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-rw-r--r--libavcodec/8svx.c253
1 files changed, 133 insertions, 120 deletions
diff --git a/libavcodec/8svx.c b/libavcodec/8svx.c
index 328fc65b41..f41b19fd58 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
*/
@@ -26,62 +26,80 @@
*
* 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 "libavutil/common.h"
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
- 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;
+ int64_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 */
@@ -89,108 +107,98 @@ static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
- int buf_size;
- uint8_t *out_data;
- int ret;
- int is_compr = (avctx->codec_id != AV_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 n, out_data_size, ret;
+ 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, *p = NULL;
+ int packet_size = avpkt->size;
- esc->data_idx = 0;
- esc->data_size = chan_size;
- if (!(esc->data[0] = av_malloc(chan_size)))
+ if (packet_size % avctx->channels) {
+ av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
+ if (packet_size < avctx->channels)
+ return packet_size;
+ packet_size -= packet_size % avctx->channels;
+ }
+ esc->samples_size = !esc->table ?
+ packet_size : avctx->channels + (packet_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]);
+
+ /* decompress */
+ if (esc->table) {
+ const uint8_t *buf = avpkt->data;
+ uint8_t *dst;
+ int buf_size = avpkt->size;
+ int i, 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);
+ dst = p = deinterleaved_samples;
+
+ /* the uncompressed starting value is contained in the first byte */
+ dst = deinterleaved_samples;
+ for (i = 0; i < avctx->channels; i++) {
+ delta_decode(dst, buf + 1, buf_size / avctx->channels - 1, buf[0], esc->table);
+ buf += buf_size / avctx->channels;
+ dst += n / avctx->channels - 1;
}
+ } 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);
- }
- /* 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;
+ if (avctx->channels == 2)
+ interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
+ else
+ memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
+ av_freep(&p);
}
/* get output buffer */
- esc->frame.nb_samples = buf_size * (is_compr + 1);
+ av_assert1(!(esc->samples_size % avctx->channels || esc->samples_idx % avctx->channels));
+ esc->frame.nb_samples = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
- out_data = esc->frame.data[0];
-
- 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;
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
- return avpkt->size;
+ dst = esc->frame.data[0];
+ src = esc->samples + esc->samples_idx;
+ out_data_size = esc->frame.nb_samples * avctx->channels;
+ for (n = out_data_size; n > 0; n--)
+ *dst++ = *src++ + 128;
+ esc->samples_idx += out_data_size;
+
+ return esc->table ?
+ (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 AV_CODEC_ID_8SVX_FIB:
- esc->table = fibonacci;
- break;
- case AV_CODEC_ID_8SVX_EXP:
- esc->table = exponential;
- break;
- case AV_CODEC_ID_PCM_S8_PLANAR:
- break;
- default:
- return -1;
+ switch (avctx->codec->id) {
+ case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
+ case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
+ case AV_CODEC_ID_PCM_S8_PLANAR:
+ case AV_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;
@@ -204,36 +212,40 @@ 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;
}
+#if CONFIG_EIGHTSVX_FIB_DECODER
AVCodec ff_eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_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 | CODEC_CAP_DR1,
+ .close = eightsvx_decode_close,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
};
-
+#endif
+#if CONFIG_EIGHTSVX_EXP_DECODER
AVCodec ff_eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_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 | CODEC_CAP_DR1,
+ .close = eightsvx_decode_close,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
};
-
+#endif
+#if CONFIG_PCM_S8_PLANAR_DECODER
AVCodec ff_pcm_s8_planar_decoder = {
.name = "pcm_s8_planar",
.type = AVMEDIA_TYPE_AUDIO,
@@ -242,6 +254,7 @@ AVCodec ff_pcm_s8_planar_decoder = {
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
- .capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
};
+#endif
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-10 05:54:17 +0000