/* * 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 "libavutil/avassert.h" #include "avcodec.h" #include "internal.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, void *data, int *got_frame_ptr, AVPacket *avpkt) { EightSvxContext *esc = avctx->priv_data; AVFrame *frame = 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_number == 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 AVCodec ff_eightsvx_fib_decoder = { .name = "8svx_fib", .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_8SVX_FIB, .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, .close = eightsvx_decode_close, .capabilities = AV_CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_NONE }, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, }; #endif #if CONFIG_EIGHTSVX_EXP_DECODER const AVCodec ff_eightsvx_exp_decoder = { .name = "8svx_exp", .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_8SVX_EXP, .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, .decode = eightsvx_decode_frame, .close = eightsvx_decode_close, .capabilities = AV_CODEC_CAP_DR1, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_NONE }, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, }; #endif