uint_fast8_t mode_count;
vorbis_mode *modes;
uint_fast8_t mode_number; // mode number for the current packet
+ uint_fast8_t previous_window;
float *channel_residues;
float *channel_floors;
float *saved;
- uint_fast16_t saved_start;
- float *ret;
- float *buf;
- float *buf_tmp;
uint_fast32_t add_bias; // for float->int conversion
uint_fast32_t exp_bias;
} vorbis_context;
double mant=val&0x1fffff;
long exp=(val&0x7fe00000L)>>21;
if (val&0x80000000) mant=-mant;
- return(ldexp(mant, exp-20-768));
+ return ldexp(mant, exp - 20 - 768);
}
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]/2)*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] * sizeof(float));
- vc->buf_tmp = av_malloc( vc->blocksize[1] * sizeof(float));
- vc->saved_start=0;
+ vc->saved = av_mallocz((vc->blocksize[1]/4)*vc->audio_channels * sizeof(float));
+ vc->previous_window=0;
ff_mdct_init(&vc->mdct[0], bl0, 1);
ff_mdct_init(&vc->mdct[1], bl1, 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 ;
}
vec[voffs+k ]+=codebook.codevectors[coffs ]; // FPMATH
vec[voffs+k+vlen]+=codebook.codevectors[coffs+1]; // FPMATH
}
+ } else if(dim==4) {
+ for(k=0;k<step;++k, voffs+=2) {
+ coffs=get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * 4;
+ vec[voffs ]+=codebook.codevectors[coffs ]; // FPMATH
+ vec[voffs+1 ]+=codebook.codevectors[coffs+2]; // FPMATH
+ vec[voffs+vlen ]+=codebook.codevectors[coffs+1]; // FPMATH
+ vec[voffs+vlen+1]+=codebook.codevectors[coffs+3]; // FPMATH
+ }
} else
for(k=0;k<step;++k) {
coffs=get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * dim;
}
}
+static void copy_normalize(float *dst, float *src, int len, int exp_bias, float add_bias)
+{
+ int i;
+ if(exp_bias) {
+ for(i=0; i<len; i++)
+ ((uint32_t*)dst)[i] = ((uint32_t*)src)[i] + exp_bias; // dst[k]=src[i]*(1<<bias)
+ } else {
+ for(i=0; i<len; i++)
+ dst[i] = src[i] + add_bias;
+ }
+}
+
// Decode the audio packet using the functions above
static int vorbis_parse_audio_packet(vorbis_context *vc) {
GetBitContext *gb=&vc->gb;
- uint_fast8_t previous_window=0,next_window=0;
+ uint_fast8_t previous_window=vc->previous_window;
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;
uint_fast8_t res_chan[vc->audio_channels];
uint_fast8_t res_num=0;
int_fast16_t retlen=0;
- uint_fast16_t saved_start=0;
float fadd_bias = vc->add_bias;
if (get_bits1(gb)) {
AV_DEBUG(" Mode number: %d , mapping: %d , blocktype %d \n", mode_number, vc->modes[mode_number].mapping, vc->modes[mode_number].blockflag);
- if (vc->modes[mode_number].blockflag) {
- previous_window=get_bits1(gb);
- next_window=get_bits1(gb);
+ blockflag=vc->modes[mode_number].blockflag;
+ blocksize=vc->blocksize[blockflag];
+ if (blockflag) {
+ skip_bits(gb, 2); // previous_window, next_window
}
- blocksize=vc->blocksize[vc->modes[mode_number].blockflag];
memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
// MDCT, overlap/add, save data for next overlapping FPMATH
- for(j=0;j<vc->audio_channels;++j) {
- uint_fast8_t step=vc->audio_channels;
- uint_fast16_t k;
- float *saved=vc->saved+j*vc->blocksize[1]/2;
- float *ret=vc->ret;
- const float *lwin=vc->win[1];
- const float *swin=vc->win[0];
- float *buf=vc->buf;
- float *buf_tmp=vc->buf_tmp;
-
- ch_floor_ptr=vc->channel_floors+j*blocksize/2;
-
- saved_start=vc->saved_start;
-
- vc->mdct[0].fft.imdct_calc(&vc->mdct[vc->modes[mode_number].blockflag], buf, ch_floor_ptr, buf_tmp);
-
- //FIXME process channels together, to allow faster simd vector_fmul_add_add?
- if (vc->modes[mode_number].blockflag) {
- // -- overlap/add
- if (previous_window) {
- vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize[1]/2, step);
- retlen=vc->blocksize[1]/2;
- } else {
- int len = (vc->blocksize[1]-vc->blocksize[0])/4;
- buf += len;
- vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize[0]/2, step);
- k = vc->blocksize[0]/2*step + j;
- buf += vc->blocksize[0]/2;
- if(vc->exp_bias){
- for(i=0; i<len; i++, k+=step)
- ((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
- } else {
- for(i=0; i<len; i++, k+=step)
- ret[k] = buf[i] + fadd_bias;
- }
- buf=vc->buf;
- retlen=vc->blocksize[0]/2+len;
- }
- // -- save
- if (next_window) {
- buf += vc->blocksize[1]/2;
- vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize[1]/2);
- saved_start=0;
- } else {
- saved_start=(vc->blocksize[1]-vc->blocksize[0])/4;
- buf += vc->blocksize[1]/2;
- for(i=0; i<saved_start; i++)
- ((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
- vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize[0]/2);
- }
+ retlen = (blocksize + vc->blocksize[previous_window])/4;
+ 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->channel_floors+j*retlen;
+ float *buf=residue;
+ const float *win=vc->win[blockflag&previous_window];
+
+ 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);
+ } else if(blockflag > previous_window) {
+ vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, bs0/4);
+ copy_normalize(ret+bs0/2, buf+bs0/4, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
} else {
- // --overlap/add
- if(vc->add_bias) {
- for(k=j, i=0;i<saved_start;++i, k+=step)
- ret[k] = saved[i] + fadd_bias;
- } else {
- for(k=j, i=0;i<saved_start;++i, k+=step)
- ret[k] = saved[i];
- }
- vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize[0]/2, step);
- retlen=saved_start+vc->blocksize[0]/2;
- // -- save
- buf += vc->blocksize[0]/2;
- vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize[0]/2);
- saved_start=0;
+ copy_normalize(ret, saved, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
+ vc->dsp.vector_fmul_window(ret+(bs1-bs0)/4, saved+(bs1-bs0)/4, buf, win, fadd_bias, bs0/4);
}
+ memcpy(saved, buf+blocksize/4, blocksize/4*sizeof(float));
}
- vc->saved_start=saved_start;
- return retlen*vc->audio_channels;
+ vc->previous_window = blockflag;
+ return retlen;
}
// Return the decoded audio packet through the standard api
{
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(data, vc->ret, len);
- *data_size=len*2;
+ 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 ;
}
NULL,
vorbis_decode_close,
vorbis_decode_frame,
- .long_name = "Vorbis",
+ .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
};
projects / ffmpeg / blobdiff