* TwinVQ decoder
* Copyright (c) 2009 Vitor Sessak
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "dsputil.h"
#include "fft.h"
#include "lsp.h"
+#include "sinewin.h"
#include <math.h>
#include <stdint.h>
* be a multiple of four.
* @return the LPC value
*
- * @todo reuse code from vorbis_dec.c: vorbis_floor0_decode
+ * @todo reuse code from Vorbis decoder: vorbis_floor0_decode
*/
static float eval_lpc_spectrum(const float *lsp, float cos_val, int order)
{
* a*b == 200 and the nearest integer is ill-defined, use a table to emulate
* the following broken float-based implementation used by the binary decoder:
*
- * \code
+ * @code
* static int very_broken_op(int a, int b)
* {
* static float test; // Ugh, force gcc to do the division first...
* test = a/400.;
* return b * test + 0.5;
* }
- * \endcode
+ * @endcode
*
* @note if this function is replaced by just ROUNDED_DIV(a*b,400.), the stddev
* between the original file (before encoding with Yamaha encoder) and the
static void imdct_and_window(TwinContext *tctx, enum FrameType ftype, int wtype,
float *in, float *prev, int ch)
{
+ FFTContext *mdct = &tctx->mdct_ctx[ftype];
const ModeTab *mtab = tctx->mtab;
int bsize = mtab->size / mtab->fmode[ftype].sub;
int size = mtab->size;
wsize = types_sizes[wtype_to_wsize[sub_wtype]];
- ff_imdct_half(&tctx->mdct_ctx[ftype], buf1 + bsize*j, in + bsize*j);
+ mdct->imdct_half(mdct, buf1 + bsize*j, in + bsize*j);
tctx->dsp.vector_fmul_window(out2,
prev_buf + (bsize-wsize)/2,
buf1 + bsize*j,
ff_sine_windows[av_log2(wsize)],
- 0.0,
wsize/2);
out2 += wsize;
dec_bark_env(tctx, bark1[i][j], bark_use_hist[i][j], i,
tctx->tmp_buf, gain[sub*i+j], ftype);
- tctx->dsp.vector_fmul(chunk + block_size*j, tctx->tmp_buf,
+ tctx->dsp.vector_fmul(chunk + block_size*j, chunk + block_size*j, tctx->tmp_buf,
block_size);
}
dec_lpc_spectrum_inv(tctx, lsp, ftype, tctx->tmp_buf);
for (j = 0; j < mtab->fmode[ftype].sub; j++) {
- tctx->dsp.vector_fmul(chunk, tctx->tmp_buf, block_size);
+ tctx->dsp.vector_fmul(chunk, chunk, tctx->tmp_buf, block_size);
chunk += block_size;
}
}
/**
* Interpret the input data as in the following table:
*
- * \verbatim
+ * @verbatim
*
* abcdefgh
* ijklmnop
* qrstuvw
* x123456
*
- * \endverbatim
+ * @endverbatim
*
* and transpose it, giving the output
* aiqxbjr1cks2dlt3emu4fvn5gow6hp
int ibps = avctx->bit_rate/(1000 * avctx->channels);
tctx->avctx = avctx;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
if (avctx->channels > CHANNELS_MAX) {
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %i\n",
return 0;
}
-AVCodec twinvq_decoder =
+AVCodec ff_twinvq_decoder =
{
"twinvq",
AVMEDIA_TYPE_AUDIO,