* Real Audio 1.0 (14.4K) encoder
* Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
*
- * 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 "avcodec.h"
#include "put_bits.h"
-#include "lpc.h"
#include "celp_filters.h"
#include "ra144.h"
static av_cold int ra144_encode_init(AVCodecContext * avctx)
{
RA144Context *ractx;
+ int ret;
- if (avctx->sample_fmt != SAMPLE_FMT_S16) {
+ if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
return -1;
}
ractx->lpc_coef[0] = ractx->lpc_tables[0];
ractx->lpc_coef[1] = ractx->lpc_tables[1];
ractx->avctx = avctx;
- dsputil_init(&ractx->dsp, avctx);
+ ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
+ FF_LPC_TYPE_LEVINSON);
+ return ret;
+}
+
+
+static av_cold int ra144_encode_close(AVCodecContext *avctx)
+{
+ RA144Context *ractx = avctx->priv_data;
+ ff_lpc_end(&ractx->lpc_ctx);
return 0;
}
/**
- * Quantizes a value by searching a sorted table for the element with the
+ * Quantize a value by searching a sorted table for the element with the
* nearest value
*
* @param value value to quantize
/**
- * Orthogonalizes a vector to another vector
+ * Orthogonalize a vector to another vector
*
* @param v vector to orthogonalize
* @param u vector against which orthogonalization is performed
/**
- * Calculates match score and gain of an LPC-filtered vector with respect to
+ * Calculate match score and gain of an LPC-filtered vector with respect to
* input data, possibly othogonalizing it to up to 2 other vectors
*
* @param work array used to calculate the filtered vector
/**
- * Creates a vector from the adaptive codebook at a given lag value
+ * Create a vector from the adaptive codebook at a given lag value
*
* @param vect array where vector is stored
* @param cb adaptive codebook
/**
- * Searches the adaptive codebook for the best entry and gain and removes its
+ * Search the adaptive codebook for the best entry and gain and remove its
* contribution from input data
*
* @param adapt_cb array from which the adaptive codebook is extracted
ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
for (i = 0; i < BLOCKSIZE; i++)
data[i] -= best_gain * work[i];
- return (best_vect - BLOCKSIZE / 2 + 1);
+ return best_vect - BLOCKSIZE / 2 + 1;
}
/**
- * Finds the best vector of a fixed codebook by applying an LPC filter to
+ * Find the best vector of a fixed codebook by applying an LPC filter to
* codebook entries, possibly othogonalizing them to up to 2 other vectors and
* matching the results with input data
*
/**
- * Searches the two fixed codebooks for the best entry and gain
+ * Search the two fixed codebooks for the best entry and gain
*
* @param work array used to calculate LPC-filtered vectors
* @param coefs coefficients of the LPC filter
/**
- * Encodes a subblock of the current frame
+ * Encode a subblock of the current frame
*
* @param ractx encoder context
* @param sblock_data input data of the subblock
energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
32)];
- ff_lpc_calc_coefs(&ractx->dsp, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
- LPC_ORDER, 16, lpc_coefs, shift, 1, ORDER_METHOD_EST, 12,
- 0);
+ ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
+ LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
+ 0, ORDER_METHOD_EST, 12, 0);
for (i = 0; i < LPC_ORDER; i++)
block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
(12 - shift[LPC_ORDER - 1]));
* The filter is unstable: use the coefficients of the previous frame.
*/
ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
- ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx);
+ if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
+ /* the filter is still unstable. set reflection coeffs to zero. */
+ memset(lpc_refl, 0, sizeof(lpc_refl));
+ }
}
init_put_bits(&pb, frame, buf_size);
for (i = 0; i < LPC_ORDER; i++) {
}
-AVCodec ra_144_encoder =
-{
- "real_144",
- CODEC_TYPE_AUDIO,
- CODEC_ID_RA_144,
- sizeof(RA144Context),
- ra144_encode_init,
- ra144_encode_frame,
- .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"),
+AVCodec ff_ra_144_encoder = {
+ .name = "real_144",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_RA_144,
+ .priv_data_size = sizeof(RA144Context),
+ .init = ra144_encode_init,
+ .encode = ra144_encode_frame,
+ .close = ra144_encode_close,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"),
};
projects / ffmpeg / blobdiff