240 lines
7.5 KiB
C
240 lines
7.5 KiB
C
/*
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* 8SVX audio decoder
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* Copyright (C) 2008 Jaikrishnan Menon
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*
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* This file is part of Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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#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,
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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,
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0, 1, 2, 4, 8, 16, 32, 64 };
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#define MAX_FRAME_SIZE 32768
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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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*/
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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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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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AVFrame *frame = data;
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int buf_size;
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int ch, ret;
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int is_compr = (avctx->codec_id != AV_CODEC_ID_PCM_S8_PLANAR);
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/* for the first packet, copy data to buffer */
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if (avpkt->data) {
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int hdr_size = is_compr ? 2 : 0;
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int chan_size = (avpkt->size - hdr_size * avctx->channels) / avctx->channels;
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if (avpkt->size < hdr_size * avctx->channels) {
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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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if (esc->data[0]) {
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av_log(avctx, AV_LOG_ERROR, "unexpected data after first packet\n");
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return AVERROR(EINVAL);
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}
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if (is_compr) {
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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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}
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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(EINVAL);
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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 * (is_compr + 1);
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 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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for (ch = 0; ch < avctx->channels; ch++) {
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if (is_compr) {
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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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} else {
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raw_decode(frame->data[ch], &esc->data[ch][esc->data_idx],
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buf_size);
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}
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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 avpkt->size;
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}
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/** initialize 8svx decoder */
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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(EINVAL);
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}
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switch(avctx->codec->id) {
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case AV_CODEC_ID_8SVX_FIB:
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esc->table = fibonacci;
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break;
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case AV_CODEC_ID_8SVX_EXP:
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esc->table = exponential;
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break;
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case AV_CODEC_ID_PCM_S8_PLANAR:
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break;
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default:
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return -1;
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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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return 0;
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}
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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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.close = eightsvx_decode_close,
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.decode = eightsvx_decode_frame,
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.capabilities = CODEC_CAP_DELAY | 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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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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.close = eightsvx_decode_close,
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.decode = eightsvx_decode_frame,
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.capabilities = CODEC_CAP_DELAY | 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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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_DELAY | 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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