float *channel_residues;
float *channel_floors;
float *saved;
- float *ret;
- float *buf;
- float *buf_tmp;
uint_fast32_t add_bias; // for float->int conversion
uint_fast32_t exp_bias;
} vorbis_context;
av_freep(&vc->channel_residues);
av_freep(&vc->channel_floors);
av_freep(&vc->saved);
- av_freep(&vc->ret);
- av_freep(&vc->buf);
- av_freep(&vc->buf_tmp);
av_freep(&vc->residues);
av_freep(&vc->modes);
vc->channel_residues= av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->channel_floors = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->saved = av_mallocz((vc->blocksize[1]/4)*vc->audio_channels * sizeof(float));
- vc->ret = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
- vc->buf = av_malloc( vc->blocksize[1]/2 * sizeof(float));
- vc->buf_tmp = av_malloc( vc->blocksize[1]/2 * sizeof(float));
vc->previous_window=0;
ff_mdct_init(&vc->mdct[0], bl0, 1);
vc->avccontext = avccontext;
dsputil_init(&vc->dsp, avccontext);
- if(vc->dsp.float_to_int16 == ff_float_to_int16_c) {
+ if(vc->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
vc->add_bias = 385;
vc->exp_bias = 0;
} else {
avccontext->channels = vc->audio_channels;
avccontext->sample_rate = vc->audio_samplerate;
avccontext->frame_size = FFMIN(vc->blocksize[0], vc->blocksize[1])>>2;
+ avccontext->sample_fmt = SAMPLE_FMT_S16;
return 0 ;
}
// Read and decode residue
-static int vorbis_residue_decode(vorbis_context *vc, vorbis_residue *vr, uint_fast8_t ch, uint_fast8_t *do_not_decode, float *vec, uint_fast16_t vlen) {
+static av_always_inline int vorbis_residue_decode_internal(vorbis_context *vc, vorbis_residue *vr, uint_fast8_t ch, uint_fast8_t *do_not_decode, float *vec, uint_fast16_t vlen, int vr_type) {
GetBitContext *gb=&vc->gb;
uint_fast8_t c_p_c=vc->codebooks[vr->classbook].dimensions;
uint_fast16_t n_to_read=vr->end-vr->begin;
uint_fast8_t i,j,l;
uint_fast16_t k;
- if (vr->type==2) {
+ if (vr_type==2) {
for(j=1;j<ch;++j) {
do_not_decode[0]&=do_not_decode[j]; // FIXME - clobbering input
}
: FASTDIV(vr->partition_size, dim);
vorbis_codebook codebook= vc->codebooks[vqbook];
- if (vr->type==0) {
+ if (vr_type==0) {
voffs=voffset+j*vlen;
for(k=0;k<step;++k) {
}
}
}
- else if (vr->type==1) {
+ else if (vr_type==1) {
voffs=voffset+j*vlen;
for(k=0;k<step;++k) {
coffs=get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * dim;
}
}
}
- else if (vr->type==2 && ch==2 && (voffset&1)==0 && (dim&1)==0) { // most frequent case optimized
+ else if (vr_type==2 && ch==2 && (voffset&1)==0 && (dim&1)==0) { // most frequent case optimized
voffs=voffset>>1;
if(dim==2) {
}
}
- else if (vr->type==2) {
+ else if (vr_type==2) {
voffs=voffset;
for(k=0;k<step;++k) {
AV_DEBUG(" pass %d offs: %d curr: %f change: %f cv offs.: %d+%d \n", pass, voffset/ch+(voffs%ch)*vlen, vec[voffset/ch+(voffs%ch)*vlen], codebook.codevectors[coffs+l], coffs, l);
}
}
- } else {
- av_log(vc->avccontext, AV_LOG_ERROR, " Invalid residue type while residue decode?! \n");
- return 1;
}
}
}
return 0;
}
+static inline int vorbis_residue_decode(vorbis_context *vc, vorbis_residue *vr, uint_fast8_t ch, uint_fast8_t *do_not_decode, float *vec, uint_fast16_t vlen)
+{
+ if (vr->type==2)
+ return vorbis_residue_decode_internal(vc, vr, ch, do_not_decode, vec, vlen, 2);
+ else if (vr->type==1)
+ return vorbis_residue_decode_internal(vc, vr, ch, do_not_decode, vec, vlen, 1);
+ else if (vr->type==0)
+ return vorbis_residue_decode_internal(vc, vr, ch, do_not_decode, vec, vlen, 0);
+ else {
+ av_log(vc->avccontext, AV_LOG_ERROR, " Invalid residue type while residue decode?! \n");
+ return 1;
+ }
+}
+
void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
{
int i;
uint_fast8_t mode_number;
uint_fast8_t blockflag;
uint_fast16_t blocksize;
- int_fast32_t i,j;
+ int_fast32_t i,j,dir;
uint_fast8_t no_residue[vc->audio_channels];
uint_fast8_t do_not_decode[vc->audio_channels];
vorbis_mapping *mapping;
// MDCT, overlap/add, save data for next overlapping FPMATH
retlen = (blocksize + vc->blocksize[previous_window])/4;
- for(j=0;j<vc->audio_channels;++j) {
+ dir = retlen <= blocksize/2; // pick an order so that ret[] can reuse floors[] without stepping on any data we need
+ for(j=dir?0:vc->audio_channels-1; (unsigned)j<vc->audio_channels; j+=dir*2-1) {
uint_fast16_t bs0=vc->blocksize[0];
uint_fast16_t bs1=vc->blocksize[1];
+ float *residue=vc->channel_residues+res_chan[j]*blocksize/2;
+ float *floor=vc->channel_floors+j*blocksize/2;
float *saved=vc->saved+j*bs1/4;
- float *ret=vc->ret+j*retlen;
- float *buf=vc->buf;
+ float *ret=vc->channel_floors+j*retlen;
+ float *buf=residue;
const float *win=vc->win[blockflag&previous_window];
- vc->mdct[0].fft.imdct_half(&vc->mdct[blockflag], buf, vc->channel_floors+j*blocksize/2, vc->buf_tmp);
+ ff_imdct_half(&vc->mdct[blockflag], buf, floor);
if(blockflag == previous_window) {
vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize/4);
{
vorbis_context *vc = avccontext->priv_data ;
GetBitContext *gb = &(vc->gb);
+ const float *channel_ptrs[vc->audio_channels];
+ int i;
int_fast16_t len;
AV_DEBUG("parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb)/8, get_bits_count(gb)%8, len);
- vc->dsp.float_to_int16_interleave(data, vc->ret, len, vc->audio_channels);
+ for(i=0; i<vc->audio_channels; i++)
+ channel_ptrs[i] = vc->channel_floors+i*len;
+ vc->dsp.float_to_int16_interleave(data, channel_ptrs, len, vc->audio_channels);
*data_size=len*2*vc->audio_channels;
return buf_size ;