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ffmpeg/libavcodec/libopusdec.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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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 "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "vorbis.h"
#include "mathops.h"
#include "libopus.h"
struct libopus_context {
AVClass *class;
OpusMSDecoder *dec;
int pre_skip;
#ifndef OPUS_SET_GAIN
union { int i; double d; } gain;
#endif
#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
int apply_phase_inv;
#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;
avc->channel_layout = avc->channels > 8 ? 0 :
ff_vorbis_channel_layouts[avc->channels - 1];
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 (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;
}
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
#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
ret = opus_multistream_decoder_ctl(opus->dec,
OPUS_SET_PHASE_INVERSION_DISABLED(!opus->apply_phase_inv));
if (ret != OPUS_OK)
av_log(avc, AV_LOG_WARNING,
"Unable to set phase inversion: %s\n",
opus_strerror(ret));
#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;
if (opus->dec) {
opus_multistream_decoder_destroy(opus->dec);
opus->dec = NULL;
}
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;
}
#define OFFSET(x) offsetof(struct libopus_context, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption libopusdec_options[] = {
#ifdef OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST
{ "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, FLAGS },
#endif
{ NULL },
};
static const AVClass libopusdec_class = {
.class_name = "libopusdec",
.item_name = av_default_item_name,
.option = libopusdec_options,
.version = LIBAVUTIL_VERSION_INT,
};
const 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 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.priv_class = &libopusdec_class,
.wrapper_name = "libopus",
};
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