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245 lines
8.1 KiB
245 lines
8.1 KiB
/* |
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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 "libavutil/avassert.h" |
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#include "avcodec.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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AVFrame frame; |
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const int8_t *table; |
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/* buffer used to store the whole audio decoded/interleaved chunk, |
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* which is sent with the first packet */ |
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uint8_t *samples; |
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int64_t samples_size; |
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int samples_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 n samples in dst. |
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* |
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* @param val starting value assumed by the delta sequence |
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* @param table delta sequence table |
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* @return size in bytes of the decoded data, must be src_size*2 |
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*/ |
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static int delta_decode(uint8_t *dst, const uint8_t *src, int src_size, |
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unsigned val, const int8_t *table) |
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{ |
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uint8_t *dst0 = dst; |
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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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return dst-dst0; |
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} |
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static void raw_decode(uint8_t *dst, const int8_t *src, int src_size) |
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{ |
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while (src_size--) |
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*dst++ = *src++ + 128; |
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} |
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/** decode a frame */ |
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static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, |
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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 n, out_data_size; |
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int ch, ret; |
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uint8_t *src; |
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/* decode and interleave the first packet */ |
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if (!esc->samples && avpkt) { |
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int packet_size = avpkt->size; |
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if (packet_size % avctx->channels) { |
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av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n"); |
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if (packet_size < avctx->channels) |
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return packet_size; |
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packet_size -= packet_size % avctx->channels; |
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} |
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esc->samples_size = !esc->table ? |
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packet_size : avctx->channels + (packet_size-avctx->channels) * 2; |
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if (!(esc->samples = av_malloc(esc->samples_size))) |
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return AVERROR(ENOMEM); |
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/* decompress */ |
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if (esc->table) { |
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const uint8_t *buf = avpkt->data; |
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uint8_t *dst; |
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int buf_size = avpkt->size; |
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int i, n = esc->samples_size; |
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if (buf_size < 2) { |
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av_log(avctx, AV_LOG_ERROR, "packet size is too small\n"); |
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return AVERROR(EINVAL); |
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} |
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/* the uncompressed starting value is contained in the first byte */ |
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dst = esc->samples; |
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for (i = 0; i < avctx->channels; i++) { |
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*(dst++) = buf[0]+128; |
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delta_decode(dst, buf + 1, buf_size / avctx->channels - 1, (buf[0]+128)&0xFF, esc->table); |
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buf += buf_size / avctx->channels; |
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dst += n / avctx->channels - 1; |
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} |
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} else { |
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raw_decode(esc->samples, avpkt->data, esc->samples_size); |
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} |
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} |
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/* get output buffer */ |
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av_assert1(!(esc->samples_size % avctx->channels || esc->samples_idx % avctx->channels)); |
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esc->frame.nb_samples = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) / avctx->channels; |
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if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) { |
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); |
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return ret; |
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} |
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*got_frame_ptr = 1; |
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*(AVFrame *)data = esc->frame; |
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out_data_size = esc->frame.nb_samples; |
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for (ch = 0; ch<avctx->channels; ch++) { |
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src = esc->samples + esc->samples_idx / avctx->channels + ch * esc->samples_size / avctx->channels; |
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memcpy(esc->frame.data[ch], src, out_data_size); |
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} |
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out_data_size *= avctx->channels; |
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esc->samples_idx += out_data_size; |
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return esc->table ? |
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(avctx->frame_number == 0)*2 + out_data_size / 2 : |
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out_data_size; |
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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->channels < 1 || avctx->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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case AV_CODEC_ID_PCM_S8_PLANAR: |
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case AV_CODEC_ID_8SVX_RAW: esc->table = NULL; break; |
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default: |
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av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id); |
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return AVERROR_INVALIDDATA; |
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} |
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avctx->sample_fmt = AV_SAMPLE_FMT_U8P; |
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avcodec_get_frame_defaults(&esc->frame); |
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avctx->coded_frame = &esc->frame; |
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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->samples); |
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esc->samples_size = 0; |
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esc->samples_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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AVCodec ff_eightsvx_fib_decoder = { |
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.name = "8svx_fib", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.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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.decode = eightsvx_decode_frame, |
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.close = eightsvx_decode_close, |
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.capabilities = CODEC_CAP_DR1, |
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.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), |
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.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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AVCodec ff_eightsvx_exp_decoder = { |
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.name = "8svx_exp", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.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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.decode = eightsvx_decode_frame, |
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.close = eightsvx_decode_close, |
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.capabilities = CODEC_CAP_DR1, |
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.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), |
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.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_PCM_S8_PLANAR_DECODER |
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AVCodec ff_pcm_s8_planar_decoder = { |
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.name = "pcm_s8_planar", |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_PCM_S8_PLANAR, |
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.priv_data_size = sizeof(EightSvxContext), |
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.init = eightsvx_decode_init, |
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.close = eightsvx_decode_close, |
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.decode = eightsvx_decode_frame, |
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.capabilities = CODEC_CAP_DR1, |
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.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"), |
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.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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