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/* -*- mode: C; mode: fold -*- */
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/*
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* LAME MP3 encoding engine
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*
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* Copyright (c) 1999 Mark Taylor
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* $Id: lame.c,v 1.100 2001/03/25 23:14:45 markt Exp $ */
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include <assert.h>
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#include "lame-analysis.h"
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#include "lame.h"
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#include "util.h"
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#include "bitstream.h"
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#include "version.h"
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#include "tables.h"
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#include "quantize_pvt.h"
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#include "VbrTag.h"
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#if defined(__FreeBSD__) && !defined(__alpha__)
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#include <floatingpoint.h>
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#endif
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#ifdef __riscos__
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#include "asmstuff.h"
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#endif
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#ifdef WITH_DMALLOC
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#include <dmalloc.h>
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#endif
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static void
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lame_init_params_ppflt_lowpass(FLOAT8 amp_lowpass[32], FLOAT lowpass1,
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FLOAT lowpass2, int *lowpass_band,
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int *minband, int *maxband)
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{
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int band;
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FLOAT8 freq;
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for (band = 0; band <= 31; band++) {
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freq = band / 31.0;
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amp_lowpass[band] = 1;
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/* this band and above will be zeroed: */
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if (freq >= lowpass2) {
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*lowpass_band = Min(*lowpass_band, band);
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amp_lowpass[band] = 0;
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}
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if (lowpass1 < freq && freq < lowpass2) {
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*minband = Min(*minband, band);
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*maxband = Max(*maxband, band);
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amp_lowpass[band] = cos((PI / 2) *
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(lowpass1 - freq) / (lowpass2 - lowpass1));
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}
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/*
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* DEBUGF("lowpass band=%i amp=%f \n",
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* band, gfc->amp_lowpass[band]);
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*/
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}
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}
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/* lame_init_params_ppflt */
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/*}}}*/
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/* static void lame_init_params_ppflt (lame_internal_flags *gfc) *//*{{{ */
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static void
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lame_init_params_ppflt(lame_global_flags * gfp)
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{
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lame_internal_flags *gfc = gfp->internal_flags;
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/**/
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/* compute info needed for polyphase filter (filter type==0, default) */
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/**/
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int band, maxband, minband;
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FLOAT8 freq;
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if (gfc->lowpass1 > 0) {
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minband = 999;
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maxband = -1;
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lame_init_params_ppflt_lowpass(gfc->amp_lowpass,
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gfc->lowpass1, gfc->lowpass2,
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&gfc->lowpass_band, &minband, &maxband);
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/* compute the *actual* transition band implemented by
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* the polyphase filter */
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if (minband == 999) {
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gfc->lowpass1 = (gfc->lowpass_band - .75) / 31.0;
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}
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else {
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gfc->lowpass1 = (minband - .75) / 31.0;
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}
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gfc->lowpass2 = gfc->lowpass_band / 31.0;
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gfc->lowpass_start_band = minband;
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gfc->lowpass_end_band = maxband;
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/* as the lowpass may have changed above
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* calculate the amplification here again
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*/
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for (band = minband; band <= maxband; band++) {
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freq = band / 31.0;
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gfc->amp_lowpass[band] =
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cos((PI / 2) * (gfc->lowpass1 - freq) /
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(gfc->lowpass2 - gfc->lowpass1));
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}
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}
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else {
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gfc->lowpass_start_band = 0;
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gfc->lowpass_end_band = -1; /* do not to run into for-loops */
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}
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/* make sure highpass filter is within 90% of what the effective
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* highpass frequency will be */
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if (gfc->highpass2 > 0) {
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if (gfc->highpass2 < .9 * (.75 / 31.0)) {
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gfc->highpass1 = 0;
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gfc->highpass2 = 0;
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MSGF(gfc, "Warning: highpass filter disabled. "
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"highpass frequency too small\n");
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}
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}
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if (gfc->highpass2 > 0) {
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minband = 999;
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maxband = -1;
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for (band = 0; band <= 31; band++) {
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freq = band / 31.0;
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gfc->amp_highpass[band] = 1;
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/* this band and below will be zereod */
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if (freq <= gfc->highpass1) {
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gfc->highpass_band = Max(gfc->highpass_band, band);
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gfc->amp_highpass[band] = 0;
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}
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if (gfc->highpass1 < freq && freq < gfc->highpass2) {
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minband = Min(minband, band);
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maxband = Max(maxband, band);
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gfc->amp_highpass[band] =
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cos((PI / 2) *
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(gfc->highpass2 - freq) /
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(gfc->highpass2 - gfc->highpass1));
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}
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/*
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DEBUGF("highpass band=%i amp=%f \n",
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band, gfc->amp_highpass[band]);
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*/
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}
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/* compute the *actual* transition band implemented by
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* the polyphase filter */
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gfc->highpass1 = gfc->highpass_band / 31.0;
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if (maxband == -1) {
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gfc->highpass2 = (gfc->highpass_band + .75) / 31.0;
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}
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else {
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gfc->highpass2 = (maxband + .75) / 31.0;
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}
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gfc->highpass_start_band = minband;
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gfc->highpass_end_band = maxband;
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/* as the highpass may have changed above
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* calculate the amplification here again
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*/
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for (band = minband; band <= maxband; band++) {
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freq = band / 31.0;
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gfc->amp_highpass[band] =
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cos((PI / 2) * (gfc->highpass2 - freq) /
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(gfc->highpass2 - gfc->highpass1));
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}
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}
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else {
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gfc->highpass_start_band = 0;
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gfc->highpass_end_band = -1; /* do not to run into for-loops */
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}
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/*
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DEBUGF("lowpass band with amp=0: %i \n",gfc->lowpass_band);
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DEBUGF("highpass band with amp=0: %i \n",gfc->highpass_band);
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DEBUGF("lowpass band start: %i \n",gfc->lowpass_start_band);
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DEBUGF("lowpass band end: %i \n",gfc->lowpass_end_band);
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DEBUGF("highpass band start: %i \n",gfc->highpass_start_band);
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DEBUGF("highpass band end: %i \n",gfc->highpass_end_band);
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*/
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}
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/*}}}*/
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static void
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optimum_bandwidth(double *const lowerlimit,
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double *const upperlimit,
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const unsigned bitrate,
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const int samplefreq,
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const double channels, lame_global_flags * gfp)
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{
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/*
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* Input:
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* bitrate total bitrate in bps
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* samplefreq output sampling frequency in Hz
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* channels 1 for mono, 2+epsilon for MS stereo, 3 for LR stereo
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* epsilon is the percentage of LR frames for typical audio
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* (I use 'Fade to Gray' by Metallica)
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*
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* Output:
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* lowerlimit: best lowpass frequency limit for input filter in Hz
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* upperlimit: best highpass frequency limit for input filter in Hz
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*/
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double f_low;
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double f_high;
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double br;
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assert(bitrate >= 8000 && bitrate <= 320000);
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assert(samplefreq >= 8000 && samplefreq <= 48000);
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assert(channels == 1 || (channels >= 2 && channels <= 3));
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if (samplefreq >= 32000)
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br =
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bitrate - (channels ==
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1 ? (17 + 4) * 8 : (32 + 4) * 8) * samplefreq / 1152;
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else
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br =
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bitrate - (channels ==
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1 ? (9 + 4) * 8 : (17 + 4) * 8) * samplefreq / 576;
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if (channels >= 2.)
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br /= 1.75 + 0.25 * (channels - 2.); // MS needs 1.75x mono, LR needs 2.00x mono (experimental data of a lot of albums)
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br *= 0.5; // the sine and cosine term must share the bitrate
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/*
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* So, now we have the bitrate for every spectral line.
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* Let's look at the current settings:
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*
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* Bitrate limit bits/line
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* 8 kbps 0.34 kHz 4.76
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* 16 kbps 1.9 kHz 2.06
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* 24 kbps 2.8 kHz 2.21
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* 32 kbps 3.85 kHz 2.14
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* 40 kbps 5.1 kHz 2.06
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* 48 kbps 5.6 kHz 2.21
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* 56 kbps 7.0 kHz 2.10
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* 64 kbps 7.7 kHz 2.14
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* 80 kbps 10.1 kHz 2.08
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* 96 kbps 11.2 kHz 2.24
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* 112 kbps 14.0 kHz 2.12
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* 128 kbps 15.4 kHz 2.17
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* 160 kbps 18.2 kHz 2.05
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* 192 kbps 21.1 kHz 2.14
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* 224 kbps 22.0 kHz 2.41
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* 256 kbps 22.0 kHz 2.78
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*
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* What can we see?
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* Value for 8 kbps is nonsense (although 8 kbps and stereo is nonsense)
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* Values are between 2.05 and 2.24 for 16...192 kbps
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* Some bitrate lack the following bitrates have: 16, 40, 80, 160 kbps
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* A lot of bits per spectral line have: 24, 48, 96 kbps
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*
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* What I propose?
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* A slightly with the bitrate increasing bits/line function. It is
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* better to decrease NMR for low bitrates to get a little bit more
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* bandwidth. So we have a better trade off between twickling and
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* muffled sound.
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*/
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f_low = br / log10(br * 4.425e-3); // Tests with 8, 16, 32, 64, 112 and 160 kbps
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/*
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* What we get now?
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*
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* Bitrate limit bits/line difference
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* 8 kbps (8) 1.89 kHz 0.86 +1.6 kHz
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* 16 kbps (8) 3.16 kHz 1.24 +1.2 kHz
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* 32 kbps(16) 5.08 kHz 1.54 +1.2 kHz
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* 56 kbps(22) 7.88 kHz 1.80 +0.9 kHz
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* 64 kbps(22) 8.83 kHz 1.86 +1.1 kHz
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* 112 kbps(32) 14.02 kHz 2.12 0.0 kHz
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* 112 kbps(44) 13.70 kHz 2.11 -0.3 kHz
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* 128 kbps 15.40 kHz 2.17 0.0 kHz
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* 160 kbps 16.80 kHz 2.22 -1.4 kHz
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* 192 kbps 19.66 kHz 2.30 -1.4 kHz
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* 256 kbps 22.05 kHz 2.78 0.0 kHz
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*/
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#if 0
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/*
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* Beginning at 128 kbps/jstereo, we can use the following additional
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* strategy:
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*
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* For every increase of f_low in a way that the ATH(f_low)
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* increases by 4 dB we force an additional NMR of 1.25 dB.
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* These are the setting of the VBR quality selecting scheme
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309 |
* for V <= 4.
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310 |
*/
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311 |
{
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312 |
double br_sw = (128000 - (32 + 4) * 8 * 44100 / 1152) / 1.75 * 0.5;
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313 |
double f_low_sw = br_sw / log10(br_sw * 4.425e-3);
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314 |
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// printf ("br_sw=%f f_low_sw=%f\n", br_sw, f_low_sw );
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// printf ("br =%f f_low =%f\n", br , f_low );
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// fflush (stdout);
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318 |
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319 |
while (f_low > f_low_sw) {
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double dATH = ATHformula(f_low, gfp) - ATHformula(f_low_sw, gfp); // [dB]
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double dNMR = br / f_low - br_sw / f_low_sw; // bit
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322 |
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// printf ("br =%f f_low =%f\n", br , f_low );
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// printf ("dATH =%f dNMR =%f\n", dATH , dNMR );
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// fflush (stdout);
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326 |
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327 |
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328 |
if (dATH / 4.0 < dNMR * 6.0206 / 1.25) // 1 bit = 6.0206... dB
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329 |
break;
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f_low -= 25.;
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}
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332 |
}
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333 |
#endif
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334 |
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335 |
/*
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336 |
* Now we try to choose a good high pass filtering frequency.
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337 |
* This value is currently not used.
