static void _preextrapolate_helper(vorbis_dsp_state *v) {
int i;
int order=32;
float *lpc=(float *)alloca(order*sizeof(*lpc));
float *work=(float *)alloca(v->pcm_current*sizeof(*work));
long j;
v->preextrapolate=1;
if(v->pcm_current-v->centerW>order*2) { /* safety */
for(i=0; i<v->vi->channels; i++) {
/* need to run the extrapolation in reverse! */
for(j=0; j<v->pcm_current; j++)
work[j]=v->pcm[i][v->pcm_current-j-1];
/* prime as above */
vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
/* run the predictor filter */
vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
order,
work+v->pcm_current-v->centerW,
v->centerW);
for(j=0; j<v->pcm_current; j++)
v->pcm[i][v->pcm_current-j-1]=work[j];
}
}
}
void CoreAudioPaddedEncoder::extrapolate(float *input, size_t ilen,
float *output, size_t olen)
{
unsigned n = getInputDescription().mChannelsPerFrame;
float lpc[LPC_ORDER];
for (unsigned i = 0; i < n; ++i) {
vorbis_lpc_from_data(input + i, lpc, ilen, LPC_ORDER, n);
vorbis_lpc_predict(lpc, &input[i + n * (ilen - LPC_ORDER)],
LPC_ORDER, output + i, olen, n);
}
}
static int extrapolate(extrapolater_t *self, const buffer_t *bp,
void *dst, unsigned nframes)
{
const pcm_sample_description_t *sfmt = pcm_get_format(self->src);
unsigned i, n = sfmt->channels_per_frame;
float lpc[LPC_ORDER];
for (i = 0; i < n; ++i) {
vorbis_lpc_from_data(bp->data + i, lpc, bp->count, LPC_ORDER, n);
vorbis_lpc_predict(lpc, &bp->data[i + n * (bp->count - LPC_ORDER)],
LPC_ORDER, (sample_t*)dst + i, nframes, n);
}
return nframes;
}
static void _preextrapolate_helper( vorbis_dsp_state *v )
{
int i;
int order = 16;
float *lpc = alloca( order * sizeof( *lpc ) );
float *work = alloca( v->pcm_current * sizeof( *work ) );
long j;
v->preextrapolate = 1;
if ( v->pcm_current - v->centerW > order*2 )
{
/* safety */
for ( i = 0;i < v->vi->channels;i++ )
{
/* need to run the extrapolation in reverse! */
for ( j = 0;j < v->pcm_current;j++ )
work[j] = v->pcm[i][v->pcm_current-j-1];
/* prime as above */
vorbis_lpc_from_data( work, lpc, v->pcm_current - v->centerW, order );
#if 0
if ( v->vi->channels == 2 )
{
if ( i == 0 )
_analysis_output( "predataL", 0, work, v->pcm_current - v->centerW, 0, 0, 0 );
else
_analysis_output( "predataR", 0, work, v->pcm_current - v->centerW, 0, 0, 0 );
}
else
{
_analysis_output( "predata", 0, work, v->pcm_current - v->centerW, 0, 0, 0 );
}
#endif
/* run the predictor filter */
vorbis_lpc_predict( lpc, work + v->pcm_current - v->centerW - order,
order,
work + v->pcm_current - v->centerW,
v->centerW );
for ( j = 0;j < v->pcm_current;j++ )
v->pcm[i][v->pcm_current-j-1] = work[j];
}
}
}
int vorbis_analysis_wrote(vorbis_dsp_state *v, int vals){
vorbis_info *vi=v->vi;
codec_setup_info *ci=vi->codec_setup;
if(vals<=0){
int order=32;
int i;
float *lpc=alloca(order*sizeof(*lpc));
/* if it wasn't done earlier (very short sample) */
if(!v->preextrapolate)
_preextrapolate_helper(v);
/* We're encoding the end of the stream. Just make sure we have
[at least] a few full blocks of zeroes at the end. */
/* actually, we don't want zeroes; that could drop a large
amplitude off a cliff, creating spread spectrum noise that will
suck to encode. Extrapolate for the sake of cleanliness. */
vorbis_analysis_buffer(v,ci->blocksizes[1]*3);
v->eofflag=v->pcm_current;
v->pcm_current+=ci->blocksizes[1]*3;
for(i=0;i<vi->channels;i++){
if(v->eofflag>order*2){
/* extrapolate with LPC to fill in */
long n;
/* make a predictor filter */
n=v->eofflag;
if(n>ci->blocksizes[1])n=ci->blocksizes[1];
vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
/* run the predictor filter */
vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
}else{
/* not enough data to extrapolate (unlikely to happen due to
guarding the overlap, but bulletproof in case that
assumtion goes away). zeroes will do. */
memset(v->pcm[i]+v->eofflag,0,
(v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
