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llvm / llvm-test-suite / 271b87d2637e0b49b1f87a70be3473181c4cf3ac / . / MultiSource / Benchmarks / MiBench / consumer-lame / lame.c
blob: eb1c260fbddae862b183fd3d876015081610c3ab [file] [log] [blame]
/*
* 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.
*/
#include <assert.h>
#ifdef HAVEGTK
#include "gtkanal.h"
#include <gtk/gtk.h>
#endif
#include "lame.h"
#include "util.h"
#include "timestatus.h"
#include "psymodel.h"
#include "newmdct.h"
#include "quantize.h"
#include "quantize-pvt.h"
#include "l3bitstream.h"
#include "formatBitstream.h"
#include "version.h"
#include "VbrTag.h"
#include "id3tag.h"
#include "tables.h"
#include "brhist.h"
#include "get_audio.h"
#ifdef __riscos__
#include "asmstuff.h"
#endif
/* Global variable definitions for lame.c */
static Bit_stream_struc bs;
static III_side_info_t l3_side;
#define MFSIZE (1152+1152+ENCDELAY-MDCTDELAY)
static short int mfbuf[2][MFSIZE];
static int mf_size;
static int mf_samples_to_encode;
static const char *mybasename(const char *str) {
const char *base = strrchr(str, '/');
return base ? base+1 : str;
}
/********************************************************************
* initialize internal params based on data in gf
* (globalflags struct filled in by calling program)
*
********************************************************************/
void lame_init_params(lame_global_flags *gfp)
{
int i;
FLOAT compression_ratio;
memset(&bs, 0, sizeof(Bit_stream_struc));
memset(&l3_side,0x00,sizeof(III_side_info_t));
gfp->frameNum=0;
InitFormatBitStream();
if (gfp->num_channels==1) {
gfp->mode = MPG_MD_MONO;
}
gfp->stereo=2;
if (gfp->mode == MPG_MD_MONO) gfp->stereo=1;
#ifdef BRHIST
if (gfp->silent) {
disp_brhist=0; /* turn of VBR historgram */
}
if (!gfp->VBR) {
disp_brhist=0; /* turn of VBR historgram */
}
#endif
/* set the output sampling rate, and resample options if necessary
samplerate = input sample rate
resamplerate = ouput sample rate
*/
if (gfp->out_samplerate==0) {
/* user did not specify output sample rate */
gfp->out_samplerate=gfp->in_samplerate; /* default */
/* if resamplerate is not valid, find a valid value */
if (gfp->out_samplerate>=48000) gfp->out_samplerate=48000;
else if (gfp->out_samplerate>=44100) gfp->out_samplerate=44100;
else if (gfp->out_samplerate>=32000) gfp->out_samplerate=32000;
else if (gfp->out_samplerate>=24000) gfp->out_samplerate=24000;
else if (gfp->out_samplerate>=22050) gfp->out_samplerate=22050;
else gfp->out_samplerate=16000;
if (gfp->brate>0) {
/* check if user specified bitrate requires downsampling */
compression_ratio = gfp->out_samplerate*16*gfp->stereo/(1000.0*gfp->brate);
if (!gfp->VBR && compression_ratio > 13 ) {
/* automatic downsample, if possible */
gfp->out_samplerate = (10*1000.0*gfp->brate)/(16*gfp->stereo);
if (gfp->out_samplerate<=16000) gfp->out_samplerate=16000;
else if (gfp->out_samplerate<=22050) gfp->out_samplerate=22050;
else if (gfp->out_samplerate<=24000) gfp->out_samplerate=24000;
else if (gfp->out_samplerate<=32000) gfp->out_samplerate=32000;
else if (gfp->out_samplerate<=44100) gfp->out_samplerate=44100;
else gfp->out_samplerate=48000;
}
}
}
gfp->mode_gr = (gfp->out_samplerate <= 24000) ? 1 : 2; /* mode_gr = 2 */
gfp->encoder_delay = ENCDELAY;
gfp->framesize = gfp->mode_gr*576;
if (gfp->brate==0) { /* user didn't specify a bitrate, use default */
gfp->brate=128;
if (gfp->mode_gr==1) gfp->brate=64;
}
gfp->resample_ratio=1;
if (gfp->out_samplerate != gfp->in_samplerate) gfp->resample_ratio = (FLOAT)gfp->in_samplerate/(FLOAT)gfp->out_samplerate;
/* estimate total frames. must be done after setting sampling rate so
* we know the framesize. */
gfp->totalframes=0;
gfp->totalframes = 2+ gfp->num_samples/(gfp->resample_ratio*gfp->framesize);
/* 44.1kHz at 56kbs/channel: compression factor of 12.6
44.1kHz at 64kbs/channel: compression factor of 11.025
44.1kHz at 80kbs/channel: compression factor of 8.82
22.05kHz at 24kbs: 14.7
22.05kHz at 32kbs: 11.025
22.05kHz at 40kbs: 8.82
16kHz at 16kbs: 16.0
16kHz at 24kbs: 10.7
compression_ratio
11 .70?
12 sox resample .66
14.7 sox resample .45
*/
if (gfp->brate >= 320) gfp->VBR=0; /* dont bother with VBR at 320kbs */
compression_ratio = gfp->out_samplerate*16*gfp->stereo/(1000.0*gfp->brate);
/* for VBR, take a guess at the compression_ratio */
/* VBR_q compression like
0 4.4 320kbs
1 5.4 256kbs
3 7.4 192kbs
4 8.8 160kbs
6 10.4 128kbs
*/
if (gfp->VBR && compression_ratio>11) {
compression_ratio = 4.4 + gfp->VBR_q;
}
/* At higher quality (lower compression) use STEREO instead of JSTEREO.