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338 |
* For fu < 16 kHz: sqrt(fu*fl) = 560 Hz
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339 |
* For fu = 18 kHz: no high pass filtering
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340 |
* This gives:
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341 |
*
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342 |
* 2 kHz => 160 Hz
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343 |
* 3 kHz => 107 Hz
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|
|
344 |
* 4 kHz => 80 Hz
|
|
|
345 |
* 8 kHz => 40 Hz
|
|
|
346 |
* 16 kHz => 20 Hz
|
|
|
347 |
* 17 kHz => 10 Hz
|
|
|
348 |
* 18 kHz => 0 Hz
|
|
|
349 |
*
|
|
|
350 |
* These are ad hoc values and these can be optimized if a high pass is available.
|
|
|
351 |
*/
|
|
|
352 |
if (f_low <= 16000)
|
|
|
353 |
f_high = 16000. * 20. / f_low;
|
|
|
354 |
else if (f_low <= 18000)
|
|
|
355 |
f_high = 180. - 0.01 * f_low;
|
|
|
356 |
else
|
|
|
357 |
f_high = 0.;
|
|
|
358 |
|
|
|
359 |
/*
|
|
|
360 |
* When we sometimes have a good highpass filter, we can add the highpass
|
|
|
361 |
* frequency to the lowpass frequency
|
|
|
362 |
*/
|
|
|
363 |
|
|
|
364 |
if (lowerlimit != NULL)
|
|
|
365 |
*lowerlimit = f_low /* + f_high */ ;
|
|
|
366 |
if (upperlimit != NULL)
|
|
|
367 |
*upperlimit = f_high;
|
|
|
368 |
/*
|
|
|
369 |
* Now the weak points:
|
|
|
370 |
*
|
|
|
371 |
* - the formula f_low=br/log10(br*4.425e-3) is an ad hoc formula
|
|
|
372 |
* (but has a physical background and is easy to tune)
|
|
|
373 |
* - the switch to the ATH based bandwidth selecting is the ad hoc
|
|
|
374 |
* value of 128 kbps
|
|
|
375 |
*/
|
|
|
376 |
}
|
|
|
377 |
|
|
|
378 |
static int
|
|
|
379 |
optimum_samplefreq(int lowpassfreq, int input_samplefreq)
|
|
|
380 |
{
|
|
|
381 |
/*
|
|
|
382 |
* Rules:
|
|
|
383 |
*
|
|
|
384 |
* - output sample frequency should NOT be decreased by more than 3% if lowpass allows this
|
|
|
385 |
* - if possible, sfb21 should NOT be used
|
|
|
386 |
*
|
|
|
387 |
* Problem: Switches to 32 kHz at 112 kbps
|
|
|
388 |
*/
|
|
|
389 |
if (input_samplefreq <= 8000 * 1.03 || lowpassfreq <= 3622)
|
|
|
390 |
return 8000;
|
|
|
391 |
if (input_samplefreq <= 11025 * 1.03 || lowpassfreq <= 4991)
|
|
|
392 |
return 11025;
|
|
|
393 |
if (input_samplefreq <= 12000 * 1.03 || lowpassfreq <= 5620)
|
|
|
394 |
return 12000;
|
|
|
395 |
if (input_samplefreq <= 16000 * 1.03 || lowpassfreq <= 7244)
|
|
|
396 |
return 16000;
|
|
|
397 |
if (input_samplefreq <= 22050 * 1.03 || lowpassfreq <= 9982)
|
|
|
398 |
return 22050;
|
|
|
399 |
if (input_samplefreq <= 24000 * 1.03 || lowpassfreq <= 11240)
|
|
|
400 |
return 24000;
|
|
|
401 |
if (input_samplefreq <= 32000 * 1.03 || lowpassfreq <= 15264)
|
|
|
402 |
return 32000;
|
|
|
403 |
if (input_samplefreq <= 44100 * 1.03)
|
|
|
404 |
return 44100;
|
|
|
405 |
return 48000;
|
|
|
406 |
}
|
|
|
407 |
|
|
|
408 |
|
|
|
409 |
/* set internal feature flags. USER should not access these since
|
|
|
410 |
* some combinations will produce strange results */
|
|
|
411 |
void
|
|
|
412 |
lame_init_qval(lame_global_flags * gfp)
|
|
|
413 |
{
|
|
|
414 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
415 |
|
|
|
416 |
switch (gfp->quality) {
|
|
|
417 |
case 9: /* no psymodel, no noise shaping */
|
|
|
418 |
gfc->filter_type = 0;
|
|
|
419 |
gfc->psymodel = 0;
|
|
|
420 |
gfc->quantization = 0;
|
|
|
421 |
gfc->noise_shaping = 0;
|
|
|
422 |
gfc->noise_shaping_amp = 0;
|
|
|
423 |
gfc->noise_shaping_stop = 0;
|
|
|
424 |
gfc->use_best_huffman = 0;
|
|
|
425 |
break;
|
|
|
426 |
|
|
|
427 |
case 8:
|
|
|
428 |
gfp->quality = 7;
|
|
|
429 |
case 7: /* use psymodel (for short block and m/s switching), but no noise shapping */
|
|
|
430 |
gfc->filter_type = 0;
|
|
|
431 |
gfc->psymodel = 1;
|
|
|
432 |
gfc->quantization = 0;
|
|
|
433 |
gfc->noise_shaping = 0;
|
|
|
434 |
gfc->noise_shaping_amp = 0;
|
|
|
435 |
gfc->noise_shaping_stop = 0;
|
|
|
436 |
gfc->use_best_huffman = 0;
|
|
|
437 |
break;
|
|
|
438 |
|
|
|
439 |
case 6:
|
|
|
440 |
gfp->quality = 5;
|
|
|
441 |
case 5: /* the default */
|
|
|
442 |
gfc->filter_type = 0;
|
|
|
443 |
gfc->psymodel = 1;
|
|
|
444 |
gfc->quantization = 0;
|
|
|
445 |
gfc->noise_shaping = 1;
|
|
|
446 |
/**/ gfc->noise_shaping_amp = 0;
|
|
|
447 |
gfc->noise_shaping_stop = 0;
|
|
|
448 |
gfc->use_best_huffman = 0;
|
|
|
449 |
break;
|
|
|
450 |
|
|
|
451 |
case 4:
|
|
|
452 |
gfp->quality = 3;
|
|
|
453 |
case 3:
|
|
|
454 |
gfc->filter_type = 0;
|
|
|
455 |
gfc->psymodel = 1;
|
|
|
456 |
gfc->quantization = 1;
|
|
|
457 |
gfc->noise_shaping = 1;
|
|
|
458 |
gfc->noise_shaping_amp = 0;
|
|
|
459 |
gfc->noise_shaping_stop = 0;
|
|
|
460 |
gfc->use_best_huffman = 1;
|
|
|
461 |
break;
|
|
|
462 |
|
|
|
463 |
case 2:
|
|
|
464 |
gfc->filter_type = 0;
|
|
|
465 |
gfc->psymodel = 1;
|
|
|
466 |
gfc->quantization = 1;
|
|
|
467 |
gfc->noise_shaping = 1;
|
|
|
468 |
gfc->noise_shaping_amp = 1;
|
|
|
469 |
gfc->noise_shaping_stop = 1;
|
|
|
470 |
gfc->use_best_huffman = 1;
|
|
|
471 |
break;
|
|
|
472 |
|
|
|
473 |
case 1:
|
|
|
474 |
gfc->filter_type = 0;
|
|
|
475 |
gfc->psymodel = 1;
|
|
|
476 |
gfc->quantization = 1;
|
|
|
477 |
gfc->noise_shaping = 1;
|
|
|
478 |
gfc->noise_shaping_amp = 2;
|
|
|
479 |
gfc->noise_shaping_stop = 1;
|
|
|
480 |
gfc->use_best_huffman = 1;
|
|
|
481 |
break;
|
|
|
482 |
|
|
|
483 |
case 0: /* 0..1 quality */
|
|
|
484 |
gfc->filter_type = 0; /* 1 not yet coded */
|
|
|
485 |
gfc->psymodel = 1;
|
|
|
486 |
gfc->quantization = 1;
|
|
|
487 |
gfc->noise_shaping = 1; /* 2=usually lowers quality */
|
|
|
488 |
gfc->noise_shaping_amp = 2;
|
|
|
489 |
gfc->noise_shaping_stop = 1;
|
|
|
490 |
gfc->use_best_huffman = 1; /* 2 not yet coded */
|
|
|
491 |
}
|
|
|
492 |
|
|
|
493 |
/* modifications to the above rules: */
|
|
|
494 |
|
|
|
495 |
/* -Z option enables scalefactor_scale: */
|
|
|
496 |
if (gfp->experimentalZ) {
|
|
|
497 |
gfc->noise_shaping = 2;
|
|
|
498 |
}
|
|
|
499 |
|
|
|
500 |
if (gfp->exp_nspsytune & 1) {
|
|
|
501 |
if (gfp->quality <= 2)
|
|
|
502 |
gfc->noise_shaping = 2; /* use scalefac_scale */
|
|
|
503 |
}
|
|
|
504 |
|
|
|
505 |
}
|
|
|
506 |
|
|
|
507 |
|
|
|
508 |
|
|
|
509 |
|
|
|
510 |
|
|
|
511 |
|
|
|
512 |
|
|
|
513 |
/* int lame_init_params (lame_global_flags *gfp) *//*{{{ */
|
|
|
514 |
|
|
|
515 |
/*
|
|
|
516 |
* initialize internal params based on data in gf
|
|
|
517 |
* (globalflags struct filled in by calling program)
|
|
|
518 |
*
|
|
|
519 |
* OUTLINE:
|
|
|
520 |
*
|
|
|
521 |
* We first have some complex code to determine bitrate,
|
|
|
522 |
* output samplerate and mode. It is complicated by the fact
|
|
|
523 |
* that we allow the user to set some or all of these parameters,
|
|
|
524 |
* and need to determine best possible values for the rest of them:
|
|
|
525 |
*
|
|
|
526 |
* 1. set some CPU related flags
|
|
|
527 |
* 2. check if we are mono->mono, stereo->mono or stereo->stereo
|
|
|
528 |
* 3. compute bitrate and output samplerate:
|
|
|
529 |
* user may have set compression ratio
|
|
|
530 |
* user may have set a bitrate
|
|
|
531 |
* user may have set a output samplerate
|
|
|
532 |
* 4. set some options which depend on output samplerate
|
|
|
533 |
* 5. compute the actual compression ratio
|
|
|
534 |
* 6. set mode based on compression ratio
|
|
|
535 |
*
|
|
|
536 |
* The remaining code is much simpler - it just sets options
|
|
|
537 |
* based on the mode & compression ratio:
|
|
|
538 |
*
|
|
|
539 |
* set allow_diff_short based on mode
|
|
|
540 |
* select lowpass filter based on compression ratio & mode
|
|
|
541 |
* set the bitrate index, and min/max bitrates for VBR modes
|
|
|
542 |
* disable VBR tag if it is not appropriate
|
|
|
543 |
* initialize the bitstream
|
|
|
544 |
* initialize scalefac_band data
|
|
|
545 |
* set sideinfo_len (based on channels, CRC, out_samplerate)
|
|
|
546 |
* write an id3v2 tag into the bitstream
|
|
|
547 |
* write VBR tag into the bitstream
|
|
|
548 |
* set mpeg1/2 flag
|
|
|
549 |
* estimate the number of frames (based on a lot of data)
|
|
|
550 |
*
|
|
|
551 |
* now we set more flags:
|
|
|
552 |
* nspsytune:
|
|
|
553 |
* see code
|
|
|
554 |
* VBR modes
|
|
|
555 |
* see code
|
|
|
556 |
* CBR/ABR
|
|
|
557 |
* see code
|
|
|
558 |
*
|
|
|
559 |
* Finally, we set the algorithm flags based on the gfp->quality value
|
|
|
560 |
* lame_init_qval(gfp);
|
|
|
561 |
*
|
|
|
562 |
*/
|
|
|
563 |
int
|
|
|
564 |
lame_init_params(lame_global_flags * const gfp)
|
|
|
565 |
{
|
|
|
566 |
|
|
|
567 |
int i;
|
|
|
568 |
int j;
|
|
|
569 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
570 |
|
|
|
571 |
gfc->gfp = gfp;
|
|
|
572 |
|
|
|
573 |
gfc->Class_ID = 0;
|
|
|
574 |
|
|
|
575 |
/* report functions */
|
|
|
576 |
gfc->report.msgf = gfp->report.msgf;
|
|
|
577 |
gfc->report.debugf = gfp->report.debugf;
|
|
|
578 |
gfc->report.errorf = gfp->report.errorf;
|
|
|
579 |
|
|
|
580 |
gfc->CPU_features.i387 = has_i387();
|
|
|
581 |
gfc->CPU_features.AMD_3DNow = has_3DNow();
|
|
|
582 |
gfc->CPU_features.MMX = has_MMX();
|
|
|
583 |
gfc->CPU_features.SIMD = has_SIMD();
|
|
|
584 |
gfc->CPU_features.SIMD2 = has_SIMD2();
|
|
|
585 |
|
|
|
586 |
|
|
|
587 |
if (NULL == gfc->ATH)
|
|
|
588 |
gfc->ATH = calloc(1, sizeof(ATH_t));
|
|
|
589 |
|
|
|
590 |
if (NULL == gfc->ATH)
|
|
|
591 |
return -2; // maybe error codes should be enumerated in lame.h ??
