* Windows Media Audio Voice decoder.
* Copyright (c) 2009 Ronald S. Bultje
*
- * 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 "acelp_filters.h"
#include "lsp.h"
#include "libavutil/lzo.h"
-#include "avfft.h"
-#include "fft.h"
+#include "dct.h"
+#include "rdft.h"
+#include "sinewin.h"
#define MAX_BLOCKS 8 ///< maximum number of blocks per frame
#define MAX_LSPS 16 ///< maximum filter order
/**
* @}
* @defgroup post_filter Postfilter values
- * Varibales used for postfilter implementation, mostly history for
+ * Variables used for postfilter implementation, mostly history for
* smoothing and so on, and context variables for FFT/iFFT.
* @{
*/
2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
- ctx->sample_fmt = SAMPLE_FMT_FLT;
+ ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
return 0;
}
int n, idx;
/* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
- ff_rdft_calc(&s->rdft, lpcs);
+ s->rdft.rdft_calc(&s->rdft, lpcs);
#define log_range(var, assign) do { \
float tmp = log10f(assign); var = tmp; \
max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
(5.0 / 14.7));
angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
for (n = 0; n <= 64; n++) {
- float pow;
+ float pwr;
idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
- pow = wmavoice_denoise_power_table[s->denoise_strength][idx];
- lpcs[n] = angle_mul * pow;
+ pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
+ lpcs[n] = angle_mul * pwr;
/* 70.57 =~ 1/log10(1.0331663) */
- idx = (pow * gain_mul - 0.0295) * 70.570526123;
+ idx = (pwr * gain_mul - 0.0295) * 70.570526123;
if (idx > 127) { // fallback if index falls outside table range
coeffs[n] = wmavoice_energy_table[127] *
powf(1.0331663, idx - 127);
* is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
* Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
* "moment" of the LPCs in this filter. */
- ff_dct_calc(&s->dct, lpcs);
- ff_dct_calc(&s->dst, lpcs);
+ s->dct.dct_calc(&s->dct, lpcs);
+ s->dst.dct_calc(&s->dst, lpcs);
/* Split out the coefficient indexes into phase/magnitude pairs */
idx = 255 + av_clip(lpcs[64], -255, 255);
coeffs[1] = last_coeff;
/* move into real domain */
- ff_rdft_calc(&s->irdft, coeffs);
+ s->irdft.rdft_calc(&s->irdft, coeffs);
/* tilt correction and normalize scale */
memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
/* apply coefficients (in frequency spectrum domain), i.e. complex
* number multiplication */
memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
- ff_rdft_calc(&s->rdft, synth_pf);
- ff_rdft_calc(&s->rdft, coeffs);
+ s->rdft.rdft_calc(&s->rdft, synth_pf);
+ s->rdft.rdft_calc(&s->rdft, coeffs);
synth_pf[0] *= coeffs[0];
synth_pf[1] *= coeffs[1];
for (n = 1; n < 64; n++) {
synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
}
- ff_rdft_calc(&s->irdft, synth_pf);
+ s->irdft.rdft_calc(&s->irdft, synth_pf);
}
/* merge filter output with the history of previous runs */
static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
int block_idx, AMRFixed *fcb)
{
- uint16_t use_mask[7]; // only 5 are used, rest is padding
+ uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
+ uint16_t *use_mask = use_mask_mem + 2;
/* in this function, idx is the index in the 80-bit (+ padding) use_mask
* bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
* of idx are the position of the bit within a particular item in the
/* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
* in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
* we exclude that range from being pulsed again in this function. */
+ memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
memset( use_mask, -1, 5 * sizeof(use_mask[0]));
memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
if (s->aw_n_pulses[block_idx] > 0)
if (*data_size < 480 * sizeof(float)) {
av_log(ctx, AV_LOG_ERROR,
- "Output buffer too small (%d given - %lu needed)\n",
+ "Output buffer too small (%d given - %zu needed)\n",
*data_size, 480 * sizeof(float));
return -1;
}
*data_size = 0;
/* Packets are sometimes a multiple of ctx->block_align, with a packet
- * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
+ * header at each ctx->block_align bytes. However, Libav'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. */
}
}
-AVCodec wmavoice_decoder = {
+AVCodec ff_wmavoice_decoder = {
"wmavoice",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_WMAVOICE,
projects / ffmpeg / blobdiff