forked from KolibriOS/kolibrios
199 lines
6.7 KiB
C
199 lines
6.7 KiB
C
/*
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* Opus decoder using libopus
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* Copyright (c) 2012 Nicolas George
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <opus.h>
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#include <opus_multistream.h>
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#include "libavutil/avassert.h"
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "internal.h"
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#include "vorbis.h"
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#include "mathops.h"
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#include "libopus.h"
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struct libopus_context {
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OpusMSDecoder *dec;
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int pre_skip;
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#ifndef OPUS_SET_GAIN
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union { int i; double d; } gain;
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#endif
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};
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#define OPUS_HEAD_SIZE 19
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static av_cold int libopus_decode_init(AVCodecContext *avc)
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{
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struct libopus_context *opus = avc->priv_data;
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int ret, channel_map = 0, gain_db = 0, nb_streams, nb_coupled;
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uint8_t mapping_arr[8] = { 0, 1 }, *mapping;
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avc->sample_rate = 48000;
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avc->sample_fmt = avc->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
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AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
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avc->channel_layout = avc->channels > 8 ? 0 :
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ff_vorbis_channel_layouts[avc->channels - 1];
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if (avc->extradata_size >= OPUS_HEAD_SIZE) {
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opus->pre_skip = AV_RL16(avc->extradata + 10);
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gain_db = sign_extend(AV_RL16(avc->extradata + 16), 16);
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channel_map = AV_RL8 (avc->extradata + 18);
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}
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if (avc->extradata_size >= OPUS_HEAD_SIZE + 2 + avc->channels) {
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nb_streams = avc->extradata[OPUS_HEAD_SIZE + 0];
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nb_coupled = avc->extradata[OPUS_HEAD_SIZE + 1];
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if (nb_streams + nb_coupled != avc->channels)
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av_log(avc, AV_LOG_WARNING, "Inconsistent channel mapping.\n");
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mapping = avc->extradata + OPUS_HEAD_SIZE + 2;
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} else {
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if (avc->channels > 2 || channel_map) {
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av_log(avc, AV_LOG_ERROR,
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"No channel mapping for %d channels.\n", avc->channels);
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return AVERROR(EINVAL);
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}
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nb_streams = 1;
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nb_coupled = avc->channels > 1;
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mapping = mapping_arr;
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}
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if (avc->channels > 2 && avc->channels <= 8) {
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const uint8_t *vorbis_offset = ff_vorbis_channel_layout_offsets[avc->channels - 1];
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int ch;
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/* Remap channels from vorbis order to ffmpeg order */
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for (ch = 0; ch < avc->channels; ch++)
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mapping_arr[ch] = mapping[vorbis_offset[ch]];
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mapping = mapping_arr;
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}
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opus->dec = opus_multistream_decoder_create(avc->sample_rate, avc->channels,
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nb_streams, nb_coupled,
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mapping, &ret);
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if (!opus->dec) {
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av_log(avc, AV_LOG_ERROR, "Unable to create decoder: %s\n",
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opus_strerror(ret));
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return ff_opus_error_to_averror(ret);
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}
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#ifdef OPUS_SET_GAIN
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ret = opus_multistream_decoder_ctl(opus->dec, OPUS_SET_GAIN(gain_db));
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if (ret != OPUS_OK)
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av_log(avc, AV_LOG_WARNING, "Failed to set gain: %s\n",
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opus_strerror(ret));
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#else
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{
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double gain_lin = pow(10, gain_db / (20.0 * 256));
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if (avc->sample_fmt == AV_SAMPLE_FMT_FLT)
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opus->gain.d = gain_lin;
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else
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opus->gain.i = FFMIN(gain_lin * 65536, INT_MAX);
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}
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#endif
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/* Decoder delay (in samples) at 48kHz */
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avc->delay = avc->internal->skip_samples = opus->pre_skip;
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return 0;
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}
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static av_cold int libopus_decode_close(AVCodecContext *avc)
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{
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struct libopus_context *opus = avc->priv_data;
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opus_multistream_decoder_destroy(opus->dec);
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return 0;
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}
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#define MAX_FRAME_SIZE (960 * 6)
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static int libopus_decode(AVCodecContext *avc, void *data,
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int *got_frame_ptr, AVPacket *pkt)
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{
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struct libopus_context *opus = avc->priv_data;
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AVFrame *frame = data;
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int ret, nb_samples;
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frame->nb_samples = MAX_FRAME_SIZE;
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if ((ret = ff_get_buffer(avc, frame, 0)) < 0)
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return ret;
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if (avc->sample_fmt == AV_SAMPLE_FMT_S16)
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nb_samples = opus_multistream_decode(opus->dec, pkt->data, pkt->size,
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(opus_int16 *)frame->data[0],
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frame->nb_samples, 0);
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else
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nb_samples = opus_multistream_decode_float(opus->dec, pkt->data, pkt->size,
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(float *)frame->data[0],
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frame->nb_samples, 0);
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if (nb_samples < 0) {
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av_log(avc, AV_LOG_ERROR, "Decoding error: %s\n",
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opus_strerror(nb_samples));
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return ff_opus_error_to_averror(nb_samples);
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}
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#ifndef OPUS_SET_GAIN
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{
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int i = avc->channels * nb_samples;
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if (avc->sample_fmt == AV_SAMPLE_FMT_FLT) {
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float *pcm = (float *)frame->data[0];
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for (; i > 0; i--, pcm++)
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*pcm = av_clipf(*pcm * opus->gain.d, -1, 1);
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} else {
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int16_t *pcm = (int16_t *)frame->data[0];
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for (; i > 0; i--, pcm++)
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*pcm = av_clip_int16(((int64_t)opus->gain.i * *pcm) >> 16);
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}
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}
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#endif
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frame->nb_samples = nb_samples;
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*got_frame_ptr = 1;
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return pkt->size;
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}
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static void libopus_flush(AVCodecContext *avc)
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{
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struct libopus_context *opus = avc->priv_data;
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opus_multistream_decoder_ctl(opus->dec, OPUS_RESET_STATE);
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/* The stream can have been extracted by a tool that is not Opus-aware.
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Therefore, any packet can become the first of the stream. */
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avc->internal->skip_samples = opus->pre_skip;
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}
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AVCodec ff_libopus_decoder = {
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.name = "libopus",
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.long_name = NULL_IF_CONFIG_SMALL("libopus Opus"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_OPUS,
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.priv_data_size = sizeof(struct libopus_context),
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.init = libopus_decode_init,
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.close = libopus_decode_close,
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.decode = libopus_decode,
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.flush = libopus_flush,
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.capabilities = AV_CODEC_CAP_DR1,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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};
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