|
|
|
592 |
|
|
|
593 |
#ifdef KLEMM_44
|
|
|
594 |
/* Select the fastest functions for this CPU */
|
|
|
595 |
init_scalar_functions(gfc);
|
|
|
596 |
#endif
|
|
|
597 |
|
|
|
598 |
gfc->channels_in = gfp->num_channels;
|
|
|
599 |
if (gfc->channels_in == 1)
|
|
|
600 |
gfp->mode = MONO;
|
|
|
601 |
gfc->channels_out = (gfp->mode == MONO) ? 1 : 2;
|
|
|
602 |
gfc->mode_ext = MPG_MD_LR_LR;
|
|
|
603 |
if (gfp->mode == MONO)
|
|
|
604 |
gfp->force_ms = 0; // don't allow forced mid/side stereo for mono output
|
|
|
605 |
|
|
|
606 |
|
|
|
607 |
if (gfp->VBR != vbr_off) {
|
|
|
608 |
gfp->free_format = 0; /* VBR can't be mixed with free format */
|
|
|
609 |
}
|
|
|
610 |
|
|
|
611 |
if (gfp->VBR == vbr_off && gfp->brate == 0) {
|
|
|
612 |
/* no bitrate or compression ratio specified, use a compression ratio of 11.025 */
|
|
|
613 |
if (gfp->compression_ratio == 0)
|
|
|
614 |
gfp->compression_ratio = 11.025;
|
|
|
615 |
/* rate to compress a CD down to exactly 128000 bps */
|
|
|
616 |
}
|
|
|
617 |
|
|
|
618 |
|
|
|
619 |
if (gfp->VBR == vbr_off && gfp->brate == 0) {
|
|
|
620 |
/* no bitrate or compression ratio specified, use 11.025 */
|
|
|
621 |
if (gfp->compression_ratio == 0)
|
|
|
622 |
gfp->compression_ratio = 11.025;
|
|
|
623 |
/* rate to compress a CD down to exactly 128000 bps */
|
|
|
624 |
}
|
|
|
625 |
|
|
|
626 |
/* find bitrate if user specify a compression ratio */
|
|
|
627 |
if (gfp->VBR == vbr_off && gfp->compression_ratio > 0) {
|
|
|
628 |
|
|
|
629 |
if (gfp->out_samplerate == 0)
|
|
|
630 |
gfp->out_samplerate = map2MP3Frequency(0.97 * gfp->in_samplerate);
|
|
|
631 |
/* round up with a margin of 3% */
|
|
|
632 |
|
|
|
633 |
/* choose a bitrate for the output samplerate which achieves
|
|
|
634 |
* specified compression ratio
|
|
|
635 |
*/
|
|
|
636 |
gfp->brate = gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 *
|
|
|
637 |
gfp->compression_ratio);
|
|
|
638 |
|
|
|
639 |
/* we need the version for the bitrate table look up */
|
|
|
640 |
gfc->samplerate_index = SmpFrqIndex(gfp->out_samplerate, &gfp->version);
|
|
|
641 |
|
|
|
642 |
if (!gfp->free_format) /* for non Free Format find the nearest allowed bitrate */
|
|
|
643 |
gfp->brate =
|
|
|
644 |
FindNearestBitrate(gfp->brate, gfp->version,
|
|
|
645 |
gfp->out_samplerate);
|
|
|
646 |
}
|
|
|
647 |
|
|
|
648 |
/*
|
|
|
649 |
* at 160 kbps (MPEG-2/2.5)/ 320 kbps (MPEG-1), only Free format
|
|
|
650 |
* or CBR are possible, no VBR
|
|
|
651 |
*/
|
|
|
652 |
if (gfp->VBR != vbr_off && gfp->brate >= 320)
|
|
|
653 |
gfp->VBR = vbr_off;
|
|
|
654 |
|
|
|
655 |
if (gfp->out_samplerate == 0) {
|
|
|
656 |
/* if output sample frequency is not given, find a useful value */
|
|
|
657 |
gfp->out_samplerate = map2MP3Frequency(0.97 * gfp->in_samplerate);
|
|
|
658 |
|
|
|
659 |
|
|
|
660 |
/* check if user specified bitrate requires downsampling, if compression */
|
|
|
661 |
/* ratio is > 13, choose a new samplerate to get the ratio down to about 10 */
|
|
|
662 |
|
|
|
663 |
if (gfp->VBR == vbr_off && gfp->brate > 0) {
|
|
|
664 |
gfp->compression_ratio = gfp->out_samplerate * 16 *
|
|
|
665 |
gfc->channels_out / (1.e3 * gfp->brate);
|
|
|
666 |
if (gfp->compression_ratio > 13.)
|
|
|
667 |
gfp->out_samplerate = map2MP3Frequency((10. * 1.e3 *
|
|
|
668 |
gfp->brate) / (16 * gfc->channels_out));
|
|
|
669 |
}
|
|
|
670 |
if (gfp->VBR == vbr_abr) {
|
|
|
671 |
gfp->compression_ratio = gfp->out_samplerate * 16 *
|
|
|
672 |
gfc->channels_out / (1.e3 * gfp->VBR_mean_bitrate_kbps);
|
|
|
673 |
if (gfp->compression_ratio > 13.)
|
|
|
674 |
gfp->out_samplerate =
|
|
|
675 |
map2MP3Frequency((10. * 1.e3 * gfp->VBR_mean_bitrate_kbps) /
|
|
|
676 |
(16 * gfc->channels_out));
|
|
|
677 |
}
|
|
|
678 |
}
|
|
|
679 |
|
|
|
680 |
if (gfp->ogg) {
|
|
|
681 |
gfp->framesize = 1024;
|
|
|
682 |
gfp->encoder_delay = ENCDELAY;
|
|
|
683 |
gfc->coding = coding_Ogg_Vorbis;
|
|
|
684 |
}
|
|
|
685 |
else {
|
|
|
686 |
gfc->mode_gr = gfp->out_samplerate <= 24000 ? 1 : 2; // Number of granules per frame
|
|
|
687 |
gfp->framesize = 576 * gfc->mode_gr;
|
|
|
688 |
gfp->encoder_delay = ENCDELAY;
|
|
|
689 |
gfc->coding = coding_MPEG_Layer_3;
|
|
|
690 |
}
|
|
|
691 |
|
|
|
692 |
gfc->frame_size = gfp->framesize;
|
|
|
693 |
|
|
|
694 |
gfc->resample_ratio = (double) gfp->in_samplerate / gfp->out_samplerate;
|
|
|
695 |
|
|
|
696 |
/*
|
|
|
697 |
* sample freq bitrate compression ratio
|
|
|
698 |
* [kHz] [kbps/channel] for 16 bit input
|
|
|
699 |
* 44.1 56 12.6
|
|
|
700 |
* 44.1 64 11.025
|
|
|
701 |
* 44.1 80 8.82
|
|
|
702 |
* 22.05 24 14.7
|
|
|
703 |
* 22.05 32 11.025
|
|
|
704 |
* 22.05 40 8.82
|
|
|
705 |
* 16 16 16.0
|
|
|
706 |
* 16 24 10.667
|
|
|
707 |
*
|
|
|
708 |
*/
|
|
|
709 |
/*
|
|
|
710 |
* For VBR, take a guess at the compression_ratio.
|
|
|
711 |
* For example:
|
|
|
712 |
*
|
|
|
713 |
* VBR_q compression like
|
|
|
714 |
* - 4.4 320 kbps/44 kHz
|
|
|
715 |
* 0...1 5.5 256 kbps/44 kHz
|
|
|
716 |
* 2 7.3 192 kbps/44 kHz
|
|
|
717 |
* 4 8.8 160 kbps/44 kHz
|
|
|
718 |
* 6 11 128 kbps/44 kHz
|
|
|
719 |
* 9 14.7 96 kbps
|
|
|
720 |
*
|
|
|
721 |
* for lower bitrates, downsample with --resample
|
|
|
722 |
*/
|
|
|
723 |
|
|
|
724 |
switch (gfp->VBR) {
|
|
|
725 |
case vbr_mt:
|
|
|
726 |
case vbr_rh:
|
|
|
727 |
case vbr_mtrh:
|
|
|
728 |
{
|
|
|
729 |
FLOAT8 cmp[] = { 5, 6, 7, 8, 9, 10, 11, 12, 13, 14 };
|
|
|
730 |
gfp->compression_ratio = cmp[gfp->VBR_q];
|
|
|
731 |
}
|
|
|
732 |
break;
|
|
|
733 |
case vbr_abr:
|
|
|
734 |
gfp->compression_ratio = gfp->out_samplerate * 16 * gfc->channels_out /
|
|
|
735 |
(1.e3 * gfp->VBR_mean_bitrate_kbps);
|
|
|
736 |
break;
|
|
|
737 |
default:
|
|
|
738 |
gfp->compression_ratio =
|
|
|
739 |
gfp->out_samplerate * 16 * gfc->channels_out / (1.e3 * gfp->brate);
|
|
|
740 |
break;
|
|
|
741 |
}
|
|
|
742 |
|
|
|
743 |
|
|
|
744 |
/* mode = -1 (not set by user) or
|
|
|
745 |
* mode = MONO (because of only 1 input channel).
|
|
|
746 |
* If mode has been set, then select between STEREO or J-STEREO
|
|
|
747 |
* At higher quality (lower compression) use STEREO instead of J-STEREO.
|
|
|
748 |
* (unless the user explicitly specified a mode)
|
|
|
749 |
*
|
|
|
750 |
* The threshold to switch to STEREO is:
|
|
|
751 |
* 48 kHz: 171 kbps (used at 192+)
|
|
|
752 |
* 44.1 kHz: 160 kbps (used at 160+)
|
|
|
753 |
* 32 kHz: 119 kbps (used at 128+)
|
|
|
754 |
*
|
|
|
755 |
* Note, that for 32 kHz/128 kbps J-STEREO FM recordings sound much
|
|
|
756 |
* better than STEREO, so I'm not so very happy with that.
|
|
|
757 |
* fs < 32 kHz I have not tested.
