remaining; lame_internal_flags *gfc = gfp->internal_flags; memset(buffer, 0, sizeof(buffer)); mp3count = 0; while (gfc->mf_samples_to_encode > 0) { mp3buffer_size_remaining = mp3buffer_size - mp3count; /* if user specifed buffer size = 0, dont check size */ if (mp3buffer_size == 0) mp3buffer_size_remaining = 0; /* send in a frame of 0 padding until all internal sample buffers * are flushed */ imp3 = lame_encode_buffer(gfp, buffer[0], buffer[1], gfp->framesize, mp3buffer, mp3buffer_size_remaining); /* don't count the above padding: */ gfc->mf_samples_to_encode -= gfp->framesize; if (imp3 < 0) { /* some type of fatal error */ return imp3; } mp3buffer += imp3; mp3count += imp3; } mp3buffer_size_remaining = mp3buffer_size - mp3count; /* if user specifed buffer size = 0, dont check size */ if (mp3buffer_size == 0) mp3buffer_size_remaining = 0; if (gfp->ogg) { #ifdef HAVE_VORBIS /* ogg related stuff */ imp3 = lame_encode_ogg_finish(gfp, mp3buffer, mp3buffer_size_remaining); #endif } else { /* mp3 related stuff. bit buffer might still contain some mp3 data */ flush_bitstream(gfp); /* write a id3 tag to the bitstream */ id3tag_write_v1(gfp); imp3 = copy_buffer(mp3buffer, mp3buffer_size_remaining, &gfc->bs); } if (imp3 < 0) { return imp3; } mp3count += imp3; return mp3count; } /*}}}*/ /* void lame_close (lame_global_flags *gfp) *//*{{{ */ /* * * lame_close () * * frees internal buffers * */ int lame_close(lame_global_flags * gfp) { lame_internal_flags *gfc = gfp->internal_flags; if (gfc->Class_ID != LAME_ID) return -3; gfc->Class_ID = 0; // this routien will free all malloc'd data in gfc, and then free gfc: freegfc(gfc); gfp->internal_flags = NULL; if (gfp->lame_allocated_gfp) free(gfp); return 0; } /*}}}*/ /* int lame_encode_finish (lame_global_flags* gfp, char* mp3buffer, int mp3buffer_size ) *//*{{{ */ /**/ /* flush internal mp3 buffers, and free internal buffers */ /**/ int lame_encode_finish(lame_global_flags * gfp, unsigned char *mp3buffer, int mp3buffer_size) { int ret = lame_encode_flush(gfp, mp3buffer, mp3buffer_size); lame_close(gfp); return ret; } /*}}}*/ /* void lame_mp3_tags_fid (lame_global_flags *gfp,FILE *fpStream) *//*{{{ */ /**/ /* write VBR Xing header, and ID3 version 1 tag, if asked for */ /**/ void lame_mp3_tags_fid(lame_global_flags * gfp, FILE * fpStream) { if (gfp->bWriteVbrTag && (gfp->VBR != vbr_off)) { /* Map VBR_q to Xing quality value: 0=worst, 100=best */ int nQuality = ((9-gfp->VBR_q) * 100) / 9; /* Write Xing header again */ if (fpStream && !fseek(fpStream, 0, SEEK_SET)) PutVbrTag(gfp, fpStream, nQuality); } } /*}}}*/ /* lame_global_flags *lame_init (void) *//*{{{ */ lame_global_flags * lame_init(void) { lame_global_flags *gfp; int ret; gfp = calloc(1, sizeof(lame_global_flags)); if (gfp == NULL) return NULL; ret = lame_init_old(gfp); if (ret != 0) { free(gfp); return NULL; } gfp->lame_allocated_gfp = 1; return gfp; } /*}}}*/ /* int lame_init_old (lame_global_flags *gfp) *//*{{{ */ /* initialize mp3 encoder */ int lame_init_old(lame_global_flags * gfp) { lame_internal_flags *gfc; disable_FPE(); // disable floating point exceptions memset(gfp, 0, sizeof(lame_global_flags)); if (NULL == (gfc = gfp->internal_flags = calloc(1, sizeof(lame_internal_flags)))) return -1; /* Global flags. set defaults here for non-zero values */ /* see lame.h for description */ /* set integer values to -1 to mean that LAME will compute the * best value, UNLESS the calling program as set it * (and the value is no longer -1) */ gfp->mode = NOT_SET; gfp->original = 1; gfp->in_samplerate = 1000 * 