* (unless the user explicitly specified a mode ) */
if ( (!gfp->mode_fixed) && (gfp->mode !=MPG_MD_MONO)) {
if (compression_ratio < 9 ) {
gfp->mode = MPG_MD_STEREO;
}
}
/****************************************************************/
/* if a filter has not been enabled, see if we should add one: */
/****************************************************************/
if (gfp->lowpassfreq == 0) {
/* If the user has not selected their own filter, add a lowpass
* filter based on the compression ratio. Formula based on
44.1 /160 4.4x
44.1 /128 5.5x keep all bands
44.1 /96kbs 7.3x keep band 28
44.1 /80kbs 8.8x keep band 25
44.1khz/64kbs 11x keep band 21 22?
16khz/24kbs 10.7x keep band 21
22kHz/32kbs 11x keep band ?
22kHz/24kbs 14.7x keep band 16
16 16 16x keep band 14
*/
/* Should we use some lowpass filters? */
int band = 1+floor(.5 + 14-18*log(compression_ratio/16.0));
if (band < 31) {
gfp->lowpass1 = band/31.0;
gfp->lowpass2 = band/31.0;
}
}
/****************************************************************/
/* apply user driven filters*/
/****************************************************************/
if ( gfp->highpassfreq > 0 ) {
gfp->highpass1 = 2.0*gfp->highpassfreq/gfp->out_samplerate; /* will always be >=0 */
if ( gfp->highpasswidth >= 0 ) {
gfp->highpass2 = 2.0*(gfp->highpassfreq+gfp->highpasswidth)/gfp->out_samplerate;
} else {
/* 15% above on default */
/* gfp->highpass2 = 1.15*2.0*gfp->highpassfreq/gfp->out_samplerate; */
gfp->highpass2 = 1.00*2.0*gfp->highpassfreq/gfp->out_samplerate;
}
gfp->highpass1 = Min( 1, gfp->highpass1 );
gfp->highpass2 = Min( 1, gfp->highpass2 );
}
if ( gfp->lowpassfreq > 0 ) {
gfp->lowpass2 = 2.0*gfp->lowpassfreq/gfp->out_samplerate; /* will always be >=0 */
if ( gfp->lowpasswidth >= 0 ) {
gfp->lowpass1 = 2.0*(gfp->lowpassfreq-gfp->lowpasswidth)/gfp->out_samplerate;
if ( gfp->lowpass1 < 0 ) { /* has to be >= 0 */
gfp->lowpass1 = 0;
}
} else {
/* 15% below on default */
/* gfp->lowpass1 = 0.85*2.0*gfp->lowpassfreq/gfp->out_samplerate; */
gfp->lowpass1 = 1.00*2.0*gfp->lowpassfreq/gfp->out_samplerate;
}
gfp->lowpass1 = Min( 1, gfp->lowpass1 );
gfp->lowpass2 = Min( 1, gfp->lowpass2 );
}
/***************************************************************/
/* compute info needed for polyphase filter */
/***************************************************************/
if (gfp->filter_type==0) {
int band,maxband,minband;
FLOAT8 amp,freq;
if (gfp->lowpass1 > 0) {
minband=999;
maxband=-1;
for (band=0; band <=31 ; ++band) {
freq = band/31.0;
amp = 1;
/* this band and above will be zeroed: */
if (freq >= gfp->lowpass2) {
gfp->lowpass_band= Min(gfp->lowpass_band,band);
amp=0;
}
if (gfp->lowpass1 < freq && freq < gfp->lowpass2) {
minband = Min(minband,band);
maxband = Max(maxband,band);
amp = cos((PI/2)*(gfp->lowpass1-freq)/(gfp->lowpass2-gfp->lowpass1));
}
/* printf("lowpass band=%i amp=%f \n",band,amp);*/
}
/* compute the *actual* transition band implemented by the polyphase filter */
if (minband==999) gfp->lowpass1 = (gfp->lowpass_band-.75)/31.0;
else gfp->lowpass1 = (minband-.75)/31.0;
gfp->lowpass2 = gfp->lowpass_band/31.0;
}
/* make sure highpass filter is within 90% of whan the effective highpass
* frequency will be */
if (gfp->highpass2 > 0)
if (gfp->highpass2 < .9*(.75/31.0) ) {
gfp->highpass1=0; gfp->highpass2=0;
fprintf(stderr,"Warning: highpass filter disabled. highpass frequency to small\n");
}
if (gfp->highpass2 > 0) {
minband=999;
maxband=-1;
for (band=0; band <=31; ++band) {
freq = band/31.0;
amp = 1;
/* this band and below will be zereod */
if (freq <= gfp->highpass1) {
gfp->highpass_band = Max(gfp->highpass_band,band);
amp=0;
}
if (gfp->highpass1 < freq && freq < gfp->highpass2) {
minband = Min(minband,band);
maxband = Max(maxband,band);
amp = cos((PI/2)*(gfp->highpass2-freq)/(gfp->highpass2-gfp->highpass1));
}
/* printf("highpass band=%i amp=%f \n",band,amp);*/
}
/* compute the *actual* transition band implemented by the polyphase filter */
gfp->highpass1 = gfp->highpass_band/31.0;
if (maxband==-1) gfp->highpass2 = (gfp->highpass_band+.75)/31.0;
else gfp->highpass2 = (maxband+.75)/31.0;
}
/*
printf("lowpass band with amp=0: %i \n",gfp->lowpass_band);
printf("highpass band with amp=0: %i \n",gfp->highpass_band);
*/
}
/***************************************************************/
/* compute info needed for FIR filter */
/***************************************************************/
if (gfp->filter_type==1) {
}
gfp->mode_ext=MPG_MD_LR_LR;
gfp->stereo = (gfp->mode == MPG_MD_MONO) ? 1 : 2;
gfp->samplerate_index = SmpFrqIndex((long)gfp->out_samplerate, &gfp->version);
if( gfp->samplerate_index < 0) {
