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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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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