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/*
* Opus decoder using libopus
* Copyright (c) 2012 Nicolas George
*
* This file is part of FFmpeg.
*
* 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.
*
* 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 FFmpeg; 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/internal.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;
int pre_skip;
#ifndef OPUS_SET_GAIN
union { int i; double d; } gain;
#endif
};
#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->channels = avc->extradata_size >= 10 ? avc->extradata[9] : (avc->channels == 1) ? 1 : 2;
if (avc->channels <= 0) {
av_log(avc, AV_LOG_WARNING,
"Invalid number of channels %d, defaulting to stereo\n", avc->channels);
avc->channels = 2;
}
avc->sample_rate = 48000;
avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
if (avc->extradata_size >= OPUS_HEAD_SIZE) {
opus->pre_skip = AV_RL16(avc->extradata + 10);
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 (channel_map == 1) {
avc->channel_layout = avc->channels > 8 ? 0 :
ff_vorbis_channel_layouts[avc->channels - 1];
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 ffmpeg order */
for (ch = 0; ch < avc->channels; ch++)
mapping_arr[ch] = mapping[vorbis_offset[ch]];
mapping = mapping_arr;
}
} else if (channel_map == 2) {
int ambisonic_order = ff_sqrt(avc->channels) - 1;
if (avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) &&
avc->channels != (ambisonic_order + 1) * (ambisonic_order + 1) + 2) {
av_log(avc, AV_LOG_ERROR,
"Channel mapping 2 is only specified for channel counts"
" which can be written as (n + 1)^2 or (n + 2)^2 + 2"
" for nonnegative integer n\n");
return AVERROR_INVALIDDATA;
}
if (avc->channels > 227) {
av_log(avc, AV_LOG_ERROR, "Too many channels\n");
return AVERROR_INVALIDDATA;
}
avc->channel_layout = 0;
} else {
avc->channel_layout = 0;
}
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);
}
#ifdef OPUS_SET_GAIN
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));
#else
{
double gain_lin = ff_exp10(gain_db / (20.0 * 256));
if (avc->sample_fmt == AV_SAMPLE_FMT_FLT)
opus->gain.d = gain_lin;
else
opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX);
}
#endif
/* Decoder delay (in samples) at 48kHz */
avc->delay = avc->internal->skip_samples = opus->pre_skip;
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 *data,
int *got_frame_ptr, AVPacket *pkt)
{
struct libopus_context *opus = avc->priv_data;
AVFrame *frame = data;
int ret, nb_samples;
frame->nb_samples = MAX_FRAME_SIZE;
if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
return ret;
if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
(opus_int16 *)frame->data[0],
frame->nb_samples, 0);
else
nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
(float *)frame->data[0],
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);
}
#ifndef OPUS_SET_GAIN
{
int i = avc->channels * nb_samples;
if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
float *pcm = (float *)frame->data[0];
for (; i > 0; i--, pcm++)
*pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
} else {
int16_t *pcm = (int16_t *)frame->data[0];
for (; i > 0; i--, pcm++)
*pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
}
}
#endif
frame->nb_samples = nb_samples;
*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);
/* The stream can have been extracted by a tool that is not Opus-aware.
Therefore, any packet can become the first of the stream. */
avc->internal->skip_samples = opus->pre_skip;
}
AVCodec ff_libopus_decoder = {
.name = "libopus",
.long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
.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 = AV_CODEC_CAP_DR1,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
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