display_bitrates(stderr);
exit(1);
}
if( (gfp->bitrate_index = BitrateIndex(gfp->brate, gfp->version,gfp->out_samplerate)) < 0) {
display_bitrates(stderr);
exit(1);
}
/* choose a min/max bitrate for VBR */
if (gfp->VBR) {
/* if the user didn't specify VBR_max_bitrate: */
if (0==gfp->VBR_max_bitrate_kbps) {
/* default max bitrate is 256kbs */
/* we do not normally allow 320bps frams with VBR, unless: */
gfp->VBR_max_bitrate=13; /* default: allow 256kbs */
if (gfp->VBR_min_bitrate_kbps>=256) gfp->VBR_max_bitrate=14;
if (gfp->VBR_q == 0) gfp->VBR_max_bitrate=14; /* allow 320kbs */
if (gfp->VBR_q >= 4) gfp->VBR_max_bitrate=12; /* max = 224kbs */
if (gfp->VBR_q >= 8) gfp->VBR_max_bitrate=9; /* low quality, max = 128kbs */
}else{
if( (gfp->VBR_max_bitrate = BitrateIndex(gfp->VBR_max_bitrate_kbps, gfp->version,gfp->out_samplerate)) < 0) {
display_bitrates(stderr);
exit(1);
}
}
if (0==gfp->VBR_min_bitrate_kbps) {
gfp->VBR_min_bitrate=1; /* 32 kbps */
}else{
if( (gfp->VBR_min_bitrate = BitrateIndex(gfp->VBR_min_bitrate_kbps, gfp->version,gfp->out_samplerate)) < 0) {
display_bitrates(stderr);
exit(1);
}
}
}
if (gfp->VBR) gfp->quality=Min(gfp->quality,2); /* always use quality <=2 with VBR */
/* dont allow forced mid/side stereo for mono output */
if (gfp->mode == MPG_MD_MONO) gfp->force_ms=0;
/* Do not write VBR tag if VBR flag is not specified */
if (gfp->VBR==0) gfp->bWriteVbrTag=0;
/* some file options not allowed if output is: not specified or stdout */
if (gfp->outPath!=NULL && gfp->outPath[0]=='-' ) {
gfp->bWriteVbrTag=0; /* turn off VBR tag */
}
if (gfp->outPath==NULL || gfp->outPath[0]=='-' ) {
id3tag.used=0; /* turn of id3 tagging */
}
if (gfp->gtkflag) {
gfp->bWriteVbrTag=0; /* disable Xing VBR tag */
}
init_bit_stream_w(&bs);
/* set internal feature flags. USER should not access these since
* some combinations will produce strange results */
/* no psymodel, no noise shaping */
if (gfp->quality==9) {
gfp->filter_type=0;
gfp->psymodel=0;
gfp->quantization=0;
gfp->noise_shaping=0;
gfp->noise_shaping_stop=0;
gfp->use_best_huffman=0;
}
if (gfp->quality==8) gfp->quality=7;
/* use psymodel (for short block and m/s switching), but no noise shapping */
if (gfp->quality==7) {
gfp->filter_type=0;
gfp->psymodel=1;
gfp->quantization=0;
gfp->noise_shaping=0;
gfp->noise_shaping_stop=0;
gfp->use_best_huffman=0;
}
if (gfp->quality==6) gfp->quality=5;
if (gfp->quality==5) {
/* the default */
gfp->filter_type=0;
gfp->psymodel=1;
gfp->quantization=0;
gfp->noise_shaping=1;
gfp->noise_shaping_stop=0;
gfp->use_best_huffman=0;
}
if (gfp->quality==4) gfp->quality=2;
if (gfp->quality==3) gfp->quality=2;
if (gfp->quality==2) {
gfp->filter_type=0;
gfp->psymodel=1;
gfp->quantization=1;
gfp->noise_shaping=1;
gfp->noise_shaping_stop=0;
gfp->use_best_huffman=1;
}
if (gfp->quality==1) {
gfp->filter_type=0;
gfp->psymodel=1;
gfp->quantization=1;
gfp->noise_shaping=1;
gfp->noise_shaping_stop=1;
gfp->use_best_huffman=1;
}
if (gfp->quality==0) {
/* 0..1 quality */
gfp->filter_type=1; /* not yet coded */
gfp->psymodel=1;
gfp->quantization=1;
gfp->noise_shaping=3; /* not yet coded */
gfp->noise_shaping_stop=2; /* not yet coded */
gfp->use_best_huffman=2; /* not yet coded */
exit(-99);
}
for (i = 0; i < SBMAX_l + 1; i++) {
scalefac_band.l[i] =
sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].l[i];
}
for (i = 0; i < SBMAX_s + 1; i++) {
scalefac_band.s[i] =
sfBandIndex[gfp->samplerate_index + (gfp->version * 3)].s[i];
}
if (gfp->bWriteVbrTag)
{
/* Write initial VBR Header to bitstream */
InitVbrTag(&bs,1-gfp->version,gfp->mode,gfp->samplerate_index);
}
#ifdef HAVEGTK
gtkflag=gfp->gtkflag;
#endif
#ifdef BRHIST
if (gfp->VBR) {
if (disp_brhist)
brhist_init(gfp,1, 14);
} else
disp_brhist = 0;
#endif
return;
}
/************************************************************************
*
* print_config
*
* PURPOSE: Prints the encoding parameters used
*
************************************************************************/
void lame_print_config(lame_global_flags *gfp)
{
static const char *mode_names[4] = { "stereo", "j-stereo", "dual-ch", "single-ch" };
FLOAT out_samplerate=gfp->out_samplerate/1000.0;
FLOAT in_samplerate = gfp->resample_ratio*out_samplerate;
FLOAT compression=
(FLOAT)(gfp->stereo*16*out_samplerate)/(FLOAT)(gfp->brate);
lame_print_version(stderr);
if (gfp->num_channels==2 && gfp->stereo==1) {
fprintf(stderr, "Autoconverting from stereo to mono. Setting encoding to mono mode.\n");
}
if (gfp->resample_ratio!=1) {
fprintf(stderr,"Resampling: input=%ikHz output=%ikHz\n",
(int)in_samplerate,(int)out_samplerate);
}
if (gfp->highpass2>0.0)
