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a247ac640d
Given that the AVCodec.next pointer has now been removed, most of the AVCodecs are not modified at all any more and can therefore be made const (as this patch does); the only exceptions are the very few codecs for external libraries that have a init_static_data callback. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
216 lines
6.7 KiB
C
216 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 "libavutil/avassert.h"
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#include "avcodec.h"
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#include "internal.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, 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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AVFrame *frame = data;
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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 / avctx->channels - hdr_size;
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if (avpkt->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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}
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if (avpkt->size < (hdr_size + 1) * avctx->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 (avctx->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 (avctx->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 (avctx->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 < avctx->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_number == 0)*hdr_size + buf_size)*avctx->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->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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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 AVCodec ff_eightsvx_fib_decoder = {
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.name = "8svx_fib",
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.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
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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 = AV_CODEC_CAP_DR1,
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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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const AVCodec ff_eightsvx_exp_decoder = {
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.name = "8svx_exp",
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.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
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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 = AV_CODEC_CAP_DR1,
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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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