|
|
|
758 |
*/
|
|
|
759 |
if (gfp->mode == NOT_SET) {
|
|
|
760 |
if (gfp->compression_ratio < 8)
|
|
|
761 |
gfp->mode = STEREO;
|
|
|
762 |
else
|
|
|
763 |
gfp->mode = JOINT_STEREO;
|
|
|
764 |
}
|
|
|
765 |
|
|
|
766 |
/* KLEMM's jstereo with ms threshold adjusted via compression ratio */
|
|
|
767 |
if (gfp->mode_automs) {
|
|
|
768 |
if (gfp->mode != MONO && gfp->compression_ratio < 6.6)
|
|
|
769 |
gfp->mode = STEREO;
|
|
|
770 |
}
|
|
|
771 |
|
|
|
772 |
|
|
|
773 |
if (gfp->allow_diff_short == -1) {
|
|
|
774 |
if (gfp->mode == STEREO)
|
|
|
775 |
gfp->allow_diff_short = 1;
|
|
|
776 |
}
|
|
|
777 |
|
|
|
778 |
|
|
|
779 |
|
|
|
780 |
|
|
|
781 |
/**/
|
|
|
782 |
/* if a filter has not been enabled, see if we should add one: */
|
|
|
783 |
/**/
|
|
|
784 |
if (gfp->lowpassfreq == 0) {
|
|
|
785 |
double lowpass;
|
|
|
786 |
double highpass;
|
|
|
787 |
double channels;
|
|
|
788 |
|
|
|
789 |
switch (gfp->mode) {
|
|
|
790 |
case MONO:
|
|
|
791 |
channels = 1.;
|
|
|
792 |
break;
|
|
|
793 |
case JOINT_STEREO:
|
|
|
794 |
channels = 2. + 0.00;
|
|
|
795 |
break;
|
|
|
796 |
case DUAL_CHANNEL:
|
|
|
797 |
case STEREO:
|
|
|
798 |
channels = 3.;
|
|
|
799 |
break;
|
|
|
800 |
default:
|
|
|
801 |
channels = 1.; // just to make data flow analysis happy :-)
|
|
|
802 |
assert(0);
|
|
|
803 |
break;
|
|
|
804 |
}
|
|
|
805 |
|
|
|
806 |
optimum_bandwidth(&lowpass,
|
|
|
807 |
&highpass,
|
|
|
808 |
gfp->out_samplerate * 16 * gfc->channels_out /
|
|
|
809 |
gfp->compression_ratio, gfp->out_samplerate, channels,
|
|
|
810 |
gfp);
|
|
|
811 |
|
|
|
812 |
if (lowpass < 0.5 * gfp->out_samplerate) {
|
|
|
813 |
//MSGF(gfc,"Lowpass @ %7.1f Hz\n", lowpass);
|
|
|
814 |
gfc->lowpass1 = gfc->lowpass2 =
|
|
|
815 |
lowpass / (0.5 * gfp->out_samplerate);
|
|
|
816 |
}
|
|
|
817 |
if (0 && gfp->out_samplerate !=
|
|
|
818 |
optimum_samplefreq(lowpass, gfp->in_samplerate)) {
|
|
|
819 |
MSGF(gfc,
|
|
|
820 |
"I would suggest to use %u Hz instead of %u Hz sample frequency\n",
|
|
|
821 |
optimum_samplefreq(lowpass, gfp->in_samplerate),
|
|
|
822 |
gfp->out_samplerate);
|
|
|
823 |
}
|
|
|
824 |
fflush(stderr);
|
|
|
825 |
}
|
|
|
826 |
|
|
|
827 |
/* apply user driven high pass filter */
|
|
|
828 |
if (gfp->highpassfreq > 0) {
|
|
|
829 |
gfc->highpass1 = 2. * gfp->highpassfreq / gfp->out_samplerate;
|
|
|
830 |
/* will always be >=0 */
|
|
|
831 |
if (gfp->highpasswidth >= 0)
|
|
|
832 |
gfc->highpass2 = 2. * (gfp->highpassfreq + gfp->highpasswidth) /
|
|
|
833 |
gfp->out_samplerate;
|
|
|
834 |
else /* 0% above on default */
|
|
|
835 |
gfc->highpass2 =
|
|
|
836 |
(1 + 0.00) * 2. * gfp->highpassfreq / gfp->out_samplerate;
|
|
|
837 |
}
|
|
|
838 |
|
|
|
839 |
/* apply user driven low pass filter */
|
|
|
840 |
if (gfp->lowpassfreq > 0) {
|
|
|
841 |
gfc->lowpass2 = 2. * gfp->lowpassfreq / gfp->out_samplerate;
|
|
|
842 |
/* will always be >=0 */
|
|
|
843 |
if (gfp->lowpasswidth >= 0) {
|
|
|
844 |
gfc->lowpass1 = 2. * (gfp->lowpassfreq - gfp->lowpasswidth) /
|
|
|
845 |
gfp->out_samplerate;
|
|
|
846 |
if (gfc->lowpass1 < 0) /* has to be >= 0 */
|
|
|
847 |
gfc->lowpass1 = 0;
|
|
|
848 |
}
|
|
|
849 |
else { /* 0% below on default */
|
|
|
850 |
gfc->lowpass1 =
|
|
|
851 |
(1 - 0.00) * 2. * gfp->lowpassfreq / gfp->out_samplerate;
|
|
|
852 |
}
|
|
|
853 |
}
|
|
|
854 |
|
|
|
855 |
/**/
|
|
|
856 |
/* compute info needed for polyphase filter (filter type==0, default) */
|
|
|
857 |
/**/
|
|
|
858 |
lame_init_params_ppflt(gfp);
|
|
|
859 |
|
|
|
860 |
|
|
|
861 |
/*
|
|
|
862 |
* compute info needed for FIR filter (filter_type==1)
|
|
|
863 |
*/
|
|
|
864 |
/* not yet coded */
|
|
|
865 |
|
|
|
866 |
|
|
|
867 |
|
|
|
868 |
/*
|
|
|
869 |
* samplerate and bitrate index
|
|
|
870 |
*/
|
|
|
871 |
gfc->samplerate_index = SmpFrqIndex(gfp->out_samplerate, &gfp->version);
|
|
|
872 |
if (gfc->samplerate_index < 0)
|
|
|
873 |
return -1;
|
|
|
874 |
|
|
|
875 |
if (gfp->VBR == vbr_off) {
|
|
|
876 |
if (gfp->free_format)
|
|
|
877 |
gfc->bitrate_index = 0;
|
|
|
878 |
else {
|
|
|
879 |
gfc->bitrate_index = BitrateIndex(gfp->brate, gfp->version,
|
|
|
880 |
gfp->out_samplerate);
|
|
|
881 |
if (gfc->bitrate_index < 0)
|
|
|
882 |
return -1;
|
|
|
883 |
}
|
|
|
884 |
}
|
|
|
885 |
else { /* choose a min/max bitrate for VBR */
|
|
|
886 |
/* if the user didn't specify VBR_max_bitrate: */
|
|
|
887 |
gfc->VBR_min_bitrate = 1; /* default: allow 8 kbps (MPEG-2) or 32 kbps (MPEG-1) */
|
|
|
888 |
gfc->VBR_max_bitrate = 14; /* default: allow 160 kbps (MPEG-2) or 320 kbps (MPEG-1) */
|
|
|
889 |
|
|
|
890 |
if (gfp->VBR_min_bitrate_kbps)
|
|
|
891 |
if (
|
|
|
892 |
(gfc->VBR_min_bitrate =
|
|
|
893 |
BitrateIndex(gfp->VBR_min_bitrate_kbps, gfp->version,
|
|
|
894 |
gfp->out_samplerate)) < 0) return -1;
|
|
|
895 |
if (gfp->VBR_max_bitrate_kbps)
|
|
|
896 |
if (
|
|
|
897 |
(gfc->VBR_max_bitrate =
|
|
|
898 |
BitrateIndex(gfp->VBR_max_bitrate_kbps, gfp->version,
|
|
|
899 |
gfp->out_samplerate)) < 0) return -1;
|
|
|
900 |
|
|
|
901 |
gfp->VBR_min_bitrate_kbps =
|
|
|
902 |
bitrate_table[gfp->version][gfc->VBR_min_bitrate];
|
|
|
903 |
gfp->VBR_max_bitrate_kbps =
|
|
|
904 |
bitrate_table[gfp->version][gfc->VBR_max_bitrate];
|
|
|
905 |
|
|
|
906 |
gfp->VBR_mean_bitrate_kbps =
|
|
|
907 |
Min(bitrate_table[gfp->version][gfc->VBR_max_bitrate],
|
|
|
908 |
gfp->VBR_mean_bitrate_kbps);
|
|
|
909 |
gfp->VBR_mean_bitrate_kbps =
|
|
|
910 |
Max(bitrate_table[gfp->version][gfc->VBR_min_bitrate],
|
|
|
911 |
gfp->VBR_mean_bitrate_kbps);
|
|
|
912 |
|
|
|
913 |
|
|
|
914 |
}
|
|
|
915 |
|
|
|
916 |
/* Do not write VBR tag if VBR flag is not specified */
|
|
|
917 |
if (gfp->VBR == vbr_off)
|
|
|
918 |
gfp->bWriteVbrTag = 0;
|
|
|
919 |
if (gfp->ogg)
|
|
|
920 |
gfp->bWriteVbrTag = 0;
|
|
|
921 |
if (gfp->analysis)
|
|
|
922 |
gfp->bWriteVbrTag = 0;
|
|
|
923 |
|
|
|
924 |
/* some file options not allowed if output is: not specified or stdout */
|
|
|
925 |
if (gfc->pinfo != NULL)
|
|
|
926 |
gfp->bWriteVbrTag = 0; /* disable Xing VBR tag */
|
|
|
927 |
|
|
|
928 |
init_bit_stream_w(gfc);
|
|
|
929 |
|
|
|
930 |
j = gfc->samplerate_index + (3 * gfp->version) + 6 * (gfp->out_samplerate <
|
|
|
931 |
16000);
|
|
|
932 |
for (i = 0; i < SBMAX_l + 1; i++)
|
|
|
933 |
gfc->scalefac_band.l[i] = sfBandIndex[j].l[i];
|
|
|
934 |
for (i = 0; i < SBMAX_s + 1; i++)
|
|
|
935 |
gfc->scalefac_band.s[i] = sfBandIndex[j].s[i];
|
|
|
936 |
|
|
|
937 |
/* determine the mean bitrate for main data */
|
|
|
938 |
if (gfp->version == 1) /* MPEG 1 */
|
|
|
939 |
gfc->sideinfo_len = (gfc->channels_out == 1) ? 4 + 17 : 4 + 32;
|
|
|
940 |
else /* MPEG 2 */
|
|
|
941 |
gfc->sideinfo_len = (gfc->channels_out == 1) ? 4 + 9 : 4 + 17;
|
|
|
942 |
|
|
|
943 |
if (gfp->error_protection)
|
|
|
944 |
gfc->sideinfo_len += 2;
|
|
|
945 |
|
|
|
946 |
|
|
|
947 |
/*
|
|
|
948 |
* Write id3v2 tag into the bitstream.
|
|
|
949 |
* This tag must be before the Xing VBR header.