fprintf(stderr, "Using polyphase highpass filter, transition band: %.0f Hz - %.0f Hz\n",
gfp->highpass1*out_samplerate*500,
gfp->highpass2*out_samplerate*500);
if (gfp->lowpass1>0.0)
fprintf(stderr, "Using polyphase lowpass filter, transition band: %.0f Hz - %.0f Hz\n",
gfp->lowpass1*out_samplerate*500,
gfp->lowpass2*out_samplerate*500);
if (gfp->gtkflag) {
fprintf(stderr, "Analyzing %s \n",gfp->inPath);
}
else {
fprintf(stderr, "Encoding %s to %s\n",
(strcmp(gfp->inPath, "-")? mybasename(gfp->inPath) : "stdin"),
(strcmp(gfp->outPath, "-")? mybasename(gfp->outPath) : "stdout"));
if (gfp->VBR)
fprintf(stderr, "Encoding as %.1fkHz VBR(q=%i) %s MPEG%i LayerIII qval=%i\n",
gfp->out_samplerate/1000.0,
gfp->VBR_q,mode_names[gfp->mode],2-gfp->version,gfp->quality);
else
fprintf(stderr, "Encoding as %.1f kHz %d kbps %s MPEG%i LayerIII (%4.1fx) qval=%i\n",
gfp->out_samplerate/1000.0,gfp->brate,
mode_names[gfp->mode],2-gfp->version,compression,gfp->quality);
}
fflush(stderr);
}
/************************************************************************
*
* encodeframe() Layer 3
*
* encode a single frame
*
************************************************************************
lame_encode_frame()
gr 0 gr 1
inbuf: |--------------|---------------|-------------|
MDCT output: |--------------|---------------|-------------|
FFT's <---------1024---------->
<---------1024-------->
inbuf = buffer of PCM data size=MP3 framesize
encoder acts on inbuf[ch][0], but output is delayed by MDCTDELAY
so the MDCT coefficints are from inbuf[ch][-MDCTDELAY]
psy-model FFT has a 1 granule day, so we feed it data for the next granule.
FFT is centered over granule: 224+576+224
So FFT starts at: 576-224-MDCTDELAY
MPEG2: FFT ends at: BLKSIZE+576-224-MDCTDELAY
MPEG1: FFT ends at: BLKSIZE+2*576-224-MDCTDELAY (1904)
FFT starts at 576-224-MDCTDELAY (304) = 576-FFTOFFSET
*/
int lame_encode_frame(lame_global_flags *gfp,
short int inbuf_l[],short int inbuf_r[],
int mf_size,char *mp3buf, int mp3buf_size)
{
static unsigned long frameBits;
static unsigned long bitsPerSlot;
static FLOAT8 frac_SpF;
static FLOAT8 slot_lag;
static unsigned long sentBits = 0;
FLOAT8 xr[2][2][576];
int l3_enc[2][2][576];
int mp3count;
III_psy_ratio masking_ratio[2][2]; /*LR ratios */
III_psy_ratio masking_MS_ratio[2][2]; /*MS ratios */
III_psy_ratio (*masking)[2][2]; /*LR ratios and MS ratios*/
III_scalefac_t scalefac[2][2];
short int *inbuf[2];
typedef FLOAT8 pedata[2][2];
pedata pe,pe_MS;
pedata *pe_use;
int ch,gr,mean_bits;
int bitsPerFrame;
int check_ms_stereo;
static FLOAT8 ms_ratio[2]={0,0};
FLOAT8 ms_ratio_next=0;
FLOAT8 ms_ratio_prev=0;
static FLOAT8 ms_ener_ratio[2]={0,0};
memset((char *) masking_ratio, 0, sizeof(masking_ratio));
memset((char *) masking_MS_ratio, 0, sizeof(masking_MS_ratio));
memset((char *) scalefac, 0, sizeof(scalefac));
inbuf[0]=inbuf_l;
inbuf[1]=inbuf_r;
gfp->mode_ext = MPG_MD_LR_LR;
if (gfp->frameNum==0 ) {
/* Figure average number of 'slots' per frame. */
FLOAT8 avg_slots_per_frame;
FLOAT8 sampfreq = gfp->out_samplerate/1000.0;
int bit_rate = gfp->brate;
sentBits = 0;
bitsPerSlot = 8;
avg_slots_per_frame = (bit_rate*gfp->framesize) /
(sampfreq* bitsPerSlot);
/* -f fast-math option causes some strange rounding here, be carefull: */
frac_SpF = avg_slots_per_frame - floor(avg_slots_per_frame + 1e-9);
if (fabs(frac_SpF) < 1e-9) frac_SpF = 0;
slot_lag = -frac_SpF;
gfp->padding = 1;
if (frac_SpF==0) gfp->padding = 0;
/* check FFT will not use a negative starting offset */
assert(576>=FFTOFFSET);
/* check if we have enough data for FFT */
assert(mf_size>=(BLKSIZE+gfp->framesize-FFTOFFSET));
}
/********************** padding *****************************/
switch (gfp->padding_type) {
case 0:
gfp->padding=0;
break;
case 1:
gfp->padding=1;
break;
case 2:
default:
if (gfp->VBR) {
gfp->padding=0;
} else {
if (gfp->disable_reservoir) {
gfp->padding = 0;
/* if the user specified --nores, dont very gfp->padding either */
/* tiny changes in frac_SpF rounding will cause file differences */
}else{
if (frac_SpF != 0) {
if (slot_lag > (frac_SpF-1.0) ) {
slot_lag -= frac_SpF;
gfp->padding = 0;
}
else {
gfp->padding = 1;
slot_lag += (1-frac_SpF);
}
}
}
}
}
/********************** status display *****************************/
if (!gfp->gtkflag && !gfp->silent) {
int mod = gfp->version == 0 ? 200 : 50;
if (gfp->frameNum%mod==0) {
timestatus(gfp->out_samplerate,gfp->frameNum,gfp->totalframes,gfp->framesize);
#ifdef BRHIST
if (disp_brhist)
{
brhist_add_count();
brhist_disp();
}
#endif
}
}
if (gfp->psymodel) {
/* psychoacoustic model
* psy model has a 1 granule (576) delay that we must compensate for
* (mt 6/99).
