#include <math.h>
#define BITSTREAM_READER_LE
+#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "fft.h"
#include "fmtconvert.h"
+#include "internal.h"
#include "vorbis.h"
#include "xiph.h"
AVFrame frame;
GetBitContext gb;
DSPContext dsp;
+ AVFloatDSPContext fdsp;
FmtConvertContext fmt_conv;
FFTContext mdct[2];
uint8_t mode_number; // mode number for the current packet
uint8_t previous_window;
float *channel_residues;
- float *channel_floors;
float *saved;
- float scale_bias; // for float->int conversion
} vorbis_context;
/* Helper functions */
int i;
av_freep(&vc->channel_residues);
- av_freep(&vc->channel_floors);
av_freep(&vc->saved);
for (i = 0; i < vc->residue_count; i++)
}
vc->channel_residues = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_residues));
- vc->channel_floors = av_malloc((vc->blocksize[1] / 2) * vc->audio_channels * sizeof(*vc->channel_floors));
vc->saved = av_mallocz((vc->blocksize[1] / 4) * vc->audio_channels * sizeof(*vc->saved));
vc->previous_window = 0;
- ff_mdct_init(&vc->mdct[0], bl0, 1, -vc->scale_bias);
- ff_mdct_init(&vc->mdct[1], bl1, 1, -vc->scale_bias);
+ ff_mdct_init(&vc->mdct[0], bl0, 1, -1.0);
+ ff_mdct_init(&vc->mdct[1], bl1, 1, -1.0);
av_dlog(NULL, " vorbis version %d \n audio_channels %d \n audio_samplerate %d \n bitrate_max %d \n bitrate_nom %d \n bitrate_min %d \n blk_0 %d blk_1 %d \n ",
vc->version, vc->audio_channels, vc->audio_samplerate, vc->bitrate_maximum, vc->bitrate_nominal, vc->bitrate_minimum, vc->blocksize[0], vc->blocksize[1]);
vc->avccontext = avccontext;
ff_dsputil_init(&vc->dsp, avccontext);
+ avpriv_float_dsp_init(&vc->fdsp, avccontext->flags & CODEC_FLAG_BITEXACT);
ff_fmt_convert_init(&vc->fmt_conv, avccontext);
- if (avccontext->request_sample_fmt == AV_SAMPLE_FMT_FLT) {
- avccontext->sample_fmt = AV_SAMPLE_FMT_FLT;
- vc->scale_bias = 1.0f;
- } else {
- avccontext->sample_fmt = AV_SAMPLE_FMT_S16;
- vc->scale_bias = 32768.0f;
- }
+ avccontext->sample_fmt = AV_SAMPLE_FMT_FLTP;
if (!headers_len) {
av_log(avccontext, AV_LOG_ERROR, "Extradata missing.\n");
if (highroom < lowroom) {
room = highroom * 2;
} else {
- room = lowroom * 2; // SPEC mispelling
+ room = lowroom * 2; // SPEC misspelling
}
if (val) {
floor1_flag[low_neigh_offs] = 1;
if (vqbook >= 0 && vc->codebooks[vqbook].codevectors) {
unsigned coffs;
unsigned dim = vc->codebooks[vqbook].dimensions;
- unsigned step = dim == 1 ? vr->partition_size
- : FASTDIV(vr->partition_size, dim);
+ unsigned step = FASTDIV(vr->partition_size << 1, dim << 1);
vorbis_codebook codebook = vc->codebooks[vqbook];
if (vr_type == 0) {
for (k = 0; k < step; ++k) {
coffs = get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * dim;
for (l = 0; l < dim; ++l)
- vec[voffs + k + l * step] += codebook.codevectors[coffs + l]; // FPMATH
+ vec[voffs + k + l * step] += codebook.codevectors[coffs + l];
}
} 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;
