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FFmpeg/libavcodec/libopusdec.c
Anton Khirnov 594d4d5df3 lavc: add a wrapper for AVCodecContext.get_buffer().
It will be useful in the upcoming transition to refcounted AVFrames.
2012-12-04 21:41:59 +01:00

169 lines
5.8 KiB
C

/*
* Opus decoder using libopus
* Copyright (c) 2012 Nicolas George
*
* This file is part of Libav.
*
* 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.
*
* 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <opus.h>
#include <opus_multistream.h>
#include "libavutil/avassert.h"
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "internal.h"
#include "vorbis.h"
#include "mathops.h"
#include "libopus.h"
struct libopus_context {
OpusMSDecoder *dec;
AVFrame frame;
};
#define OPUS_HEAD_SIZE 19
static av_cold int libopus_decode_init(AVCodecContext *avc)
{
struct libopus_context *opus = avc->priv_data;
int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
avc->sample_rate = 48000;
avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
avc->channel_layout = avc->channels > 8 ? 0 :
ff_vorbis_channel_layouts[avc->channels - 1];
if (avc->extradata_size >= OPUS_HEAD_SIZE) {
gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16);
channel_map = AV_RL8 (avc->extradata + 18);
}
if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0];
nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
if (nb_streams + nb_coupled != avc->channels)
av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
} else {
if (avc->channels > 2 || channel_map) {
av_log(avc, AV_LOG_ERROR,
"No channel mapping for %d channels.\n", avc->channels);
return AVERROR(EINVAL);
}
nb_streams = 1;
nb_coupled = avc->channels > 1;
mapping = mapping_arr;
}
if (avc->channels > 2 && avc->channels <= 8) {
const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
int ch;
/* Remap channels from vorbis order to libav order */
for (ch = 0; ch < avc->channels; ch++)
mapping_arr[ch] = mapping[vorbis_offset[ch]];
mapping = mapping_arr;
}
opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
nb_streams, nb_coupled,
mapping, &ret);
if (!opus->dec) {
av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
opus_strerror(ret));
return ff_opus_error_to_averror(ret);
}
ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
if (ret != OPUS_OK)
av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
opus_strerror(ret));
avc->delay = 3840; /* Decoder delay (in samples) at 48kHz */
avcodec_get_frame_defaults(&opus->frame);
avc->coded_frame = &opus->frame;
return 0;
}
static av_cold int libopus_decode_close(AVCodecContext *avc)
{
struct libopus_context *opus = avc->priv_data;
opus_multistream_decoder_destroy(opus->dec);
return 0;
}
#define MAX_FRAME_SIZE (960 * 6)
static int libopus_decode(AVCodecContext *avc, void *frame,
int *got_frame_ptr, AVPacket *pkt)
{
struct libopus_context *opus = avc->priv_data;
int ret, nb_samples;
opus->frame.nb_samples = MAX_FRAME_SIZE;
ret = ff_get_buffer(avc, &opus->frame);
if (ret < 0) {
av_log(avc, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
(opus_int16 *)opus->frame.data[0],
opus->frame.nb_samples, 0);
else
nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
(float *)opus->frame.data[0],
opus->frame.nb_samples, 0);
if (nb_samples < 0) {
av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
opus_strerror(nb_samples));
return ff_opus_error_to_averror(nb_samples);
}
opus->frame.nb_samples = nb_samples;
*(AVFrame *)frame = opus->frame;
*got_frame_ptr = 1;
return pkt->size;
}
static void libopus_flush(AVCodecContext *avc)
{
struct libopus_context *opus = avc->priv_data;
opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE);
}
AVCodec ff_libopus_decoder = {
.name = "libopus",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_OPUS,
.priv_data_size = sizeof(struct libopus_context),
.init = libopus_decode_init,
.close = libopus_decode_close,
.decode = libopus_decode,
.flush = libopus_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};