*/
short int *bufp[2]; /* address of beginning of left & right granule */
int blocktype[2];
ms_ratio_prev=ms_ratio[gfp->mode_gr-1];
for (gr=0; gr < gfp->mode_gr ; gr++) {
for ( ch = 0; ch < gfp->stereo; ch++ )
bufp[ch] = &inbuf[ch][576 + gr*576-FFTOFFSET];
L3psycho_anal( gfp,bufp, gr,
&ms_ratio[gr],&ms_ratio_next,&ms_ener_ratio[gr],
masking_ratio, masking_MS_ratio,
pe[gr],pe_MS[gr],blocktype);
for ( ch = 0; ch < gfp->stereo; ch++ )
l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch];
}
}else{
for (gr=0; gr < gfp->mode_gr ; gr++)
for ( ch = 0; ch < gfp->stereo; ch++ ) {
l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE;
pe[gr][ch]=700;
}
}
/* block type flags */
for( gr = 0; gr < gfp->mode_gr; gr++ ) {
for ( ch = 0; ch < gfp->stereo; ch++ ) {
gr_info *cod_info = &l3_side.gr[gr].ch[ch].tt;
cod_info->mixed_block_flag = 0; /* never used by this model */
if (cod_info->block_type == NORM_TYPE )
cod_info->window_switching_flag = 0;
else
cod_info->window_switching_flag = 1;
}
}
/* polyphase filtering / mdct */
mdct_sub48(gfp,inbuf[0], inbuf[1], xr, &l3_side);
/* use m/s gfp->stereo? */
check_ms_stereo = (gfp->mode == MPG_MD_JOINT_STEREO);
if (check_ms_stereo) {
/* make sure block type is the same in each channel */
check_ms_stereo =
(l3_side.gr[0].ch[0].tt.block_type==l3_side.gr[0].ch[1].tt.block_type) &&
(l3_side.gr[1].ch[0].tt.block_type==l3_side.gr[1].ch[1].tt.block_type);
}
if (check_ms_stereo) {
/* ms_ratio = is like the ratio of side_energy/total_energy */
FLOAT8 ms_ratio_ave,ms_ener_ratio_ave;
/* ms_ratio_ave = .5*(ms_ratio[0] + ms_ratio[1]);*/
ms_ratio_ave = .25*(ms_ratio[0] + ms_ratio[1]+
ms_ratio_prev + ms_ratio_next);
ms_ener_ratio_ave = .5*(ms_ener_ratio[0]+ms_ener_ratio[1]);
if ( ms_ratio_ave <.35 /*&& ms_ener_ratio_ave<.75*/ ) gfp->mode_ext = MPG_MD_MS_LR;
}
if (gfp->force_ms) gfp->mode_ext = MPG_MD_MS_LR;
#ifdef HAVEGTK
if (gfp->gtkflag) {
int j;
for ( gr = 0; gr < gfp->mode_gr; gr++ ) {
for ( ch = 0; ch < gfp->stereo; ch++ ) {
pinfo->ms_ratio[gr]=ms_ratio[gr];
pinfo->ms_ener_ratio[gr]=ms_ener_ratio[gr];
pinfo->blocktype[gr][ch]=
l3_side.gr[gr].ch[ch].tt.block_type;
for ( j = 0; j < 576; j++ ) pinfo->xr[gr][ch][j]=xr[gr][ch][j];
/* if MS stereo, switch to MS psy data */
if (gfp->mode_ext==MPG_MD_MS_LR) {
pinfo->pe[gr][ch]=pinfo->pe[gr][ch+2];
pinfo->ers[gr][ch]=pinfo->ers[gr][ch+2];
memcpy(pinfo->energy[gr][ch],pinfo->energy[gr][ch+2],
sizeof(pinfo->energy[gr][ch]));
}
}
}
}
#endif
/* bit and noise allocation */
if (MPG_MD_MS_LR == gfp->mode_ext) {
masking = &masking_MS_ratio; /* use MS masking */
pe_use=&pe_MS;
} else {
masking = &masking_ratio; /* use LR masking */
pe_use=&pe;
}
/*
VBR_iteration_loop_new( gfp,*pe_use, ms_ratio, xr, masking, &l3_side, l3_enc,
&scalefac);
*/
if (gfp->VBR) {
VBR_iteration_loop( gfp,*pe_use, ms_ratio, xr, *masking, &l3_side, l3_enc,
scalefac);
}else{
iteration_loop( gfp,*pe_use, ms_ratio, xr, *masking, &l3_side, l3_enc,
scalefac);
}
#ifdef BRHIST
brhist_temp[gfp->bitrate_index]++;
#endif
/* write the frame to the bitstream */
getframebits(gfp,&bitsPerFrame,&mean_bits);
III_format_bitstream( gfp,bitsPerFrame, l3_enc, &l3_side,
scalefac, &bs);
frameBits = bs.totbit - sentBits;
if ( frameBits % bitsPerSlot ) /* a program failure */
fprintf( stderr, "Sent %ld bits = %ld slots plus %ld\n",