|
|
|
950 |
*/
|
|
|
951 |
if (!gfp->ogg)
|
|
|
952 |
id3tag_write_v2(gfp);
|
|
|
953 |
|
|
|
954 |
|
|
|
955 |
/* Write initial VBR Header to bitstream */
|
|
|
956 |
if (gfp->bWriteVbrTag)
|
|
|
957 |
InitVbrTag(gfp);
|
|
|
958 |
|
|
|
959 |
if (gfp->version == 1) /* 0 indicates use lower sample freqs algorithm */
|
|
|
960 |
gfc->is_mpeg1 = 1; /* yes */
|
|
|
961 |
else
|
|
|
962 |
gfc->is_mpeg1 = 0; /* no */
|
|
|
963 |
|
|
|
964 |
/* estimate total frames. */
|
|
|
965 |
gfp->totalframes =
|
|
|
966 |
2 + gfp->num_samples / (gfc->resample_ratio * gfp->framesize);
|
|
|
967 |
gfc->Class_ID = LAME_ID;
|
|
|
968 |
|
|
|
969 |
if (gfp->exp_nspsytune & 1) {
|
|
|
970 |
int i;
|
|
|
971 |
|
|
|
972 |
gfc->nsPsy.use = 1;
|
|
|
973 |
gfc->nsPsy.safejoint = (gfp->exp_nspsytune & 2) != 0;
|
|
|
974 |
for (i = 0; i < 19; i++)
|
|
|
975 |
gfc->nsPsy.pefirbuf[i] = 700;
|
|
|
976 |
|
|
|
977 |
if (gfp->VBR == vbr_mtrh || gfp->VBR == vbr_mt) {
|
|
|
978 |
ERRORF(gfc, "\n**** nspsytune doesn't support --vbr-new **** \n\n");
|
|
|
979 |
gfp->VBR = vbr_rh;
|
|
|
980 |
}
|
|
|
981 |
|
|
|
982 |
if (gfp->ATHtype == -1)
|
|
|
983 |
gfp->ATHtype = 0;
|
|
|
984 |
|
|
|
985 |
gfc->nsPsy.bass = gfc->nsPsy.alto = gfc->nsPsy.treble = 0;
|
|
|
986 |
|
|
|
987 |
i = (gfp->exp_nspsytune >> 2) & 63;
|
|
|
988 |
if (i >= 32)
|
|
|
989 |
i -= 64;
|
|
|
990 |
gfc->nsPsy.bass = pow(10, i / 4.0 / 10.0);
|
|
|
991 |
i = (gfp->exp_nspsytune >> 8) & 63;
|
|
|
992 |
if (i >= 32)
|
|
|
993 |
i -= 64;
|
|
|
994 |
gfc->nsPsy.alto = pow(10, i / 4.0 / 10.0);
|
|
|
995 |
i = (gfp->exp_nspsytune >> 14) & 63;
|
|
|
996 |
if (i >= 32)
|
|
|
997 |
i -= 64;
|
|
|
998 |
gfc->nsPsy.treble = pow(10, i / 4.0 / 10.0);
|
|
|
999 |
}
|
|
|
1000 |
|
|
|
1001 |
switch (gfp->VBR) {
|
|
|
1002 |
case vbr_mtrh:
|
|
|
1003 |
/* default quality for --vbr-mtrh is 1
|
|
|
1004 |
*/
|
|
|
1005 |
if (gfp->quality < 0)
|
|
|
1006 |
gfp->quality = 1;
|
|
|
1007 |
|
|
|
1008 |
/* tonality
|
|
|
1009 |
*/
|
|
|
1010 |
if (gfp->cwlimit <= 0)
|
|
|
1011 |
gfp->cwlimit = 0.454 * gfp->out_samplerate;
|
|
|
1012 |
|
|
|
1013 |
/* fall through */
|
|
|
1014 |
case vbr_mt:
|
|
|
1015 |
/* use Gaby's ATH for vbr-mtrh by default
|
|
|
1016 |
*/
|
|
|
1017 |
if (gfp->ATHtype == -1)
|
|
|
1018 |
gfp->ATHtype = 2;
|
|
|
1019 |
|
|
|
1020 |
/* fall through */
|
|
|
1021 |
case vbr_rh:
|
|
|
1022 |
/* use Roel's tweaked Gaby-ATH for VBR by default
|
|
|
1023 |
*/
|
|
|
1024 |
if (gfp->ATHtype == -1)
|
|
|
1025 |
gfp->ATHtype = 2;
|
|
|
1026 |
|
|
|
1027 |
/* automatic ATH adjustment on, VBR modes need it
|
|
|
1028 |
*/
|
|
|
1029 |
gfc->ATH->use_adjust = 1;
|
|
|
1030 |
|
|
|
1031 |
/* sfb21 extra only with MPEG-1 at higher sampling rates
|
|
|
1032 |
*/
|
|
|
1033 |
gfc->sfb21_extra = (gfp->out_samplerate > 44000);
|
|
|
1034 |
|
|
|
1035 |
/* VBR needs at least the output of GPSYCHO,
|
|
|
1036 |
* so we have to garantee that by setting a minimum
|
|
|
1037 |
* quality level, actually level 5 does it.
|
|
|
1038 |
* the -v and -V x settings switch the quality to level 2
|
|
|
1039 |
* you would have to add a -q 5 to reduce the quality
|
|
|
1040 |
* down to level 5
|
|
|
1041 |
*/
|
|
|
1042 |
if (gfp->quality > 5)
|
|
|
1043 |
gfp->quality = 5;
|
|
|
1044 |
|
|
|
1045 |
/* default quality setting is 2
|
|
|
1046 |
*/
|
|
|
1047 |
if (gfp->quality < 0)
|
|
|
1048 |
gfp->quality = 2;
|
|
|
1049 |
|
|
|
1050 |
/* allow left and right channels to have different block types
|
|
|
1051 |
*/
|
|
|
1052 |
gfp->allow_diff_short = 1;
|
|
|
1053 |
break;
|
|
|
1054 |
default:
|
|
|
1055 |
/* automatic ATH adjustment off, not so important for CBR code
|
|
|
1056 |
*/
|
|
|
1057 |
gfc->ATH->use_adjust = 0;
|
|
|
1058 |
|
|
|
1059 |
/* use Frank's ATH for CBR/ABR by default
|
|
|
1060 |
*/
|
|
|
1061 |
if (gfp->ATHtype == -1)
|
|
|
1062 |
gfp->ATHtype = 2;
|
|
|
1063 |
|
|
|
1064 |
/* no sfb21 extra with CBR code
|
|
|
1065 |
*/
|
|
|
1066 |
gfc->sfb21_extra = 0;
|
|
|
1067 |
|
|
|
1068 |
/* default quality setting for CBR/ABR is 5
|
|
|
1069 |
*/
|
|
|
1070 |
if (gfp->quality < 0)
|
|
|
1071 |
gfp->quality = 5;
|
|
|
1072 |
break;
|
|
|
1073 |
}
|
|
|
1074 |
|
|
|
1075 |
/* initialize internal qval settings */
|
|
|
1076 |
lame_init_qval(gfp);
|
|
|
1077 |
|
|
|
1078 |
#ifdef KLEMM_44
|
|
|
1079 |
gfc->mfbuf[0] = (sample_t *) calloc(sizeof(sample_t), MFSIZE);
|
|
|
1080 |
gfc->mfbuf[1] = (sample_t *) calloc(sizeof(sample_t), MFSIZE);
|
|
|
1081 |
gfc->sampfreq_in = unround_samplefrequency(gfp->in_samplerate);
|
|
|
1082 |
gfc->sampfreq_out = gfp->out_samplerate;
|
|
|
1083 |
gfc->resample_in = resample_open(gfc->sampfreq_in, gfc->sampfreq_out,
|
|
|
1084 |
-1 .0 /* Auto */ , 32);
|
|
|
1085 |
#endif
|
|
|
1086 |
return 0;
|
|
|
1087 |
}
|
|
|
1088 |
|
|
|
1089 |
/*}}}*/
|
|
|
1090 |
/* void lame_print_config (lame_global_flags *gfp) *//*{{{ */
|
|
|
1091 |
|
|
|
1092 |
/*
|
|
|
1093 |
* print_config
|
|
|
1094 |
*
|
|
|
1095 |
* Prints some selected information about the coding parameters via
|
|
|
1096 |
* the macro command MSGF(), which is currently mapped to lame_errorf
|
|
|
1097 |
* (reports via a error function?), which is a printf-like function
|
|
|
1098 |
* for <stderr>.
|
|
|
1099 |
*/
|
|
|
1100 |
|
|
|
1101 |
void
|
|
|
1102 |
lame_print_config(const lame_global_flags * gfp)
|
|
|
1103 |
{
|
|
|
1104 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1105 |
double out_samplerate = gfp->out_samplerate;
|
|
|
1106 |
double in_samplerate = gfp->out_samplerate * gfc->resample_ratio;
|
|
|
1107 |
|
|
|
1108 |
MSGF(gfc, "mp3enc (from lame version %s (%s))\n", get_lame_version(), get_lame_url());
|
|
|
1109 |
|
|
|
1110 |
if (gfc->CPU_features.MMX
|
|
|
1111 |
|| gfc->CPU_features.AMD_3DNow
|
|
|
1112 |
|| gfc->CPU_features.SIMD || gfc->CPU_features.SIMD2) {
|
|
|
1113 |
MSGF(gfc, "CPU features:");
|
|
|
1114 |
|
|
|
1115 |
if (gfc->CPU_features.i387)
|
|
|
1116 |
MSGF(gfc, " i387");
|
|
|
1117 |
if (gfc->CPU_features.MMX)
|
|
|
1118 |
#ifdef MMX_choose_table
|
|
|
1119 |
MSGF(gfc, ", MMX (ASM used)");
|
|
|
1120 |
#else
|
|
|
1121 |
MSGF(gfc, ", MMX");
|
|
|
1122 |
#endif
|
|
|
1123 |
if (gfc->CPU_features.AMD_3DNow)
|
|
|
1124 |
MSGF(gfc, ", 3DNow!");
|
|
|
1125 |
if (gfc->CPU_features.SIMD)
|
|
|
1126 |
MSGF(gfc, ", SIMD");
|
|
|
1127 |
if (gfc->CPU_features.SIMD2)
|
|
|
1128 |
MSGF(gfc, ", SIMD2");
|
|
|
1129 |
MSGF(gfc, "\n");
|
|
|
1130 |
}
|
|
|
1131 |
|
|
|
1132 |
if (gfp->num_channels == 2 && gfc->channels_out == 1 /* mono */ ) {
|
|
|
1133 |
MSGF
|
|
|
1134 |
(gfc,
|
|
|
1135 |
"Autoconverting from stereo to mono. Setting encoding to mono mode.\n");
|
|
|
1136 |
}
|
|
|
1137 |
|
|
|
1138 |
if (gfc->resample_ratio != 1.) {
|
|
|
1139 |
MSGF(gfc, "Resampling: input %g kHz output %g kHz\n",
|
|
|
1140 |
1.e-3 * in_samplerate, 1.e-3 * out_samplerate);
|
|
|
1141 |
}
|
|
|
1142 |
|
|
|
1143 |
if (gfc->filter_type == 0) {
|
|
|
1144 |
if (gfc->highpass2 > 0.)
|
|
|
1145 |
MSGF
|
|
|
1146 |
(gfc,
|
|
|
1147 |
"Using polyphase highpass filter, transition band: %5.0f Hz - %5.0f Hz\n",
|
|
|
1148 |
0.5 * gfc->highpass1 * out_samplerate,
|
|
|
1149 |
0.5 * gfc->highpass2 * out_samplerate);
|
|
|
1150 |
if (gfc->lowpass1 > 0.) {
|
|
|
1151 |
MSGF
|
|
|
1152 |
(gfc,
|
|
|
1153 |
"Using polyphase lowpass filter, transition band: %5.0f Hz - %5.0f Hz\n",
|
|
|
1154 |
0.5 * gfc->lowpass1 * out_samplerate,
|
|
|
1155 |
0.5 * gfc->lowpass2 * out_samplerate);
|
|
|
1156 |
}
|
|
|
1157 |
else {
|
|
|
1158 |
MSGF(gfc, "polyphase lowpass filter disabled\n");
|
|
|
1159 |
}
|
|
|
1160 |
}
|
|
|
1161 |
else {
|
|
|
1162 |
MSGF(gfc, "polyphase filters disabled\n");
|
|
|
1163 |
}
|
|
|
1164 |
|
|
|
1165 |
if (gfp->free_format) {
|
|
|
1166 |
MSGF(gfc,
|
|
|
1167 |
"Warning: many decoders cannot handle free format bitstreams\n");
|
|
|
1168 |
if (gfp->brate > 320) {
|
|
|
1169 |
MSGF
|
|
|
1170 |
(gfc,
|
|
|
1171 |
"Warning: many decoders cannot handle free format bitrates >320 kbps (see documentation)\n");
|
|
|
1172 |
}
|
|
|
1173 |
}
|
|
|
1174 |
}
|
|
|
1175 |
|
|
|
1176 |
|
|
|
1177 |
/* int lame_encode_frame (lame_global_flags *gfp, sample_t inbuf_l[],sample_t inbuf_r[], char *mp3buf, int mp3buf_size) *//*{{{ */
|
|
|
1178 |
|
|
|
1179 |
/* routine to feed exactly one frame (gfp->framesize) worth of data to the
|
|
|
1180 |
encoding engine. All buffering, resampling, etc, handled by calling
|
|
|
1181 |
program.
