* Copyright (c) 2008 Vladimir Voroshilov
* 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
*/
}
/**
- * Extracts decoding parameters from the input bitstream.
+ * Extract decoding parameters from the input bitstream.
* @param parms parameters structure
* @param pgb pointer to initialized GetBitContext structure
*/
}
}
-static void lsp2lpc_sipr(const double *lsp, float *Az)
-{
- int lp_half_order = LP_FILTER_ORDER >> 1;
- double buf[(LP_FILTER_ORDER >> 1) + 1];
- double pa[(LP_FILTER_ORDER >> 1) + 1];
- double *qa = buf + 1;
- int i,j;
-
- qa[-1] = 0.0;
-
- ff_lsp2polyf(lsp , pa, lp_half_order );
- ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1);
-
- for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) {
- double paf = pa[i] * (1 + lsp[LP_FILTER_ORDER - 1]);
- double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]);
- Az[i-1] = (paf + qaf) * 0.5;
- Az[j-1] = (paf - qaf) * 0.5;
- }
-
- Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) *
- pa[lp_half_order] * 0.5;
-
- Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1];
-}
-
static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
int num_subfr)
{
for (j = 0; j < LP_FILTER_ORDER; j++)
lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
- lsp2lpc_sipr(lsfint, Az);
+ ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
Az += LP_FILTER_ORDER;
t += t0;
}
}
/**
- * Evaluates the adaptive impulse response.
+ * Evaluate the adaptive impulse response.
*/
static void eval_ir(const float *Az, int pitch_lag, float *freq,
float pitch_sharp_factor)
}
/**
- * Evaluates the convolution of a vector with a sparse vector.
+ * Evaluate the convolution of a vector with a sparse vector.
*/
static void convolute_with_sparse(float *out, const AMRFixed *pulses,
const float *shape, int length)
for (i = 0; i < 4; i++)
ctx->energy_history[i] = -14;
- avctx->sample_fmt = SAMPLE_FMT_FLT;
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
dsputil_init(&ctx->dsp, avctx);
mode_par->subframe_count * sizeof(float);
return mode_par->bits_per_frame >> 3;
-};
+}
-AVCodec sipr_decoder = {
+AVCodec ff_sipr_decoder = {
"sipr",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_SIPR,