frameBits, frameBits/bitsPerSlot,
frameBits%bitsPerSlot );
sentBits += frameBits;
/* copy mp3 bit buffer into array */
mp3count = copy_buffer(mp3buf,mp3buf_size,&bs);
if (gfp->bWriteVbrTag) AddVbrFrame((int)(sentBits/8));
#ifdef HAVEGTK
if (gfp->gtkflag) {
int j;
for ( ch = 0; ch < gfp->stereo; ch++ ) {
for ( j = 0; j < FFTOFFSET; j++ )
pinfo->pcmdata[ch][j] = pinfo->pcmdata[ch][j+gfp->framesize];
for ( j = FFTOFFSET; j < 1600; j++ ) {
pinfo->pcmdata[ch][j] = inbuf[ch][j-FFTOFFSET];
}
}
}
#endif
gfp->frameNum++;
return mp3count;
}
int fill_buffer_resample(lame_global_flags *gfp,short int *outbuf,int desired_len,
short int *inbuf,int len,int *num_used,int ch) {
static FLOAT8 itime[2];
#define OLDBUFSIZE 5
static short int inbuf_old[2][OLDBUFSIZE];
static int init[2]={0,0};
int i,j=0,k,linear,value;
if (gfp->frameNum==0 && !init[ch]) {
init[ch]=1;
itime[ch]=0;
memset((char *) inbuf_old[ch], 0, sizeof(short int)*OLDBUFSIZE);
}
if (gfp->frameNum!=0) init[ch]=0; /* reset, for next time framenum=0 */
/* if downsampling by an integer multiple, use linear resampling,
* otherwise use quadratic */
linear = ( fabs(gfp->resample_ratio - floor(.5+gfp->resample_ratio)) < .0001 );
/* time of j'th element in inbuf = itime + j/ifreq; */
/* time of k'th element in outbuf = j/ofreq */
for (k=0;k<desired_len;k++) {
int y0,y1,y2,y3;
FLOAT8 x0,x1,x2,x3;
FLOAT8 time0;
time0 = k*gfp->resample_ratio; /* time of k'th output sample */
j = floor( time0 -itime[ch] );
/* itime[ch] + j; */ /* time of j'th input sample */
if (j+2 >= len) break; /* not enough data in input buffer */
x1 = time0-(itime[ch]+j);
x2 = x1-1;
y1 = (j<0) ? inbuf_old[ch][OLDBUFSIZE+j] : inbuf[j];
y2 = ((1+j)<0) ? inbuf_old[ch][OLDBUFSIZE+1+j] : inbuf[1+j];
/* linear resample */
if (linear) {
outbuf[k] = floor(.5 + (y2*x1-y1*x2) );
} else {
/* quadratic */
x0 = x1+1;
x3 = x1-2;
y0 = ((j-1)<0) ? inbuf_old[ch][OLDBUFSIZE+(j-1)] : inbuf[j-1];
y3 = ((j+2)<0) ? inbuf_old[ch][OLDBUFSIZE+(j+2)] : inbuf[j+2];
value = floor(.5 +
-y0*x1*x2*x3/6 + y1*x0*x2*x3/2 - y2*x0*x1*x3/2 +y3*x0*x1*x2/6
);
if (value > 32767) outbuf[k]=32767;
else if (value < -32767) outbuf[k]=-32767;
else outbuf[k]=value;
/*
printf("k=%i new=%i [ %i %i %i %i ]\n",k,outbuf[k],
y0,y1,y2,y3);
*/
}
}
/* k = number of samples added to outbuf */
/* last k sample used data from j,j+1, or j+1 overflowed buffer */
/* remove num_used samples from inbuf: */
*num_used = Min(len,j+2);
itime[ch] += *num_used - k*gfp->resample_ratio;
for (i=0;i<OLDBUFSIZE;i++)
inbuf_old[ch][i]=inbuf[*num_used + i -OLDBUFSIZE];
return k;
}
int fill_buffer(lame_global_flags *gfp,short int *outbuf,int desired_len,short int *inbuf,int len) {
int j;
j=Min(desired_len,len);
memcpy( (char *) outbuf,(char *)inbuf,sizeof(short int)*j);
return j;
}
/*
* 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(lame_global_flags *gfp,
short int buffer_l[], short int buffer_r[],int nsamples,
char *mp3buf, int mp3buf_size)
{
static int frame_buffered=0;
int mp3size=0,ret,i,ch,mf_needed;
short int *in_buffer[2];
in_buffer[0] = buffer_l;
in_buffer[1] = buffer_r;
/* some sanity checks */
assert(ENCDELAY>=MDCTDELAY);
assert(BLKSIZE-FFTOFFSET >= 0);
mf_needed = BLKSIZE+gfp->framesize-FFTOFFSET;
assert(MFSIZE>=mf_needed);
/* 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.