|
|
|
1182 |
*/
|
|
|
1183 |
int
|
|
|
1184 |
lame_encode_frame(lame_global_flags * gfp,
|
|
|
1185 |
sample_t inbuf_l[], sample_t inbuf_r[],
|
|
|
1186 |
unsigned char *mp3buf, int mp3buf_size)
|
|
|
1187 |
{
|
|
|
1188 |
int ret;
|
|
|
1189 |
if (gfp->ogg) {
|
|
|
1190 |
#ifdef HAVE_VORBIS
|
|
|
1191 |
ret = lame_encode_ogg_frame(gfp, inbuf_l, inbuf_r, mp3buf, mp3buf_size);
|
|
|
1192 |
#else
|
|
|
1193 |
return -5; /* wanna encode ogg without vorbis */
|
|
|
1194 |
#endif
|
|
|
1195 |
}
|
|
|
1196 |
else {
|
|
|
1197 |
ret = lame_encode_mp3_frame(gfp, inbuf_l, inbuf_r, mp3buf, mp3buf_size);
|
|
|
1198 |
}
|
|
|
1199 |
|
|
|
1200 |
/* check to see if we underestimated totalframes */
|
|
|
1201 |
gfp->frameNum++;
|
|
|
1202 |
if (gfp->totalframes < gfp->frameNum)
|
|
|
1203 |
gfp->totalframes = gfp->frameNum;
|
|
|
1204 |
return ret;
|
|
|
1205 |
}
|
|
|
1206 |
|
|
|
1207 |
/*}}}*/
|
|
|
1208 |
/* int lame_encode_buffer (lame_global_flags* gfp, short int buffer_l[], short int buffer_r[], int nsamples, char* mp3buf, int mp3buf_size )*//*{{{ */
|
|
|
1209 |
|
|
|
1210 |
|
|
|
1211 |
|
|
|
1212 |
/*
|
|
|
1213 |
* THE MAIN LAME ENCODING INTERFACE
|
|
|
1214 |
* mt 3/00
|
|
|
1215 |
*
|
|
|
1216 |
* input pcm data, output (maybe) mp3 frames.
|
|
|
1217 |
* This routine handles all buffering, resampling and filtering for you.
|
|
|
1218 |
* The required mp3buffer_size can be computed from num_samples,
|
|
|
1219 |
* samplerate and encoding rate, but here is a worst case estimate:
|
|
|
1220 |
*
|
|
|
1221 |
* mp3buffer_size in bytes = 1.25*num_samples + 7200
|
|
|
1222 |
*
|
|
|
1223 |
* return code = number of bytes output in mp3buffer. can be 0
|
|
|
1224 |
*/
|
|
|
1225 |
int
|
|
|
1226 |
lame_encode_buffer_sample_t(lame_global_flags * gfp,
|
|
|
1227 |
sample_t buffer_l[],
|
|
|
1228 |
sample_t buffer_r[],
|
|
|
1229 |
int nsamples, unsigned char *mp3buf, const int mp3buf_size)
|
|
|
1230 |
{
|
|
|
1231 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1232 |
int mp3size = 0, ret, i, ch, mf_needed;
|
|
|
1233 |
sample_t *mfbuf[2];
|
|
|
1234 |
sample_t *in_buffer[2];
|
|
|
1235 |
|
|
|
1236 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1237 |
return -3;
|
|
|
1238 |
|
|
|
1239 |
if (nsamples == 0)
|
|
|
1240 |
return 0;
|
|
|
1241 |
|
|
|
1242 |
in_buffer[0]=buffer_l;
|
|
|
1243 |
in_buffer[1]=buffer_r;
|
|
|
1244 |
|
|
|
1245 |
|
|
|
1246 |
/* some sanity checks */
|
|
|
1247 |
#if ENCDELAY < MDCTDELAY
|
|
|
1248 |
# error ENCDELAY is less than MDCTDELAY, see encoder.h
|
|
|
1249 |
#endif
|
|
|
1250 |
#if FFTOFFSET > BLKSIZE
|
|
|
1251 |
# error FFTOFFSET is greater than BLKSIZE, see encoder.h
|
|
|
1252 |
#endif
|
|
|
1253 |
|
|
|
1254 |
mf_needed = BLKSIZE + gfp->framesize - FFTOFFSET; /* amount needed for FFT */
|
|
|
1255 |
mf_needed = Max(mf_needed, 286 + 576 * (1 + gfc->mode_gr)); /* amount needed for MDCT/filterbank */
|
|
|
1256 |
assert(MFSIZE >= mf_needed);
|
|
|
1257 |
|
|
|
1258 |
mfbuf[0] = gfc->mfbuf[0];
|
|
|
1259 |
mfbuf[1] = gfc->mfbuf[1];
|
|
|
1260 |
|
|
|
1261 |
if (gfp->num_channels == 2 && gfc->channels_out == 1) {
|
|
|
1262 |
/* downsample to mono */
|
|
|
1263 |
for (i = 0; i < nsamples; ++i) {
|
|
|
1264 |
in_buffer[0][i] =
|
|
|
1265 |
0.5 * ((FLOAT8) in_buffer[0][i] + in_buffer[1][i]);
|
|
|
1266 |
in_buffer[1][i] = 0.0;
|
|
|
1267 |
}
|
|
|
1268 |
}
|
|
|
1269 |
|
|
|
1270 |
|
|
|
1271 |
while (nsamples > 0) {
|
|
|
1272 |
int n_in = 0; /* number of input samples processed with fill_buffer */
|
|
|
1273 |
int n_out = 0; /* number of samples output with fill_buffer */
|
|
|
1274 |
/* n_in <> n_out if we are resampling */
|
|
|
1275 |
|
|
|
1276 |
/* copy in new samples into mfbuf, with resampling & scaling if necessary */
|
|
|
1277 |
fill_buffer(gfp, mfbuf, in_buffer, nsamples, &n_in, &n_out);
|
|
|
1278 |
|
|
|
1279 |
/* update in_buffer counters */
|
|
|
1280 |
nsamples -= n_in;
|
|
|
1281 |
in_buffer[0] += n_in;
|
|
|
1282 |
if (gfc->channels_out == 2)
|
|
|
1283 |
in_buffer[1] += n_in;
|
|
|
1284 |
|
|
|
1285 |
/* update mfbuf[] counters */
|
|
|
1286 |
gfc->mf_size += n_out;
|
|
|
1287 |
assert(gfc->mf_size <= MFSIZE);
|
|
|
1288 |
gfc->mf_samples_to_encode += n_out;
|
|
|
1289 |
|
|
|
1290 |
|
|
|
1291 |
if (gfc->mf_size >= mf_needed) {
|
|
|
1292 |
/* encode the frame. */
|
|
|
1293 |
ret =
|
|
|
1294 |
lame_encode_frame(gfp, mfbuf[0], mfbuf[1], mp3buf, mp3buf_size);
|
|
|
1295 |
|
|
|
1296 |
if (ret < 0)
|
|
|
1297 |
goto retr;
|
|
|
1298 |
mp3buf += ret;
|
|
|
1299 |
mp3size += ret;
|
|
|
1300 |
|
|
|
1301 |
/* shift out old samples */
|
|
|
1302 |
gfc->mf_size -= gfp->framesize;
|
|
|
1303 |
gfc->mf_samples_to_encode -= gfp->framesize;
|
|
|
1304 |
for (ch = 0; ch < gfc->channels_out; ch++)
|
|
|
1305 |
for (i = 0; i < gfc->mf_size; i++)
|
|
|
1306 |
mfbuf[ch][i] = mfbuf[ch][i + gfp->framesize];
|
|
|
1307 |
}
|
|
|
1308 |
}
|
|
|
1309 |
assert(nsamples == 0);
|
|
|
1310 |
ret = mp3size;
|
|
|
1311 |
|
|
|
1312 |
retr:
|
|
|
1313 |
return ret;
|
|
|
1314 |
}
|
|
|
1315 |
|
|
|
1316 |
|
|
|
1317 |
int
|
|
|
1318 |
lame_encode_buffer(lame_global_flags * gfp,
|
|
|
1319 |
const short int buffer_l[],
|
|
|
1320 |
const short int buffer_r[],
|
|
|
1321 |
int nsamples, unsigned char *mp3buf, const int mp3buf_size)
|
|
|
1322 |
{
|
|
|
1323 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1324 |
int ret, i;
|
|
|
1325 |
sample_t *in_buffer[2];
|
|
|
1326 |
|
|
|
1327 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1328 |
return -3;
|
|
|
1329 |
|
|
|
1330 |
if (nsamples == 0)
|
|
|
1331 |
return 0;
|
|
|
1332 |
|
|
|
1333 |
in_buffer[0] = calloc(sizeof(sample_t), nsamples);
|
|
|
1334 |
in_buffer[1] = calloc(sizeof(sample_t), nsamples);
|
|
|
1335 |
|
|
|
1336 |
if (in_buffer[0] == NULL || in_buffer[1] == NULL) {
|
|
|
1337 |
ERRORF(gfc, "Error: can't allocate in_buffer buffer\n");
|
|
|
1338 |
return -2;
|
|
|
1339 |
}
|
|
|
1340 |
|
|
|
1341 |
/* make a copy of input buffer, changing type to sample_t */
|
|
|
1342 |
for (i = 0; i < nsamples; i++) {
|
|
|
1343 |
in_buffer[0][i] = buffer_l[i];
|
|
|
1344 |
in_buffer[1][i] = buffer_r[i];
|
|
|
1345 |
}
|
|
|
1346 |
|
|
|
1347 |
ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1],
|
|
|
1348 |
nsamples, mp3buf, mp3buf_size);
|
|
|
1349 |
|
|
|
1350 |
free(in_buffer[0]);
|
|
|
1351 |
free(in_buffer[1]);
|
|
|
1352 |
return ret;
|
|
|
1353 |
}
|
|
|
1354 |
|
|
|
1355 |
|
|
|
1356 |
int
|
|
|
1357 |
lame_encode_buffer_float(lame_global_flags * gfp,
|
|
|
1358 |
const float buffer_l[],
|
|
|
1359 |
const float buffer_r[],
|
|
|
1360 |
int nsamples, unsigned char *mp3buf, const int mp3buf_size)
|
|
|
1361 |
{
|
|
|
1362 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1363 |
int ret, i;
|
|
|
1364 |
sample_t *in_buffer[2];
|
|
|
1365 |
|
|
|
1366 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1367 |
return -3;
|
|
|
1368 |
|
|
|
1369 |
if (nsamples == 0)
|
|
|
1370 |
return 0;
|
|
|
1371 |
|
|
|
1372 |
in_buffer[0] = calloc(sizeof(sample_t), nsamples);
|
|
|
1373 |
in_buffer[1] = calloc(sizeof(sample_t), nsamples);
|
|
|
1374 |
|
|
|
1375 |
if (in_buffer[0] == NULL || in_buffer[1] == NULL) {
|
|
|
1376 |
ERRORF(gfc, "Error: can't allocate in_buffer buffer\n");
|
|
|
1377 |
return -2;
|
|
|
1378 |
}
|
|
|
1379 |
|
|
|
1380 |
/* make a copy of input buffer, changing type to sample_t */
|
|
|
1381 |
for (i = 0; i < nsamples; i++) {
|
|
|
1382 |
in_buffer[0][i] = buffer_l[i];
|
|
|
1383 |
in_buffer[1][i] = buffer_r[i];
|
|
|
1384 |
}
|
|
|
1385 |
|
|
|
1386 |
ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1],
|
|
|
1387 |
nsamples, mp3buf, mp3buf_size);
|
|
|
1388 |
|
|
|
1389 |
free(in_buffer[0]);
|
|
|
1390 |
free(in_buffer[1]);
