/* -*- 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 */
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