/*
- * 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
*/
*
* 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;
const int8_t *table;
+
+ /* 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 2048
+
+/**
+ * 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 size the size in bytes of the dst and src buffer
+ */
+static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
+{
+ uint8_t *dst_end = dst + size;
+ size = size>>1;
+
+ while (dst < dst_end) {
+ *dst++ = *src;
+ *dst++ = *(src+size);
+ src++;
+ }
+}
/**
* Delta decode the compressed values in src, and put the resulting
- * decoded samples in dst.
+ * decoded n samples in dst.
*
- * @param[in,out] state starting value. it is saved for use in the next call.
+ * @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 void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
- uint8_t *state, const int8_t *table)
+static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
+ int8_t val, const int8_t *table)
{
- uint8_t val = *state;
+ int n = src_size;
+ int8_t *dst0 = dst;
- while (src_size--) {
+ while (n--) {
uint8_t d = *src++;
- val = av_clip_uint8(val + table[d & 0xF]);
+ val = av_clip(val + table[d & 0x0f], -127, 128);
*dst++ = val;
- val = av_clip_uint8(val + table[d >> 4]);
+ val = av_clip(val + table[d >> 4] , -127, 128);
*dst++ = val;
}
- *state = val;
+ return dst-dst0;
}
-/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
{
- const uint8_t *buf = avpkt->data;
- int buf_size = avpkt->size;
EightSvxContext *esc = avctx->priv_data;
- uint8_t *out_data = data;
- int consumed = buf_size;
-
- if(avctx->frame_number == 0) {
- if (buf_size < 2) {
- av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
- return AVERROR(EINVAL);
+ int out_data_size, n;
+ uint8_t *src, *dst;
+
+ /* decode and interleave the first packet */
+ if (!esc->samples && avpkt) {
+ uint8_t *deinterleaved_samples;
+
+ 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);
+
+ /* 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;
}
- esc->fib_acc = (int8_t)buf[1] + 128;
- buf_size -= 2;
- buf += 2;
+
+ if (avctx->channels == 2)
+ interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
+ else
+ memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
}
- if (*data_size < buf_size * 2) {
- av_log(avctx, AV_LOG_ERROR, "Provided buffer with size %d is too small.\n",
- *data_size);
+ /* 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);
return AVERROR(EINVAL);
}
- delta_decode(out_data, buf, buf_size, &esc->fib_acc, esc->table);
+ *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;
- *data_size = buf_size * 2;
-
- return consumed;
+ 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;
- switch(avctx->codec->id) {
- case CODEC_ID_8SVX_FIB:
- esc->table = fibonacci;
- break;
- case CODEC_ID_8SVX_EXP:
- esc->table = exponential;
- break;
- default:
- return -1;
+ if (avctx->channels < 1 || avctx->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 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;
+}
+
+static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
+{
+ EightSvxContext *esc = avctx->priv_data;
+
+ av_freep(&esc->samples);
+ esc->samples_size = 0;
+ esc->samples_idx = 0;
+
return 0;
}
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
+ .close = eightsvx_decode_close,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
+ .close = eightsvx_decode_close,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
+
+AVCodec ff_pcm_s8_planar_decoder = {
+ .name = "pcm_s8_planar",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_PCM_S8_PLANAR,
+ .priv_data_size = sizeof(EightSvxContext),
+ .init = eightsvx_decode_init,
+ .close = eightsvx_decode_close,
+ .decode = eightsvx_decode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
+};
projects / ffmpeg / blobdiff