* 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 <float.h>
#include "avcodec.h"
-#include "put_bits.h"
-#include "lpc.h"
+#include "audio_frame_queue.h"
#include "celp_filters.h"
+#include "internal.h"
+#include "mathops.h"
+#include "put_bits.h"
#include "ra144.h"
+static av_cold int ra144_encode_close(AVCodecContext *avctx)
+{
+ RA144Context *ractx = avctx->priv_data;
+ ff_lpc_end(&ractx->lpc_ctx);
+ ff_af_queue_close(&ractx->afq);
+ return 0;
+}
+
+
static av_cold int ra144_encode_init(AVCodecContext * avctx)
{
RA144Context *ractx;
+ int ret;
- if (avctx->sample_fmt != SAMPLE_FMT_S16) {
- av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
- return -1;
- }
if (avctx->channels != 1) {
av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
avctx->channels);
return -1;
}
avctx->frame_size = NBLOCKS * BLOCKSIZE;
+ avctx->initial_padding = avctx->frame_size;
avctx->bit_rate = 8000;
ractx = avctx->priv_data;
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);
+ if (ret < 0)
+ goto error;
+
+ ff_af_queue_init(avctx, &ractx->afq);
+
return 0;
+error:
+ ra144_encode_close(avctx);
+ return ret;
}
/**
* Calculate match score and gain of an LPC-filtered vector with respect to
- * input data, possibly othogonalizing it to up to 2 other vectors
+ * input data, possibly orthogonalizing it to up to two other vectors.
*
* @param work array used to calculate the filtered vector
* @param coefs coefficients of the LPC filter
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;
}
/**
* 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
+ * codebook entries, possibly orthogonalizing them to up to two other vectors
+ * and matching the results with input data.
*
* @param work array used to calculate the filtered vectors
* @param coefs coefficients of the LPC filter
const int16_t *lpc_coefs, unsigned int rms,
PutBitContext *pb)
{
- float data[BLOCKSIZE], work[LPC_ORDER + BLOCKSIZE];
+ float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
float coefs[LPC_ORDER];
float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
int16_t cba_vect[BLOCKSIZE];
* Calculate the zero-input response of the LPC filter and subtract it from
* input data.
*/
- memset(data, 0, sizeof(data));
ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
LPC_ORDER);
for (i = 0; i < BLOCKSIZE; i++) {
}
-static int ra144_encode_frame(AVCodecContext *avctx, uint8_t *frame,
- int buf_size, void *data)
+static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
{
static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
- RA144Context *ractx;
+ RA144Context *ractx = avctx->priv_data;
PutBitContext pb;
int32_t lpc_data[NBLOCKS * BLOCKSIZE];
int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
int16_t block_coefs[NBLOCKS][LPC_ORDER];
int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */
unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
+ const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
int energy = 0;
- int i, idx;
+ int i, idx, ret;
- if (buf_size < FRAMESIZE) {
- av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+ if (ractx->last_frame)
return 0;
+
+ if ((ret = ff_alloc_packet(avpkt, FRAMESIZE))) {
+ av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+ return ret;
}
- ractx = avctx->priv_data;
/**
* Since the LPC coefficients are calculated on a frame centered over the
lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
energy += (lpc_data[i] * lpc_data[i]) >> 4;
}
- for (i = 2 * BLOCKSIZE + BLOCKSIZE / 2; i < NBLOCKS * BLOCKSIZE; i++) {
- lpc_data[i] = *((int16_t *)data + i - 2 * BLOCKSIZE - BLOCKSIZE / 2) >>
- 2;
- energy += (lpc_data[i] * lpc_data[i]) >> 4;
+ if (frame) {
+ int j;
+ for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
+ lpc_data[i] = samples[j] >> 2;
+ energy += (lpc_data[i] * lpc_data[i]) >> 4;
+ }
}
+ if (i < NBLOCKS * BLOCKSIZE)
+ memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
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);
+ init_put_bits(&pb, avpkt->data, avpkt->size);
for (i = 0; i < LPC_ORDER; i++) {
idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
put_bits(&pb, bit_sizes[i], idx);
ractx->old_energy = energy;
ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);
- for (i = 0; i < NBLOCKS * BLOCKSIZE; i++)
- ractx->curr_block[i] = *((int16_t *)data + i) >> 2;
- return FRAMESIZE;
+
+ /* copy input samples to current block for processing in next call */
+ i = 0;
+ if (frame) {
+ for (; i < frame->nb_samples; i++)
+ ractx->curr_block[i] = samples[i] >> 2;
+
+ if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
+ return ret;
+ } else
+ ractx->last_frame = 1;
+ memset(&ractx->curr_block[i], 0,
+ (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));
+
+ /* Get the next frame pts/duration */
+ ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
+ &avpkt->duration);
+
+ avpkt->size = FRAMESIZE;
+ *got_packet_ptr = 1;
+ return 0;
}
-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",
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_RA_144,
+ .priv_data_size = sizeof(RA144Context),
+ .init = ra144_encode_init,
+ .encode2 = ra144_encode_frame,
+ .close = ra144_encode_close,
+ .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE },
};
projects / ffmpeg / blobdiff