44.1; gfp->num_channels = 2; gfp->num_samples = MAX_U_32_NUM; gfp->bWriteVbrTag = 1; gfp->quality = -1; gfp->allow_diff_short = -1; gfp->lowpassfreq = 0; gfp->highpassfreq = 0; gfp->lowpasswidth = -1; gfp->highpasswidth = -1; gfp->padding_type = 2; gfp->VBR = vbr_off; gfp->VBR_q = 4; gfp->VBR_mean_bitrate_kbps = 128; gfp->VBR_min_bitrate_kbps = 0; gfp->VBR_max_bitrate_kbps = 0; gfp->VBR_hard_min = 0; gfc->resample_ratio = 1; gfc->lowpass_band = 32; gfc->highpass_band = -1; gfc->VBR_min_bitrate = 1; /* not 0 ????? */ gfc->VBR_max_bitrate = 13; /* not 14 ????? */ gfc->OldValue[0] = 180; gfc->OldValue[1] = 180; gfc->CurrentStep = 4; gfc->masking_lower = 1; gfp->ATHtype = -1; /* default = -1 = set in lame_init_params */ gfp->useTemporal = 1; /* The reason for * int mf_samples_to_encode = ENCDELAY + 288; * ENCDELAY = internal encoder delay. And then we have to add 288 * because of the 50% MDCT overlap. A 576 MDCT granule decodes to * 1152 samples. To synthesize the 576 samples centered under this granule * we need the previous granule for the first 288 samples (no problem), and * the next granule for the next 288 samples (not possible if this is last * granule). So we need to pad with 288 samples to make sure we can * encode the 576 samples we are interested in. */ gfc->mf_samples_to_encode = ENCDELAY + 288; gfc->mf_size = ENCDELAY - MDCTDELAY; /* we pad input with this many 0's */ #ifdef KLEMM_44 /* XXX: this wasn't protectes by KLEMM_44 initially! */ gfc->last_ampl = gfc->ampl = +1.0; #endif return 0; } /*}}}*/ /* * * some simple statistics * * Robert Hegemann 2000-10-11 * */ /* histogram of used bitrate indexes: * One has to weight them to calculate the average bitrate in kbps * * bitrate indices: * there are 14 possible bitrate indices, 0 has the special meaning * "free format" which is not possible to mix with VBR and 15 is forbidden * anyway. * * stereo modes: * 0: LR number of left-right encoded frames * 1: LR-I number of left-right and intensity encoded frames * 2: MS number of mid-side encoded frames * 3: MS-I number of mid-side and intensity encoded frames * * 4: number of encoded frames * */ void lame_bitrate_hist(const lame_global_flags * const gfp, int bitrate_count[14]) { const lame_internal_flags *gfc; int i; if (NULL == bitrate_count) return; if (NULL == gfp) return; gfc = gfp->internal_flags; if (NULL == gfc) return; for (i = 0; i < 14; i++) bitrate_count[i] = gfc->bitrate_stereoMode_Hist[i + 1][4]; } void lame_bitrate_kbps(const lame_global_flags * const gfp, int bitrate_kbps[14]) { const lame_internal_flags *gfc; int i; if (NULL == bitrate_kbps) return; if (NULL == gfp) return; gfc = gfp->internal_flags; if (NULL == gfc) return; for (i = 0; i < 14; i++) bitrate_kbps[i] = bitrate_table[gfp->version][i + 1]; } void lame_stereo_mode_hist(const lame_global_flags * const gfp, int stmode_count[4]) { const lame_internal_flags *gfc; int i; if (NULL == stmode_count) return; if (NULL == gfp) return; gfc = gfp->internal_flags; if (NULL == gfc) return; for (i = 0; i < 4; i++) { int j, sum = 0; for (j = 0; j < 14; j++) sum += gfc->bitrate_stereoMode_Hist[j + 1][i]; stmode_count[i] = sum; } } void lame_bitrate_stereo_mode_hist(const lame_global_flags * const gfp, int bitrate_stmode_count[14][4]) { const lame_internal_flags *gfc; int i; int j; if (NULL == bitrate_stmode_count) return; if (NULL == gfp) return; gfc = gfp->internal_flags; if (NULL == gfc) return; for (j = 0; j < 14; j++) for (i = 0; i < 4; i++) bitrate_stmode_count[j][i] = gfc->bitrate_stereoMode_Hist[j + 1][i]; } /* end of lame.c */