* Copyright (c) 2012 Andrew D'Addesio
* Copyright (c) 2013-2014 Mozilla Corporation
*
- * 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
*/
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
-#include "libavresample/avresample.h"
+#include "libswresample/swresample.h"
#include "avcodec.h"
#include "celp_filters.h"
{
int celt_size = av_audio_fifo_size(s->celt_delay);
int ret, i;
-
- ret = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size, nb_samples,
- NULL, 0, 0);
+ ret = swr_convert(s->swr,
+ (uint8_t**)s->out, nb_samples,
+ NULL, 0);
if (ret < 0)
return ret;
else if (ret != nb_samples) {
static int opus_init_resample(OpusStreamContext *s)
{
- float delay[16] = { 0.0 };
- uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
+ static const float delay[16] = { 0.0 };
+ const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
int ret;
- av_opt_set_int(s->avr, "in_sample_rate", s->silk_samplerate, 0);
- ret = avresample_open(s->avr);
+ av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
+ ret = swr_init(s->swr);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
return ret;
}
- ret = avresample_convert(s->avr, NULL, 0, 0, delayptr, sizeof(delay),
- silk_resample_delay[s->packet.bandwidth]);
+ ret = swr_convert(s->swr,
+ NULL, 0,
+ delayptr, silk_resample_delay[s->packet.bandwidth]);
if (ret < 0) {
av_log(s->avctx, AV_LOG_ERROR,
"Error feeding initial silence to the resampler.\n");
/* decode the silk frame */
if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
- if (!avresample_is_open(s->avr)) {
+ if (!swr_is_initialized(s->swr)) {
ret = opus_init_resample(s);
if (ret < 0)
return ret;
av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
return samples;
}
-
- samples = avresample_convert(s->avr, (uint8_t**)s->out, s->out_size,
- s->packet.frame_duration,
- (uint8_t**)s->silk_output,
- sizeof(s->silk_buf[0]),
- samples);
+ samples = swr_convert(s->swr,
+ (uint8_t**)s->out, s->packet.frame_duration,
+ (const uint8_t**)s->silk_output, samples);
if (samples < 0) {
av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
return samples;
}
+ av_assert2((samples & 7) == 0);
s->delayed_samples += s->packet.frame_duration - samples;
} else
ff_silk_flush(s->silk);
int i, j, ret;
/* check if we need to flush the resampler */
- if (avresample_is_open(s->avr)) {
+ if (swr_is_initialized(s->swr)) {
if (buf) {
int64_t cur_samplerate;
- av_opt_get_int(s->avr, "in_sample_rate", 0, &cur_samplerate);
+ av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
} else {
flush_needed = !!s->delayed_samples;
av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
return ret;
}
- avresample_close(s->avr);
+ swr_close(s->swr);
output_samples += s->delayed_samples;
s->delayed_samples = 0;
if (s->celt_delay)
av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
- avresample_close(s->avr);
+ swr_close(s->swr);
ff_silk_flush(s->silk);
ff_celt_flush(s->celt);
s->out_dummy_allocated_size = 0;
av_audio_fifo_free(s->celt_delay);
- avresample_free(&s->avr);
+ swr_free(&s->swr);
}
av_freep(&c->streams);
s->fdsp = &c->fdsp;
- s->avr = avresample_alloc_context();
- if (!s->avr)
+ s->swr =swr_alloc();
+ if (!s->swr)
goto fail;
layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
- av_opt_set_int(s->avr, "in_sample_fmt", avctx->sample_fmt, 0);
- av_opt_set_int(s->avr, "out_sample_fmt", avctx->sample_fmt, 0);
- av_opt_set_int(s->avr, "in_channel_layout", layout, 0);
- av_opt_set_int(s->avr, "out_channel_layout", layout, 0);
- av_opt_set_int(s->avr, "out_sample_rate", avctx->sample_rate, 0);
+ av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
+ av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
+ av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
+ av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
+ av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
+ av_opt_set_int(s->swr, "filter_size", 16, 0);
ret = ff_silk_init(avctx, &s->silk, s->output_channels);
if (ret < 0)
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