246 lines
8.0 KiB
C
246 lines
8.0 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 "avcodec.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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size_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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* Interleave samples in buffer containing all left channel samples
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* at the beginning, and right channel samples at the end.
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* Each sample is assumed to be in signed 8-bit format.
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*
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* @param size the size in bytes of the dst and src buffer
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*/
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static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
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{
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uint8_t *dst_end = dst + size;
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size = size>>1;
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while (dst < dst_end) {
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*dst++ = *src;
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*dst++ = *(src+size);
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src++;
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}
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}
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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(int8_t *dst, const uint8_t *src, int src_size,
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int8_t val, const int8_t *table)
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{
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int n = src_size;
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int8_t *dst0 = dst;
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while (n--) {
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uint8_t d = *src++;
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val = av_clip(val + table[d & 0x0f], -127, 128);
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*dst++ = val;
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val = av_clip(val + table[d >> 4] , -127, 128);
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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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/** 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, ret;
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uint8_t *src, *dst;
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/* decode and interleave the first packet */
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if (!esc->samples && avpkt) {
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uint8_t *deinterleaved_samples, *p = NULL;
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esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
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avpkt->size : avctx->channels + (avpkt->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 (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int 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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if (!(deinterleaved_samples = av_mallocz(n)))
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return AVERROR(ENOMEM);
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p = deinterleaved_samples;
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/* the uncompressed starting value is contained in the first byte */
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if (avctx->channels == 2) {
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delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
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buf += buf_size/2;
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delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
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} else
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delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
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} else {
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deinterleaved_samples = avpkt->data;
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}
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if (avctx->channels == 2)
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interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
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else
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memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
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av_freep(&p);
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}
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/* get output buffer */
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esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / 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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dst = esc->frame.data[0];
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src = esc->samples + esc->samples_idx;
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out_data_size = esc->frame.nb_samples * avctx->channels;
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for (n = out_data_size; n > 0; n--)
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*dst++ = *src++ + 128;
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esc->samples_idx += out_data_size;
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return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
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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 CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
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case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
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case CODEC_ID_PCM_S8_PLANAR:
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case 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_U8;
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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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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 = 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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};
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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 = 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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};
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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 = 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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};
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