for (l = 0; l < dim; ++l, ++voffs) {
- vec[voffs]+=codebook.codevectors[coffs+l]; // FPMATH
+ vec[voffs]+=codebook.codevectors[coffs+l];
av_dlog(NULL, " pass %d offs: %d curr: %f change: %f cv offs.: %d \n",
pass, voffs, vec[voffs], codebook.codevectors[coffs+l], coffs);
if (dim == 2) {
for (k = 0; k < step; ++k) {
coffs = get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * 2;
- vec[voffs + k ] += codebook.codevectors[coffs ]; // FPMATH
- vec[voffs + k + vlen] += codebook.codevectors[coffs + 1]; // FPMATH
+ vec[voffs + k ] += codebook.codevectors[coffs ];
+ vec[voffs + k + vlen] += codebook.codevectors[coffs + 1];
}
} 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
+ vec[voffs ] += codebook.codevectors[coffs ];
+ vec[voffs + 1 ] += codebook.codevectors[coffs + 2];
+ vec[voffs + vlen ] += codebook.codevectors[coffs + 1];
+ vec[voffs + vlen + 1] += codebook.codevectors[coffs + 3];
}
} else
for (k = 0; k < step; ++k) {
coffs = get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * dim;
for (l = 0; l < dim; l += 2, voffs++) {
- vec[voffs ] += codebook.codevectors[coffs + l ]; // FPMATH
- vec[voffs + vlen] += codebook.codevectors[coffs + l + 1]; // FPMATH
+ vec[voffs ] += codebook.codevectors[coffs + l ];
+ vec[voffs + vlen] += codebook.codevectors[coffs + l + 1];
av_dlog(NULL, " pass %d offs: %d curr: %f change: %f cv offs.: %d+%d \n",
pass, voffset / ch + (voffs % ch) * vlen,
}
} else if (vr_type == 2) {
- voffs = voffset;
+ unsigned voffs_div = FASTDIV(voffset << 1, ch <<1);
+ unsigned voffs_mod = voffset - voffs_div * ch;
for (k = 0; k < step; ++k) {
coffs = get_vlc2(gb, codebook.vlc.table, codebook.nb_bits, 3) * dim;
- for (l = 0; l < dim; ++l, ++voffs) {
- vec[voffs / ch + (voffs % ch) * vlen] += codebook.codevectors[coffs + l]; // FPMATH FIXME use if and counter instead of / and %
+ for (l = 0; l < dim; ++l) {
+ vec[voffs_div + voffs_mod * vlen] +=
+ codebook.codevectors[coffs + l];
av_dlog(NULL, " 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],
+ pass, voffs_div + voffs_mod * vlen,
+ vec[voffs_div + voffs_mod * vlen],
codebook.codevectors[coffs + l], coffs, l);
+
+ if (++voffs_mod == ch) {
+ voffs_div++;
+ voffs_mod = 0;
+ }
}
}
}
// Decode the audio packet using the functions above
-static int vorbis_parse_audio_packet(vorbis_context *vc)
+static int vorbis_parse_audio_packet(vorbis_context *vc, float **floor_ptr)
{
GetBitContext *gb = &vc->gb;
FFTContext *mdct;
uint8_t do_not_decode[255];
vorbis_mapping *mapping;
float *ch_res_ptr = vc->channel_residues;
- float *ch_floor_ptr = vc->channel_floors;
uint8_t res_chan[255];
unsigned res_num = 0;
int retlen = 0;
}
memset(ch_res_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
- memset(ch_floor_ptr, 0, sizeof(float) * vc->audio_channels * vlen); //FIXME can this be removed ?
+ for (i = 0; i < vc->audio_channels; ++i)
+ memset(floor_ptr[i], 0, vlen * sizeof(floor_ptr[0][0])); //FIXME can this be removed ?
