diff options
Diffstat (limited to 'libavcodec/8svx.c')
-rw-r--r-- | libavcodec/8svx.c | 236 |
1 files changed, 116 insertions, 120 deletions
diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c index 3e3eae6c87..efe554adc2 100644 --- a/libavcodec/8svx.c +++ b/libavcodec/8svx.c @@ -1,21 +1,21 @@ /* - * 8SVX audio decoder * Copyright (C) 2008 Jaikrishnan Menon + * Copyright (C) 2011 Stefano Sabatini * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -26,166 +26,164 @@ * * 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 "avcodec.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; + /* buffer used to store the whole audio decoded/interleaved chunk, + * which is sent with the first packet */ + uint8_t *samples; + size_t samples_size; + int samples_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 }; +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 32768 +#define MAX_FRAME_SIZE 2048 /** - * Delta decode the compressed values in src, and put the resulting - * decoded samples in dst. + * Interleave samples in buffer containing all left channel samples + * at the beginning, and right channel samples at the end. + * Each sample is assumed to be in signed 8-bit format. * - * @param[in,out] state starting value. it is saved for use in the next call. + * @param size the size in bytes of the dst and src buffer */ -static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size, - uint8_t *state, const int8_t *table, int channels) +static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size) { - uint8_t val = *state; + uint8_t *dst_end = dst + size; + size = size>>1; - while (src_size--) { - uint8_t d = *src++; - val = av_clip_uint8(val + table[d & 0xF]); - *dst = val; - dst += channels; - val = av_clip_uint8(val + table[d >> 4]); - *dst = val; - dst += channels; + while (dst < dst_end) { + *dst++ = *src; + *dst++ = *(src+size); + src++; } - - *state = val; } -static void raw_decode(uint8_t *dst, const int8_t *src, int src_size, - int channels) +/** + * Delta decode the compressed values in src, and put the resulting + * decoded n samples in dst. + * + * @param val starting value assumed by the delta sequence + * @param table delta sequence table + * @return size in bytes of the decoded data, must be src_size*2 + */ +static int delta_decode(int8_t *dst, const uint8_t *src, int src_size, + int8_t val, const int8_t *table) { - while (src_size--) { - *dst = *src++ + 128; - dst += channels; + int n = src_size; + int8_t *dst0 = dst; + + while (n--) { + uint8_t d = *src++; + val = av_clip(val + table[d & 0x0f], -127, 128); + *dst++ = val; + val = av_clip(val + table[d >> 4] , -127, 128); + *dst++ = val; } + + return dst-dst0; } -/** decode a frame */ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { EightSvxContext *esc = avctx->priv_data; - int buf_size; - uint8_t *out_data = data; - int out_data_size; - int is_compr = (avctx->codec_id != CODEC_ID_PCM_S8_PLANAR); - - /* for the first packet, copy data to buffer */ - if (avpkt->data) { - int hdr_size = is_compr ? 2 : 0; - int chan_size = (avpkt->size - hdr_size * avctx->channels) / avctx->channels; - - if (avpkt->size < hdr_size * avctx->channels) { - av_log(avctx, AV_LOG_ERROR, "packet size is too small\n"); - return AVERROR(EINVAL); - } - if (esc->data[0]) { - av_log(avctx, AV_LOG_ERROR, "unexpected data after first packet\n"); - return AVERROR(EINVAL); - } + int out_data_size, n; + uint8_t *src, *dst; - if (is_compr) { - esc->fib_acc[0] = avpkt->data[1] + 128; - if (avctx->channels == 2) - esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128; - } + /* decode and interleave the first packet */ + if (!esc->samples && avpkt) { + uint8_t *deinterleaved_samples; - esc->data_idx = 0; - esc->data_size = chan_size; - if (!(esc->data[0] = av_malloc(chan_size))) + esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR? + avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2; + if (!(esc->samples = av_malloc(esc->samples_size))) return AVERROR(ENOMEM); - if (avctx->channels == 2) { - if (!(esc->data[1] = av_malloc(chan_size))) { - av_freep(&esc->data[0]); - return AVERROR(ENOMEM); + + /* decompress */ + if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) { + const uint8_t *buf = avpkt->data; + int buf_size = avpkt->size; + int n = esc->samples_size; + + if (buf_size < 2) { + av_log(avctx, AV_LOG_ERROR, "packet size is too small\n"); + return AVERROR(EINVAL); } + if (!