*/
if (gfp->frameNum==0 && !frame_buffered) {
memset((char *) mfbuf, 0, sizeof(mfbuf));
frame_buffered=1;
mf_samples_to_encode = ENCDELAY+288;
mf_size=ENCDELAY-MDCTDELAY; /* we pad input with this many 0's */
}
if (gfp->frameNum==1) {
/* reset, for the next time frameNum==0 */
frame_buffered=0;
}
if (gfp->num_channels==2 && gfp->stereo==1) {
/* downsample to mono */
for (i=0; i<nsamples; ++i) {
in_buffer[0][i]=((int)in_buffer[0][i]+(int)in_buffer[1][i])/2;
in_buffer[1][i]=0;
}
}
while (nsamples > 0) {
int n_in=0;
int n_out=0;
/* copy in new samples */
for (ch=0; ch<gfp->stereo; ch++) {
if (gfp->resample_ratio!=1) {
n_out=fill_buffer_resample(gfp,&mfbuf[ch][mf_size],gfp->framesize,
in_buffer[ch],nsamples,&n_in,ch);
} else {
n_out=fill_buffer(gfp,&mfbuf[ch][mf_size],gfp->framesize,in_buffer[ch],nsamples);
n_in = n_out;
}
in_buffer[ch] += n_in;
}
nsamples -= n_in;
mf_size += n_out;
assert(mf_size<=MFSIZE);
mf_samples_to_encode += n_out;
if (mf_size >= mf_needed) {
/* encode the frame */
ret = lame_encode_frame(gfp,mfbuf[0],mfbuf[1],mf_size,mp3buf,mp3buf_size);
if (ret == -1) {
/* fatel error: mp3buffer was too small */
return -1;
}
mp3buf += ret;
mp3size += ret;
/* shift out old samples */
mf_size -= gfp->framesize;
mf_samples_to_encode -= gfp->framesize;
for (ch=0; ch<gfp->stereo; ch++)
for (i=0; i<mf_size; i++)
mfbuf[ch][i]=mfbuf[ch][i+gfp->framesize];
}
}
assert(nsamples==0);
return mp3size;
}
int lame_encode_buffer_interleaved(lame_global_flags *gfp,
short int buffer[], int nsamples, char *mp3buf, int mp3buf_size)
{
static int frame_buffered=0;
int mp3size=0,ret,i,ch,mf_needed;
/* some sanity checks */
assert(ENCDELAY>=MDCTDELAY);
assert(BLKSIZE-FFTOFFSET >= 0);
mf_needed = BLKSIZE+gfp->framesize-FFTOFFSET;
assert(MFSIZE>=mf_needed);
if (gfp->num_channels == 1) {
return lame_encode_buffer(gfp,buffer, NULL ,nsamples,mp3buf,mp3buf_size);
}
if (gfp->resample_ratio!=1) {
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 -1;
}
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;
}
if (gfp->frameNum==0 && !frame_buffered) {
memset((char *) mfbuf, 0, sizeof(mfbuf));
frame_buffered=1;
mf_samples_to_encode = ENCDELAY+288;
mf_size=ENCDELAY-MDCTDELAY; /* we pad input with this many 0's */
}
if (gfp->frameNum==1) {
/* reset, for the next time frameNum==0 */
frame_buffered=0;
}
if (gfp->num_channels==2 && gfp->stereo==1) {
/* downsample to mono */
for (i=0; i<nsamples; ++i) {
buffer[2*i]=((int)buffer[2*i]+(int)buffer[2*i+1])/2;
buffer[2*i+1]=0;
}
}
while (nsamples > 0) {
int n_out;
/* copy in new samples */
n_out = Min(gfp->framesize,nsamples);
for (i=0; i<n_out; ++i) {
mfbuf[0][mf_size+i]=buffer[2*i];
mfbuf[1][mf_size+i]=buffer[2*i+1];
}
buffer += 2*n_out;
nsamples -= n_out;
mf_size += n_out;
assert(mf_size<=MFSIZE);
mf_samples_to_encode += n_out;
if (mf_size >= mf_needed) {
/* encode the frame */
ret = lame_encode_frame(gfp,mfbuf[0],mfbuf[1],mf_size,mp3buf,mp3buf_size);
if (ret == -1) {
/* fatel error: mp3buffer was too small */
return -1;
}
mp3buf += ret;
mp3size += ret;
/* shift out old samples */
mf_size -= gfp->framesize;
mf_samples_to_encode -= gfp->framesize;
for (ch=0; ch<gfp->stereo; ch++)
for (i=0; i<mf_size; i++)
mfbuf[ch][i]=mfbuf[ch][i+gfp->framesize];
}
}
assert(nsamples==0);
return mp3size;
}
/* old LAME interface */
/* With this interface, it is the users responsibilty to keep track of the
* buffered, unencoded samples. Thus mf_samples_to_encode is not incremented.
*
* lame_encode() is also used to flush the PCM input buffer by
* lame_encode_finish()
*/
int lame_encode(lame_global_flags *gfp, short int in_buffer[2][1152],char *mp3buf,int size){
int imp3,save;
save = mf_samples_to_encode;
imp3= lame_encode_buffer(gfp,in_buffer[0],in_buffer[1],576*gfp->mode_gr,
mp3buf,size);
mf_samples_to_encode = save;
return imp3;
}
#ifdef __FreeBSD__
# include <floatingpoint.h>
#endif
/* initialize mp3 encoder */
void lame_init(lame_global_flags *gfp)
{
/*
* Disable floating point exceptions
*/
#if defined(__FreeBSD__) && defined(FP_X_INV) && defined(FP_X_DZ)
{
/* seet floating point mask to the Linux default */
fp_except_t mask;
mask=fpgetmask();
/* if bit is set, we get SIGFPE on that error! */
fpsetmask(mask & ~(FP_X_INV|FP_X_DZ));
/* fprintf(stderr,"FreeBSD mask is 0x%x\n",mask); */
}
#endif
#if defined(__riscos__) && !defined(ABORTFP)
/* Disable FPE's under RISC OS */
/* if bit is set, we disable trapping that error! */
/* _FPE_IVO : invalid operation */
/* _FPE_DVZ : divide by zero */
/* _FPE_OFL : overflow */
/* _FPE_UFL : underflow */
/* _FPE_INX : inexact */
DisableFPETraps( _FPE_IVO | _FPE_DVZ | _FPE_OFL );
#endif
/*
* Debugging stuff
* The default is to ignore FPE's, unless compiled with -DABORTFP
* so add code below to ENABLE FPE's.