|
|
|
1391 |
return ret;
|
|
|
1392 |
}
|
|
|
1393 |
|
|
|
1394 |
|
|
|
1395 |
|
|
|
1396 |
int
|
|
|
1397 |
lame_encode_buffer_long(lame_global_flags * gfp,
|
|
|
1398 |
const long buffer_l[],
|
|
|
1399 |
const long buffer_r[],
|
|
|
1400 |
int nsamples, unsigned char *mp3buf, const int mp3buf_size)
|
|
|
1401 |
{
|
|
|
1402 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1403 |
int ret, i;
|
|
|
1404 |
sample_t *in_buffer[2];
|
|
|
1405 |
|
|
|
1406 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1407 |
return -3;
|
|
|
1408 |
|
|
|
1409 |
if (nsamples == 0)
|
|
|
1410 |
return 0;
|
|
|
1411 |
|
|
|
1412 |
in_buffer[0] = calloc(sizeof(sample_t), nsamples);
|
|
|
1413 |
in_buffer[1] = calloc(sizeof(sample_t), nsamples);
|
|
|
1414 |
|
|
|
1415 |
if (in_buffer[0] == NULL || in_buffer[1] == NULL) {
|
|
|
1416 |
ERRORF(gfc, "Error: can't allocate in_buffer buffer\n");
|
|
|
1417 |
return -2;
|
|
|
1418 |
}
|
|
|
1419 |
|
|
|
1420 |
/* make a copy of input buffer, changing type to sample_t */
|
|
|
1421 |
for (i = 0; i < nsamples; i++) {
|
|
|
1422 |
in_buffer[0][i] = buffer_l[i];
|
|
|
1423 |
in_buffer[1][i] = buffer_r[i];
|
|
|
1424 |
}
|
|
|
1425 |
|
|
|
1426 |
ret = lame_encode_buffer_sample_t(gfp,in_buffer[0],in_buffer[1],
|
|
|
1427 |
nsamples, mp3buf, mp3buf_size);
|
|
|
1428 |
|
|
|
1429 |
free(in_buffer[0]);
|
|
|
1430 |
free(in_buffer[1]);
|
|
|
1431 |
return ret;
|
|
|
1432 |
}
|
|
|
1433 |
|
|
|
1434 |
|
|
|
1435 |
|
|
|
1436 |
|
|
|
1437 |
|
|
|
1438 |
|
|
|
1439 |
|
|
|
1440 |
|
|
|
1441 |
|
|
|
1442 |
|
|
|
1443 |
|
|
|
1444 |
int
|
|
|
1445 |
lame_encode_buffer_interleaved(lame_global_flags * gfp,
|
|
|
1446 |
short int buffer[],
|
|
|
1447 |
int nsamples,
|
|
|
1448 |
unsigned char *mp3buf, int mp3buf_size)
|
|
|
1449 |
{
|
|
|
1450 |
int ret, i;
|
|
|
1451 |
short int *buffer_l;
|
|
|
1452 |
short int *buffer_r;
|
|
|
1453 |
|
|
|
1454 |
buffer_l = malloc(sizeof(short int) * nsamples);
|
|
|
1455 |
buffer_r = malloc(sizeof(short int) * nsamples);
|
|
|
1456 |
if (buffer_l == NULL || buffer_r == NULL) {
|
|
|
1457 |
return -2;
|
|
|
1458 |
}
|
|
|
1459 |
for (i = 0; i < nsamples; i++) {
|
|
|
1460 |
buffer_l[i] = buffer[2 * i];
|
|
|
1461 |
buffer_r[i] = buffer[2 * i + 1];
|
|
|
1462 |
}
|
|
|
1463 |
ret =
|
|
|
1464 |
lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf,
|
|
|
1465 |
mp3buf_size);
|
|
|
1466 |
free(buffer_l);
|
|
|
1467 |
free(buffer_r);
|
|
|
1468 |
return ret;
|
|
|
1469 |
|
|
|
1470 |
}
|
|
|
1471 |
|
|
|
1472 |
|
|
|
1473 |
/*}}}*/
|
|
|
1474 |
/* int lame_encode (lame_global_flags* gfp, short int in_buffer[2][1152], char* mp3buf, int size ) *//*{{{ */
|
|
|
1475 |
|
|
|
1476 |
|
|
|
1477 |
/* old LAME interface. use lame_encode_buffer instead */
|
|
|
1478 |
|
|
|
1479 |
int
|
|
|
1480 |
lame_encode(lame_global_flags * const gfp,
|
|
|
1481 |
const short int in_buffer[2][1152],
|
|
|
1482 |
unsigned char *const mp3buf, const int size)
|
|
|
1483 |
{
|
|
|
1484 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1485 |
|
|
|
1486 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1487 |
return -3;
|
|
|
1488 |
|
|
|
1489 |
return lame_encode_buffer(gfp, in_buffer[0], in_buffer[1], gfp->framesize,
|
|
|
1490 |
mp3buf, size);
|
|
|
1491 |
}
|
|
|
1492 |
|
|
|
1493 |
/*}}}*/
|
|
|
1494 |
/* int lame_encode_flush (lame_global_flags* gfp, char* mp3buffer, int mp3buffer_size ) *//*{{{ */
|
|
|
1495 |
|
|
|
1496 |
/**/
|
|
|
1497 |
/* flush internal mp3 buffers, */
|
|
|
1498 |
/**/
|
|
|
1499 |
|
|
|
1500 |
int
|
|
|
1501 |
lame_encode_flush(lame_global_flags * gfp,
|
|
|
1502 |
unsigned char *mp3buffer, int mp3buffer_size)
|
|
|
1503 |
{
|
|
|
1504 |
short int buffer[2][1152];
|
|
|
1505 |
int imp3 = 0, mp3count, mp3buffer_size_remaining;
|
|
|
1506 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1507 |
|
|
|
1508 |
memset(buffer, 0, sizeof(buffer));
|
|
|
1509 |
mp3count = 0;
|
|
|
1510 |
|
|
|
1511 |
while (gfc->mf_samples_to_encode > 0) {
|
|
|
1512 |
|
|
|
1513 |
mp3buffer_size_remaining = mp3buffer_size - mp3count;
|
|
|
1514 |
|
|
|
1515 |
/* if user specifed buffer size = 0, dont check size */
|
|
|
1516 |
if (mp3buffer_size == 0)
|
|
|
1517 |
mp3buffer_size_remaining = 0;
|
|
|
1518 |
|
|
|
1519 |
/* send in a frame of 0 padding until all internal sample buffers
|
|
|
1520 |
* are flushed
|
|
|
1521 |
*/
|
|
|
1522 |
imp3 = lame_encode_buffer(gfp, buffer[0], buffer[1], gfp->framesize,
|
|
|
1523 |
mp3buffer, mp3buffer_size_remaining);
|
|
|
1524 |
/* don't count the above padding: */
|
|
|
1525 |
gfc->mf_samples_to_encode -= gfp->framesize;
|
|
|
1526 |
|
|
|
1527 |
if (imp3 < 0) {
|
|
|
1528 |
/* some type of fatal error */
|
|
|
1529 |
return imp3;
|
|
|
1530 |
}
|
|
|
1531 |
mp3buffer += imp3;
|
|
|
1532 |
mp3count += imp3;
|
|
|
1533 |
}
|
|
|
1534 |
|
|
|
1535 |
mp3buffer_size_remaining = mp3buffer_size - mp3count;
|
|
|
1536 |
/* if user specifed buffer size = 0, dont check size */
|
|
|
1537 |
if (mp3buffer_size == 0)
|
|
|
1538 |
mp3buffer_size_remaining = 0;
|
|
|
1539 |
|
|
|
1540 |
if (gfp->ogg) {
|
|
|
1541 |
#ifdef HAVE_VORBIS
|
|
|
1542 |
/* ogg related stuff */
|
|
|
1543 |
imp3 = lame_encode_ogg_finish(gfp, mp3buffer, mp3buffer_size_remaining);
|
|
|
1544 |
#endif
|
|
|
1545 |
}
|
|
|
1546 |
else {
|
|
|
1547 |
/* mp3 related stuff. bit buffer might still contain some mp3 data */
|
|
|
1548 |
flush_bitstream(gfp);
|
|
|
1549 |
/* write a id3 tag to the bitstream */
|
|
|
1550 |
id3tag_write_v1(gfp);
|
|
|
1551 |
imp3 = copy_buffer(mp3buffer, mp3buffer_size_remaining, &gfc->bs);
|
|
|
1552 |
}
|
|
|
1553 |
|
|
|
1554 |
if (imp3 < 0) {
|
|
|
1555 |
return imp3;
|
|
|
1556 |
}
|
|
|
1557 |
mp3count += imp3;
|
|
|
1558 |
return mp3count;
|
|
|
1559 |
}
|
|
|
1560 |
|
|
|
1561 |
/*}}}*/
|
|
|
1562 |
/* void lame_close (lame_global_flags *gfp) *//*{{{ */
|
|
|
1563 |
|
|
|
1564 |
/*
|
|
|
1565 |
*
|
|
|
1566 |
* lame_close ()
|
|
|
1567 |
*
|
|
|
1568 |
* frees internal buffers
|
|
|
1569 |
*
|
|
|
1570 |
*/
|
|
|
1571 |
|
|
|
1572 |
int
|
|
|
1573 |
lame_close(lame_global_flags * gfp)
|
|
|
1574 |
{
|
|
|
1575 |
lame_internal_flags *gfc = gfp->internal_flags;
|
|
|
1576 |
|
|
|
1577 |
if (gfc->Class_ID != LAME_ID)
|
|
|
1578 |
return -3;
|
|
|
1579 |
|
|
|
1580 |
gfc->Class_ID = 0;
|
|
|
1581 |
|
|
|
1582 |
// this routien will free all malloc'd data in gfc, and then free gfc:
|
|
|
1583 |
freegfc(gfc);
|
|
|
1584 |
|
|
|
1585 |
gfp->internal_flags = NULL;
|
|
|
1586 |
|
|
|
1587 |
if (gfp->lame_allocated_gfp)
|
|
|
1588 |
free(gfp);
|
|
|
1589 |
|
|
|
1590 |
return 0;
|
|
|
1591 |
}
|
|
|
1592 |
|
|
|
1593 |
|
|
|
1594 |
/*}}}*/
|
|
|
1595 |
/* int lame_encode_finish (lame_global_flags* gfp, char* mp3buffer, int mp3buffer_size ) *//*{{{ */
|
|
|
1596 |
|
|
|
1597 |
|
|
|
1598 |
/**/
|
|
|
1599 |
/* flush internal mp3 buffers, and free internal buffers */
|
|
|
1600 |
/**/
|
|
|
1601 |
|
|
|
1602 |
int
|
|
|
1603 |
lame_encode_finish(lame_global_flags * gfp,
|
|
|
1604 |
unsigned char *mp3buffer, int mp3buffer_size)
|
|
|
1605 |
{
|
|
|
1606 |
int ret = lame_encode_flush(gfp, mp3buffer, mp3buffer_size);
|
|
|
1607 |
|
|
|
1608 |
lame_close(gfp);
|
|
|
1609 |
|
|
|
1610 |
return ret;
|
|
|
1611 |
}
|
|
|
1612 |
|
|
|
1613 |
/*}}}*/
|
|
|
1614 |
/* void lame_mp3_tags_fid (lame_global_flags *gfp,FILE *fpStream) *//*{{{ */
|
|
|
1615 |
|
|
|
1616 |
/**/
|
|
|
1617 |
/* write VBR Xing header, and ID3 version 1 tag, if asked for */
|
|
|
1618 |
/**/
|
|
|
1619 |
void
|
|
|
1620 |
lame_mp3_tags_fid(lame_global_flags * gfp, FILE * fpStream)
|
|
|
1621 |
{
|
|
|
1622 |
if (gfp->bWriteVbrTag && (gfp->VBR != vbr_off)) {
|
|
|
1623 |
/* Map VBR_q to Xing quality value: 0=worst, 100=best */