(deinterleaved_samples = av_mallocz(n))) + return AVERROR(ENOMEM); + + /* the uncompressed starting value is contained in the first byte */ + if (avctx->channels == 2) { + delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table); + buf += buf_size/2; + delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table); + } else + delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table); + } else { + deinterleaved_samples = avpkt->data; } - memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size); + if (avctx->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(EINVAL); + interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size); + else + memcpy(esc->samples, deinterleaved_samples, esc->samples_size); } - /* decode next piece of data from the buffer */ - buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx); - if (buf_size <= 0) { - *data_size = 0; - return avpkt->size; - } - out_data_size = buf_size * (is_compr + 1) * avctx->channels; + /* return single packed with fixed size */ + out_data_size = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx); if (*data_size < out_data_size) { - av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", - *data_size); + av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n", *data_size); return AVERROR(EINVAL); } - if (is_compr) { - delta_decode(out_data, &esc->data[0][esc->data_idx], buf_size, - &esc->fib_acc[0], esc->table, avctx->channels); - if (avctx->channels == 2) { - delta_decode(&out_data[1], &esc->data[1][esc->data_idx], buf_size, - &esc->fib_acc[1], esc->table, avctx->channels); - } - } else { - int ch; - for (ch = 0; ch < avctx->channels; ch++) { - raw_decode((int8_t *)&out_data[ch], &esc->data[ch][esc->data_idx], - buf_size, avctx->channels); - } - } - esc->data_idx += buf_size; - *data_size = out_data_size; - return avpkt->size; + *data_size = out_data_size; + dst = data; + src = esc->samples + esc->samples_idx; + for (n = out_data_size; n > 0; n--) + *dst++ = *src++ + 128; + esc->samples_idx += *data_size; + + return avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ? + (avctx->frame_number == 0)*2 + out_data_size / 2 : + out_data_size; } -/** initialize 8svx decoder */ static av_cold int eightsvx_decode_init(AVCodecContext *avctx) { EightSvxContext *esc = avctx->priv_data; if (avctx->channels < 1 || avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n"); - return AVERROR(EINVAL); + return AVERROR_INVALIDDATA; } - switch(avctx->codec->id) { - case CODEC_ID_8SVX_FIB: - esc->table = fibonacci; - break; - case CODEC_ID_8SVX_EXP: - esc->table = exponential; - break; - case CODEC_ID_PCM_S8_PLANAR: - break; - default: - return -1; + switch (avctx->codec->id) { + case CODEC_ID_8SVX_FIB: esc->table = fibonacci; break; + case CODEC_ID_8SVX_EXP: esc->table = exponential; break; + case CODEC_ID_PCM_S8_PLANAR: + case CODEC_ID_8SVX_RAW: esc->table = NULL; break; + default: + av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id); + return AVERROR_INVALIDDATA; } avctx->sample_fmt = AV_SAMPLE_FMT_U8; + return 0; } @@ -193,8 +191,9 @@ static av_cold int eightsvx_decode_close(AVCodecContext *avctx) { EightSvxContext *esc = avctx->priv_data; - av_freep(&esc->data[0]); - av_freep(&esc->data[1]); + av_freep(&esc->samples); + esc->samples_size = 0; + esc->samples_idx = 0; return 0; } @@ -205,9 +204,8 @@ AVCodec ff_eightsvx_fib_decoder = { .id = CODEC_ID_8SVX_FIB, .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, - .close = eightsvx_decode_close, .decode = eightsvx_decode_frame, - .capabilities = CODEC_CAP_DELAY, + .close = eightsvx_decode_close, .long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"), }; @@ -217,9 +215,8 @@ AVCodec ff_eightsvx_exp_decoder = { .id = CODEC_ID_8SVX_EXP, .priv_data_size = sizeof (EightSvxContext), .init = eightsvx_decode_init, - .close = eightsvx_decode_close, .decode = eightsvx_decode_frame, - .capabilities = CODEC_CAP_DELAY, + .close = eightsvx_decode_close, .long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"), }; @@ -231,6 +228,5 @@ AVCodec ff_pcm_s8_planar_decoder = { .init = eightsvx_decode_init, .close = eightsvx_decode_close, .decode = eightsvx_decode_frame, - .capabilities = CODEC_CAP_DELAY, .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"), }; |