*/
#if defined(ABORTFP) && !defined(__riscos__)
#if defined(_MSC_VER)
{
#include <float.h>
unsigned int mask;
mask=_controlfp( 0, 0 );
mask&=~(_EM_OVERFLOW|_EM_UNDERFLOW|_EM_ZERODIVIDE|_EM_INVALID);
mask=_controlfp( mask, _MCW_EM );
}
#elif defined(__CYGWIN__)
# define _FPU_GETCW(cw) __asm__ ("fnstcw %0" : "=m" (*&cw))
# define _FPU_SETCW(cw) __asm__ ("fldcw %0" : : "m" (*&cw))
# define _EM_INEXACT 0x00000001 /* inexact (precision) */
# define _EM_UNDERFLOW 0x00000002 /* underflow */
# define _EM_OVERFLOW 0x00000004 /* overflow */
# define _EM_ZERODIVIDE 0x00000008 /* zero divide */
# define _EM_INVALID 0x00000010 /* invalid */
{
unsigned int mask;
_FPU_GETCW(mask);
/* Set the FPU control word to abort on most FPEs */
mask &= ~(_EM_UNDERFLOW | _EM_OVERFLOW | _EM_ZERODIVIDE | _EM_INVALID);
_FPU_SETCW(mask);
}
# else
{
# include <fpu_control.h>
#ifndef _FPU_GETCW
#define _FPU_GETCW(cw) __asm__ ("fnstcw %0" : "=m" (*&cw))
#endif
#ifndef _FPU_SETCW
#define _FPU_SETCW(cw) __asm__ ("fldcw %0" : : "m" (*&cw))
#endif
unsigned int mask;
_FPU_GETCW(mask);
/* Set the Linux mask to abort on most FPE's */
/* if bit is set, we _mask_ SIGFPE on that error! */
/* mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM | _FPU_MASK_UM );*/
mask &= ~( _FPU_MASK_IM | _FPU_MASK_ZM | _FPU_MASK_OM );
_FPU_SETCW(mask);
}
#endif
#endif /* ABORTFP && !__riscos__ */
/* Global flags. set defaults here */
gfp->allow_diff_short=0;
gfp->ATHonly=0;
gfp->noATH=0;
gfp->bWriteVbrTag=1;
gfp->cwlimit=0;
gfp->disable_reservoir=0;
gfp->experimentalX = 0;
gfp->experimentalY = 0;
gfp->experimentalZ = 0;
gfp->frameNum=0;
gfp->gtkflag=0;
gfp->quality=5;
gfp->input_format=sf_unknown;
gfp->filter_type=0;
gfp->lowpassfreq=0;
gfp->highpassfreq=0;
gfp->lowpasswidth=-1;
gfp->highpasswidth=-1;
gfp->lowpass1=0;
gfp->lowpass2=0;
gfp->highpass1=0;
gfp->highpass2=0;
gfp->lowpass_band=32;
gfp->highpass_band=-1;
gfp->no_short_blocks=0;
gfp->resample_ratio=1;
gfp->padding_type=2;
gfp->padding=0;
gfp->swapbytes=0;
gfp->silent=0;
gfp->totalframes=0;
gfp->VBR=0;
gfp->VBR_q=4;
gfp->VBR_min_bitrate_kbps=0;
gfp->VBR_max_bitrate_kbps=0;
gfp->VBR_min_bitrate=1;
gfp->VBR_max_bitrate=13;
gfp->version = 1; /* =1 Default: MPEG-1 */
gfp->mode = MPG_MD_JOINT_STEREO;
gfp->mode_fixed=0;
gfp->force_ms=0;
gfp->brate=0;
gfp->copyright=0;
gfp->original=1;
gfp->extension=0;
gfp->error_protection=0;
gfp->emphasis=0;
gfp->in_samplerate=1000*44.1;
gfp->out_samplerate=0;
gfp->num_channels=2;
gfp->num_samples=MAX_U_32_NUM;
gfp->inPath=NULL;
gfp->outPath=NULL;
id3tag.used=0;
}
/*****************************************************************/
/* flush internal mp3 buffers, */
/*****************************************************************/
int lame_encode_finish(lame_global_flags *gfp,char *mp3buffer, int mp3buffer_size)
{
int imp3,mp3count,mp3buffer_size_remaining;
short int buffer[2][1152];
memset((char *)buffer,0,sizeof(buffer));
mp3count = 0;
while (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;
imp3=lame_encode(gfp,buffer,mp3buffer,mp3buffer_size_remaining);
if (imp3 == -1) {
/* fatel error: mp3buffer too small */
desalloc_buffer(&bs); /* Deallocate all buffers */
return -1;
}
mp3buffer += imp3;
mp3count += imp3;
mf_samples_to_encode -= gfp->framesize;
}
gfp->frameNum--;
if (!gfp->gtkflag && !gfp->silent) {
timestatus(gfp->out_samplerate,gfp->frameNum,gfp->totalframes,gfp->framesize);
#ifdef BRHIST
if (disp_brhist)
{
brhist_add_count();
brhist_disp();
brhist_disp_total(gfp);
}
#endif
fprintf(stderr,"\n");
fflush(stderr);
}
III_FlushBitstream();
mp3buffer_size_remaining = mp3buffer_size - mp3count;
/* if user specifed buffer size = 0, dont check size */
if (mp3buffer_size == 0) mp3buffer_size_remaining=0;
imp3= copy_buffer(mp3buffer,mp3buffer_size_remaining,&bs);
if (imp3 == -1) {
/* fatel error: mp3buffer too small */
desalloc_buffer(&bs); /* Deallocate all buffers */
return -1;
}
mp3count += imp3;
desalloc_buffer(&bs); /* Deallocate all buffers */
return mp3count;
}
/*****************************************************************/
/* write VBR Xing header, and ID3 tag, if asked for */
/*****************************************************************/
void lame_mp3_tags(lame_global_flags *gfp)
{
if (gfp->bWriteVbrTag)
{
/* Calculate relative quality of VBR stream
* 0=best, 100=worst */
int nQuality=gfp->VBR_q*100/9;
/* Write Xing header again */
PutVbrTag(gfp->outPath,nQuality,1-gfp->version);
}
/* write an ID3 tag */
if(id3tag.used) {
id3_buildtag(&id3tag);
id3_writetag(gfp->outPath, &id3tag);
}
}
void lame_version(lame_global_flags *gfp,char *ostring) {
strncpy(ostring,get_lame_version(),20);
}