// Decode floor
floor = &vc->floors[mapping->submap_floor[0]];
}
- ret = floor->decode(vc, &floor->data, ch_floor_ptr);
+ ret = floor->decode(vc, &floor->data, floor_ptr[i]);
if (ret < 0) {
av_log(vc->avccontext, AV_LOG_ERROR, "Invalid codebook in vorbis_floor_decode.\n");
return AVERROR_INVALIDDATA;
}
no_residue[i] = ret;
- ch_floor_ptr += vlen;
}
// Nonzero vector propagate
mdct = &vc->mdct[blockflag];
for (j = vc->audio_channels-1;j >= 0; j--) {
- ch_floor_ptr = vc->channel_floors + j * blocksize / 2;
ch_res_ptr = vc->channel_residues + res_chan[j] * blocksize / 2;
- vc->dsp.vector_fmul(ch_floor_ptr, ch_floor_ptr, ch_res_ptr, blocksize / 2);
- mdct->imdct_half(mdct, ch_res_ptr, ch_floor_ptr);
+ vc->fdsp.vector_fmul(floor_ptr[j], floor_ptr[j], ch_res_ptr, blocksize / 2);
+ mdct->imdct_half(mdct, ch_res_ptr, floor_ptr[j]);
}
-// Overlap/add, save data for next overlapping FPMATH
+// Overlap/add, save data for next overlapping
retlen = (blocksize + vc->blocksize[previous_window]) / 4;
for (j = 0; j < vc->audio_channels; j++) {
unsigned bs1 = vc->blocksize[1];
float *residue = vc->channel_residues + res_chan[j] * blocksize / 2;
float *saved = vc->saved + j * bs1 / 4;
- float *ret = vc->channel_floors + j * retlen;
+ float *ret = floor_ptr[j];
float *buf = residue;
const float *win = vc->win[blockflag & previous_window];
int buf_size = avpkt->size;
vorbis_context *vc = avccontext->priv_data;
GetBitContext *gb = &vc->gb;
- const float *channel_ptrs[255];
+ float *channel_ptrs[255];
int i, len, ret;
av_dlog(NULL, "packet length %d \n", buf_size);
+ /* get output buffer */
+ vc->frame.nb_samples = vc->blocksize[1] / 2;
+ if ((ret = ff_get_buffer(avccontext, &vc->frame)) < 0) {
+ av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+
+ if (vc->audio_channels > 8) {
+ for (i = 0; i < vc->audio_channels; i++)
+ channel_ptrs[i] = (float *)vc->frame.extended_data[i];
+ } else {
+ for (i = 0; i < vc->audio_channels; i++) {
+ int ch = ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
+ channel_ptrs[ch] = (float *)vc->frame.extended_data[i];
+ }
+ }
+
init_get_bits(gb, buf, buf_size*8);
- if ((len = vorbis_parse_audio_packet(vc)) <= 0)
+ if ((len = vorbis_parse_audio_packet(vc, channel_ptrs)) <= 0)
return len;
if (!vc->first_frame) {
av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n",
get_bits_count(gb) / 8, get_bits_count(gb) % 8, len);
- /* get output buffer */
vc->frame.nb_samples = len;
- if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
- av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
- return ret;
- }
-
- if (vc->audio_channels > 8) {
- for (i = 0; i < vc->audio_channels; i++)
- channel_ptrs[i] = vc->channel_floors + i * len;
- } else {
- for (i = 0; i < vc->audio_channels; i++)
- channel_ptrs[i] = vc->channel_floors +
- len * ff_vorbis_channel_layout_offsets[vc->audio_channels - 1][i];
- }
-
- if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
- vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
- len, vc->audio_channels);
- else
- vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
- channel_ptrs, len,
- vc->audio_channels);
-
*got_frame_ptr = 1;
*(AVFrame *)data = vc->frame;
AVCodec ff_vorbis_decoder = {
.name = "vorbis",
.type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_VORBIS,
+ .id = AV_CODEC_ID_VORBIS,
.priv_data_size = sizeof(vorbis_context),
.init = vorbis_decode_init,
.close = vorbis_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
.channel_layouts = ff_vorbis_channel_layouts,
- .sample_fmts = (const enum AVSampleFormat[]) {
- AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
- },
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff