mirror of https://git.ffmpeg.org/ffmpeg.git
220 lines
6.7 KiB
C
220 lines
6.7 KiB
C
/*
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* Copyright (C) 2008 Jaikrishnan Menon
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* Copyright (C) 2011 Stefano Sabatini
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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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/**
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* @file
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* 8svx audio decoder
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* @author Jaikrishnan Menon
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*
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* supports: fibonacci delta encoding
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* : exponential encoding
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*
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* For more information about the 8SVX format:
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* http://netghost.narod.ru/gff/vendspec/iff/iff.txt
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* http://sox.sourceforge.net/AudioFormats-11.html
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* http://aminet.net/package/mus/misc/wavepak
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* http://amigan.1emu.net/reg/8SVX.txt
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*
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* Samples can be found here:
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* http://aminet.net/mods/smpl/
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*/
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#include "config_components.h"
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#include "libavutil/avassert.h"
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#include "libavutil/mem.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "decode.h"
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#include "libavutil/common.h"
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/** decoder context */
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typedef struct EightSvxContext {
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uint8_t fib_acc[2];
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const int8_t *table;
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/* buffer used to store the whole first packet.
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data is only sent as one large packet */
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uint8_t *data[2];
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int data_size;
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int data_idx;
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} EightSvxContext;
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static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
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static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
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#define MAX_FRAME_SIZE 2048
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/**
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* Delta decode the compressed values in src, and put the resulting
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* decoded samples in dst.
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*
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* @param[in,out] state starting value. it is saved for use in the next call.
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* @param table delta sequence table
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*/
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static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
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uint8_t *state, const int8_t *table)
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{
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uint8_t val = *state;
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while (src_size--) {
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uint8_t d = *src++;
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val = av_clip_uint8(val + table[d & 0xF]);
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*dst++ = val;
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val = av_clip_uint8(val + table[d >> 4]);
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*dst++ = val;
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}
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*state = val;
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}
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/** decode a frame */
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static int eightsvx_decode_frame(AVCodecContext *avctx, AVFrame *frame,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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EightSvxContext *esc = avctx->priv_data;
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int channels = avctx->ch_layout.nb_channels;
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int buf_size;
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int ch, ret;
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int hdr_size = 2;
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/* decode and interleave the first packet */
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if (!esc->data[0] && avpkt) {
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int chan_size = avpkt->size / channels - hdr_size;
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if (avpkt->size % channels) {
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av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
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}
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if (avpkt->size < (hdr_size + 1) * channels) {
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av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
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return AVERROR_INVALIDDATA;
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}
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esc->fib_acc[0] = avpkt->data[1] + 128;
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if (channels == 2)
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esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;
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esc->data_idx = 0;
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esc->data_size = chan_size;
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if (!(esc->data[0] = av_malloc(chan_size)))
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return AVERROR(ENOMEM);
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if (channels == 2) {
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if (!(esc->data[1] = av_malloc(chan_size))) {
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av_freep(&esc->data[0]);
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return AVERROR(ENOMEM);
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}
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}
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memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size);
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if (channels == 2)
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memcpy(esc->data[1], &avpkt->data[2*hdr_size+chan_size], chan_size);
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}
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if (!esc->data[0]) {
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av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
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return AVERROR_INVALIDDATA;
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}
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/* decode next piece of data from the buffer */
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buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
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if (buf_size <= 0) {
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*got_frame_ptr = 0;
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return avpkt->size;
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}
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/* get output buffer */
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frame->nb_samples = buf_size * 2;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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for (ch = 0; ch < channels; ch++) {
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delta_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
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buf_size, &esc->fib_acc[ch], esc->table);
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}
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esc->data_idx += buf_size;
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*got_frame_ptr = 1;
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return ((avctx->frame_num == 0) * hdr_size + buf_size) * channels;
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}
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static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
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{
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EightSvxContext *esc = avctx->priv_data;
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if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
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return AVERROR_INVALIDDATA;
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}
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switch (avctx->codec->id) {
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case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
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case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
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default:
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av_assert1(0);
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
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return 0;
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}
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static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
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{
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EightSvxContext *esc = avctx->priv_data;
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av_freep(&esc->data[0]);
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av_freep(&esc->data[1]);
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esc->data_size = 0;
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esc->data_idx = 0;
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return 0;
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}
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#if CONFIG_EIGHTSVX_FIB_DECODER
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const FFCodec ff_eightsvx_fib_decoder = {
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.p.name = "8svx_fib",
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CODEC_LONG_NAME("8SVX fibonacci"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_8SVX_FIB,
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.priv_data_size = sizeof (EightSvxContext),
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.init = eightsvx_decode_init,
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FF_CODEC_DECODE_CB(eightsvx_decode_frame),
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.close = eightsvx_decode_close,
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.p.capabilities = AV_CODEC_CAP_DR1,
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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#if CONFIG_EIGHTSVX_EXP_DECODER
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const FFCodec ff_eightsvx_exp_decoder = {
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.p.name = "8svx_exp",
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CODEC_LONG_NAME("8SVX exponential"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_8SVX_EXP,
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.priv_data_size = sizeof (EightSvxContext),
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.init = eightsvx_decode_init,
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FF_CODEC_DECODE_CB(eightsvx_decode_frame),
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.close = eightsvx_decode_close,
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.p.capabilities = AV_CODEC_CAP_DR1,
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.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_NONE },
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};
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#endif
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