📄 lame.c
字号:
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, "-")? gfp->inPath : "stdin"),
(strcmp(gfp->outPath, "-")? 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) {
⌨️ 快捷键说明
复制代码
Ctrl + C
搜索代码
Ctrl + F
全屏模式
F11
切换主题
Ctrl + Shift + D
显示快捷键
?
增大字号
Ctrl + =
减小字号
Ctrl + -