|
|
|
1624 |
int nQuality = ((9-gfp->VBR_q) * 100) / 9;
|
|
|
1625 |
|
|
|
1626 |
/* Write Xing header again */
|
|
|
1627 |
if (fpStream && !fseek(fpStream, 0, SEEK_SET))
|
|
|
1628 |
PutVbrTag(gfp, fpStream, nQuality);
|
|
|
1629 |
}
|
|
|
1630 |
|
|
|
1631 |
|
|
|
1632 |
}
|
|
|
1633 |
/*}}}*/
|
|
|
1634 |
/* lame_global_flags *lame_init (void) *//*{{{ */
|
|
|
1635 |
|
|
|
1636 |
lame_global_flags *
|
|
|
1637 |
lame_init(void)
|
|
|
1638 |
{
|
|
|
1639 |
lame_global_flags *gfp;
|
|
|
1640 |
int ret;
|
|
|
1641 |
|
|
|
1642 |
gfp = calloc(1, sizeof(lame_global_flags));
|
|
|
1643 |
if (gfp == NULL)
|
|
|
1644 |
return NULL;
|
|
|
1645 |
|
|
|
1646 |
ret = lame_init_old(gfp);
|
|
|
1647 |
if (ret != 0) {
|
|
|
1648 |
free(gfp);
|
|
|
1649 |
return NULL;
|
|
|
1650 |
}
|
|
|
1651 |
|
|
|
1652 |
gfp->lame_allocated_gfp = 1;
|
|
|
1653 |
return gfp;
|
|
|
1654 |
}
|
|
|
1655 |
|
|
|
1656 |
/*}}}*/
|
|
|
1657 |
/* int lame_init_old (lame_global_flags *gfp) *//*{{{ */
|
|
|
1658 |
|
|
|
1659 |
/* initialize mp3 encoder */
|
|
|
1660 |
int
|
|
|
1661 |
lame_init_old(lame_global_flags * gfp)
|
|
|
1662 |
{
|
|
|
1663 |
lame_internal_flags *gfc;
|
|
|
1664 |
|
|
|
1665 |
disable_FPE(); // disable floating point exceptions
|
|
|
1666 |
|
|
|
1667 |
memset(gfp, 0, sizeof(lame_global_flags));
|
|
|
1668 |
|
|
|
1669 |
if (NULL ==
|
|
|
1670 |
(gfc = gfp->internal_flags =
|
|
|
1671 |
calloc(1, sizeof(lame_internal_flags)))) return -1;
|
|
|
1672 |
|
|
|
1673 |
/* Global flags. set defaults here for non-zero values */
|
|
|
1674 |
/* see lame.h for description */
|
|
|
1675 |
/* set integer values to -1 to mean that LAME will compute the
|
|
|
1676 |
* best value, UNLESS the calling program as set it
|
|
|
1677 |
* (and the value is no longer -1)
|
|
|
1678 |
*/
|
|
|
1679 |
|
|
|
1680 |
|
|
|
1681 |
gfp->mode = NOT_SET;
|
|
|
1682 |
gfp->original = 1;
|
|
|
1683 |
gfp->in_samplerate = 1000 * 44.1;
|
|
|
1684 |
gfp->num_channels = 2;
|
|
|
1685 |
gfp->num_samples = MAX_U_32_NUM;
|
|
|
1686 |
|
|
|
1687 |
gfp->bWriteVbrTag = 1;
|
|
|
1688 |
gfp->quality = -1;
|
|
|
1689 |
gfp->allow_diff_short = -1;
|
|
|
1690 |
|
|
|
1691 |
gfp->lowpassfreq = 0;
|
|
|
1692 |
gfp->highpassfreq = 0;
|
|
|
1693 |
gfp->lowpasswidth = -1;
|
|
|
1694 |
gfp->highpasswidth = -1;
|
|
|
1695 |
|
|
|
1696 |
gfp->padding_type = 2;
|
|
|
1697 |
gfp->VBR = vbr_off;
|
|
|
1698 |
gfp->VBR_q = 4;
|
|
|
1699 |
gfp->VBR_mean_bitrate_kbps = 128;
|
|
|
1700 |
gfp->VBR_min_bitrate_kbps = 0;
|
|
|
1701 |
gfp->VBR_max_bitrate_kbps = 0;
|
|
|
1702 |
gfp->VBR_hard_min = 0;
|
|
|
1703 |
|
|
|
1704 |
|
|
|
1705 |
gfc->resample_ratio = 1;
|
|
|
1706 |
gfc->lowpass_band = 32;
|
|
|
1707 |
gfc->highpass_band = -1;
|
|
|
1708 |
gfc->VBR_min_bitrate = 1; /* not 0 ????? */
|
|
|
1709 |
gfc->VBR_max_bitrate = 13; /* not 14 ????? */
|
|
|
1710 |
|
|
|
1711 |
gfc->OldValue[0] = 180;
|
|
|
1712 |
gfc->OldValue[1] = 180;
|
|
|
1713 |
gfc->CurrentStep = 4;
|
|
|
1714 |
gfc->masking_lower = 1;
|
|
|
1715 |
|
|
|
1716 |
gfp->ATHtype = -1; /* default = -1 = set in lame_init_params */
|
|
|
1717 |
gfp->useTemporal = 1;
|
|
|
1718 |
|
|
|
1719 |
/* The reason for
|
|
|
1720 |
* int mf_samples_to_encode = ENCDELAY + 288;
|
|
|
1721 |
* ENCDELAY = internal encoder delay. And then we have to add 288
|
|
|
1722 |
* because of the 50% MDCT overlap. A 576 MDCT granule decodes to
|
|
|
1723 |
* 1152 samples. To synthesize the 576 samples centered under this granule
|
|
|
1724 |
* we need the previous granule for the first 288 samples (no problem), and
|
|
|
1725 |
* the next granule for the next 288 samples (not possible if this is last
|
|
|
1726 |
* granule). So we need to pad with 288 samples to make sure we can
|
|
|
1727 |
* encode the 576 samples we are interested in.
|
|
|
1728 |
*/
|
|
|
1729 |
gfc->mf_samples_to_encode = ENCDELAY + 288;
|
|
|
1730 |
gfc->mf_size = ENCDELAY - MDCTDELAY; /* we pad input with this many 0's */
|
|
|
1731 |
|
|
|
1732 |
#ifdef KLEMM_44
|
|
|
1733 |
/* XXX: this wasn't protectes by KLEMM_44 initially! */
|
|
|
1734 |
gfc->last_ampl = gfc->ampl = +1.0;
|
|
|
1735 |
#endif
|
|
|
1736 |
|
|
|
1737 |
return 0;
|
|
|
1738 |
}
|
|
|
1739 |
|
|
|
1740 |
/*}}}*/
|
|
|
1741 |
|
|
|
1742 |
/*
|
|
|
1743 |
*
|
|
|
1744 |
* some simple statistics
|
|
|
1745 |
*
|
|
|
1746 |
* Robert Hegemann 2000-10-11
|
|
|
1747 |
*
|
|
|
1748 |
*/
|
|
|
1749 |
|
|
|
1750 |
/* histogram of used bitrate indexes:
|
|
|
1751 |
* One has to weight them to calculate the average bitrate in kbps
|
|
|
1752 |
*
|
|
|
1753 |
* bitrate indices:
|
|
|
1754 |
* there are 14 possible bitrate indices, 0 has the special meaning
|
|
|
1755 |
* "free format" which is not possible to mix with VBR and 15 is forbidden
|
|
|
1756 |
* anyway.
|
|
|
1757 |
*
|
|
|
1758 |
* stereo modes:
|
|
|
1759 |
* 0: LR number of left-right encoded frames
|
|
|
1760 |
* 1: LR-I number of left-right and intensity encoded frames
|
|
|
1761 |
* 2: MS number of mid-side encoded frames
|
|
|
1762 |
* 3: MS-I number of mid-side and intensity encoded frames
|
|
|
1763 |
*
|
|
|
1764 |
* 4: number of encoded frames
|
|
|
1765 |
*
|
|
|
1766 |
*/
|
|
|
1767 |
|
|
|
1768 |
void
|
|
|
1769 |
lame_bitrate_hist(const lame_global_flags * const gfp, int bitrate_count[14])
|
|
|
1770 |
{
|
|
|
1771 |
const lame_internal_flags *gfc;
|
|
|
1772 |
int i;
|
|
|
1773 |
|
|
|
1774 |
if (NULL == bitrate_count)
|
|
|
1775 |
return;
|
|
|
1776 |
if (NULL == gfp)
|
|
|
1777 |
return;
|
|
|
1778 |
gfc = gfp->internal_flags;
|
|
|
1779 |
if (NULL == gfc)
|
|
|
1780 |
return;
|
|
|
1781 |
|
|
|
1782 |
for (i = 0; i < 14; i++)
|
|
|
1783 |
bitrate_count[i] = gfc->bitrate_stereoMode_Hist[i + 1][4];
|
|
|
1784 |
}
|
|
|
1785 |
|
|
|
1786 |
|
|
|
1787 |
void
|
|
|
1788 |
lame_bitrate_kbps(const lame_global_flags * const gfp, int bitrate_kbps[14])
|
|
|
1789 |
{
|
|
|
1790 |
const lame_internal_flags *gfc;
|
|
|
1791 |
int i;
|
|
|
1792 |
|
|
|
1793 |
if (NULL == bitrate_kbps)
|
|
|
1794 |
return;
|
|
|
1795 |
if (NULL == gfp)
|
|
|
1796 |
return;
|
|
|
1797 |
gfc = gfp->internal_flags;
|
|
|
1798 |
if (NULL == gfc)
|
|
|
1799 |
return;
|
|
|
1800 |
|
|
|
1801 |
for (i = 0; i < 14; i++)
|
|
|
1802 |
bitrate_kbps[i] = bitrate_table[gfp->version][i + 1];
|
|
|
1803 |
}
|
|
|
1804 |
|
|
|
1805 |
|
|
|
1806 |
|
|
|
1807 |
void
|
|
|
1808 |
lame_stereo_mode_hist(const lame_global_flags * const gfp, int stmode_count[4])
|
|
|
1809 |
{
|
|
|
1810 |
const lame_internal_flags *gfc;
|
|
|
1811 |
int i;
|
|
|
1812 |
|
|
|
1813 |
if (NULL == stmode_count)
|
|
|
1814 |
return;
|
|
|
1815 |
if (NULL == gfp)
|
|
|
1816 |
return;
|
|
|
1817 |
gfc = gfp->internal_flags;
|
|
|
1818 |
if (NULL == gfc)
|
|
|
1819 |
return;
|
|
|
1820 |
|
|
|
1821 |
for (i = 0; i < 4; i++) {
|
|
|
1822 |
int j, sum = 0;
|
|
|
1823 |
for (j = 0; j < 14; j++)
|
|
|
1824 |
sum += gfc->bitrate_stereoMode_Hist[j + 1][i];
|
|
|
1825 |
stmode_count[i] = sum;
|
|
|
1826 |
}
|
|
|
1827 |
}
|
|
|
1828 |
|
|
|
1829 |
|
|
|
1830 |
|
|
|
1831 |
void
|
|
|
1832 |
lame_bitrate_stereo_mode_hist(const lame_global_flags * const gfp,
|
|
|
1833 |
int bitrate_stmode_count[14][4])
|
|
|
1834 |
{
|
|
|
1835 |
const lame_internal_flags *gfc;
|
|
|
1836 |
int i;
|
|
|
1837 |
int j;
|
|
|
1838 |
|
|
|
1839 |
if (NULL == bitrate_stmode_count)
|
|
|
1840 |
return;
|
|
|
1841 |
if (NULL == gfp)
|
|
|
1842 |
return;
|
|
|
1843 |
gfc = gfp->internal_flags;
|
|
|
1844 |
if (NULL == gfc)
|
|
|
1845 |
return;
|
|
|
1846 |
|
|
|
1847 |
for (j = 0; j < 14; j++)
|
|
|
1848 |
for (i = 0; i < 4; i++)
|
|
|
1849 |
bitrate_stmode_count[j][i] = gfc->bitrate_stereoMode_Hist[j + 1][i];
|
|
|
1850 |
}
|
|
|
1851 |
|
|
|
1852 |
/* end of lame.c */
|