}
}
}else{
if(v->pcm_current+vals>v->pcm_storage)
return(OV_EINVAL);
v->pcm_current+=vals;
/* we may want to reverse extrapolate the beginning of a stream
too... in case we're beginning on a cliff! */
/* clumsy, but simple. It only runs once, so simple is good. */
if(!v->preextrapolate && v->pcm_current-v->centerW>ci->blocksizes[1])
_preextrapolate_helper(v);
}
return(0);
}
void dumb_it_add_lpc(struct DUMB_IT_SIGDATA *sigdata){
float lpc[lpc_order * 2];
float lpc_input[lpc_max * 2];
float lpc_output[lpc_extra * 2];
signed char * s8;
signed short * s16;
int n, o, offset, lpc_samples;
for ( n = 0; n < sigdata->n_samples; n++ ) {
IT_SAMPLE * sample = sigdata->sample + n;
if ( ( sample->flags & ( IT_SAMPLE_EXISTS | IT_SAMPLE_LOOP) ) == IT_SAMPLE_EXISTS ) {
/* If we have enough sample data to train the filter, use the filter to generate the padding */
if ( sample->length >= lpc_order ) {
lpc_samples = sample->length;
if (lpc_samples > lpc_max) lpc_samples = lpc_max;
offset = sample->length - lpc_samples;
if ( sample->flags & IT_SAMPLE_STEREO )
{
if ( sample->flags & IT_SAMPLE_16BIT )
{
s16 = ( signed short * ) sample->data;
s16 += offset * 2;
for ( o = 0; o < lpc_samples; o++ )
{
lpc_input[ o ] = s16[ o * 2 + 0 ];
lpc_input[ o + lpc_max ] = s16[ o * 2 + 1 ];
}
}
else
{
s8 = ( signed char * ) sample->data;
s8 += offset * 2;
for ( o = 0; o < lpc_samples; o++ )
{
lpc_input[ o ] = s8[ o * 2 + 0 ];
lpc_input[ o + lpc_max ] = s8[ o * 2 + 1 ];
}
}
vorbis_lpc_from_data( lpc_input, lpc, lpc_samples, lpc_order );
vorbis_lpc_from_data( lpc_input + lpc_max, lpc + lpc_order, lpc_samples, lpc_order );
vorbis_lpc_predict( lpc, lpc_input + lpc_samples - lpc_order, lpc_order, lpc_output, lpc_extra );
vorbis_lpc_predict( lpc + lpc_order, lpc_input + lpc_max + lpc_samples - lpc_order, lpc_order, lpc_output + lpc_extra, lpc_extra );
if ( sample->flags & IT_SAMPLE_16BIT )
{
s16 = ( signed short * ) realloc( sample->data, ( sample->length + lpc_extra ) * 2 * sizeof(short) );
sample->data = s16;
s16 += sample->length * 2;
sample->length += lpc_extra;
for ( o = 0; o < lpc_extra; o++ )
{
s16[ o * 2 + 0 ] = (signed short)lpc_output[ o ];
s16[ o * 2 + 1 ] = (signed short)lpc_output[ o + lpc_extra ];
}
}
else
{
s8 = ( signed char * ) realloc( sample->data, ( sample->length + lpc_extra ) * 2 );
sample->data = s8;
s8 += sample->length * 2;
sample->length += lpc_extra;
for ( o = 0; o < lpc_extra; o++ )
{
s8[ o * 2 + 0 ] = (signed char)lpc_output[ o ];
s8[ o * 2 + 1 ] = (signed char)lpc_output[ o + lpc_extra ];
}
}
}
else
{
if ( sample->flags & IT_SAMPLE_16BIT )
{
s16 = ( signed short * ) sample->data;
s16 += offset;
for ( o = 0; o < lpc_samples; o++ )
{
lpc_input[ o ] = s16[ o ];
}
}
else
{
s8 = ( signed char * ) sample->data;
s8 += offset;
for ( o = 0; o < lpc_samples; o++ )
{
lpc_input[ o ] = s8[ o ];
}
}
vorbis_lpc_from_data( lpc_input, lpc, lpc_samples, lpc_order );
vorbis_lpc_predict( lpc, lpc_input + lpc_samples - lpc_order, lpc_order, lpc_output, lpc_extra );
if ( sample->flags & IT_SAMPLE_16BIT )
{
s16 = ( signed short * ) realloc( sample->data, ( sample->length + lpc_extra ) * sizeof(short) );
sample->data = s16;
s16 += sample->length;
sample->length += lpc_extra;
for ( o = 0; o < lpc_extra; o++ )
{
s16[ o ] = (signed short)lpc_output[ o ];
}
}
else
{
s8 = ( signed char * ) realloc( sample->data, sample->length + lpc_extra );
sample->data = s8;
s8 += sample->length;
sample->length += lpc_extra;
for ( o = 0; o < lpc_extra; o++ )
{
s8[ o ] = (signed char)lpc_output[ o ];
}
}
}
}
else
/* Otherwise, pad with silence. */
{
offset = sample->length;
lpc_samples = lpc_extra;
sample->length += lpc_samples;
n = 1;
if ( sample->flags & IT_SAMPLE_STEREO ) n *= 2;
if ( sample->flags & IT_SAMPLE_16BIT ) n *= 2;
offset *= n;
lpc_samples *= n;
sample->data = realloc( sample->data, offset + lpc_samples );
memset( (char*)sample->data + offset, 0, lpc_samples );
}
}
}
}