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