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FFmpeg/libavcodec/libopusdec.c
wm4 b945fed629 avcodec: add metadata to identify wrappers and hardware decoders
Explicitly identify decoder/encoder wrappers with a common name. This
saves API users from guessing by the name suffix. For example, they
don't have to guess that "h264_qsv" is the h264 QSV implementation, and
instead they can just check the AVCodec .codec and .wrapper_name fields.

Explicitly mark AVCodec entries that are hardware decoders or most
likely hardware decoders with new AV_CODEC_CAPs. The purpose is allowing
API users listing hardware decoders in a more generic way. The proposed
AVCodecHWConfig does not provide this information fully, because it's
concerned with decoder configuration, not information about the fact
whether the hardware is used or not.

AV_CODEC_CAP_HYBRID exists specifically for QSV, which can have software
implementations in case the hardware is not capable.

Based on a patch by Philip Langdale <philipl@overt.org>.

Merges Libav commit 47687a2f8a.
2017-12-14 19:37:56 +01:00

228 lines
7.9 KiB
C

/*
* 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 "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 },
.wrapper_name = "libopus",
};