* Windows Media Audio Voice decoder.
* Copyright (c) 2009 Ronald S. Bultje
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define UNCHECKED_BITSTREAM_READER 1
#include <math.h>
+
+#include "dsputil.h"
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "wmavoice_data.h"
-#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
#include "acelp_filters.h"
float optimal_gain = 0, dot;
const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
*end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
- *best_hist_ptr;
+ *best_hist_ptr = NULL;
/* find best fitting point in history */
do {
- dot = ff_dot_productf(in, ptr, size);
+ dot = ff_scalarproduct_float_c(in, ptr, size);
if (dot > optimal_gain) {
optimal_gain = dot;
best_hist_ptr = ptr;
if (optimal_gain <= 0)
return -1;
- dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size);
+ dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
if (dot <= 0) // would be 1.0
return -1;
{
float rh0, rh1;
- rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs);
- rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1);
+ rh0 = 1.0 + ff_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
+ rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
return rh1 / rh0;
}
-1.8 * tilt_factor(coeffs, remainder - 1),
coeffs, remainder);
}
- sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder));
+ sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
for (n = 0; n < remainder; n++)
coeffs[n] *= sq;
}
*synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
*synth_filter_in = zero_exc_pf;
- assert(size <= MAX_FRAMESIZE / 2);
+ av_assert0(size <= MAX_FRAMESIZE / 2);
/* generate excitation from input signal */
ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
float gain;
int n, r_idx;
- assert(size <= MAX_FRAMESIZE);
+ av_assert0(size <= MAX_FRAMESIZE);
/* Set the offset from which we start reading wmavoice_std_codebook */
if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
int n, idx, gain_weight;
AMRFixed fcb;
- assert(size <= MAX_FRAMESIZE / 2);
+ av_assert0(size <= MAX_FRAMESIZE / 2);
memset(pulses, 0, sizeof(*pulses) * size);
fcb.pitch_lag = block_pitch_sh2 >> 2;
/* Calculate gain for adaptive & fixed codebook signal.
* see ff_amr_set_fixed_gain(). */
idx = get_bits(gb, 7);
- fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
+ fcb_gain = expf(ff_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) -
5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
acb_gain = wmavoice_gain_codebook_acb[idx];
pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
/* initialize a copy */
init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
skip_bits_long(gb, get_bits_count(orig_gb));
- assert(get_bits_left(gb) == get_bits_left(orig_gb));
+ av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
/* superframe header */
if (get_bits_left(gb) < 14)
int size, res, pos;
/* Packets are sometimes a multiple of ctx->block_align, with a packet
- * header at each ctx->block_align bytes. However, Libav's ASF demuxer
+ * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
* feeds us ASF packets, which may concatenate multiple "codec" packets
* in a single "muxer" packet, so we artificially emulate that by
* capping the packet size at ctx->block_align. */
/* rewind bit reader to start of last (incomplete) superframe... */
init_get_bits(gb, avpkt->data, size << 3);
skip_bits_long(gb, (size << 3) - pos);
- assert(get_bits_left(gb) == pos);
+ av_assert1(get_bits_left(gb) == pos);
/* ...and cache it for spillover in next packet */
init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);