**/
#include <stdlib.h>
+#include "libavutil/avstring.h"
+#include "libavutil/bswap.h"
+#include "libavcodec/get_bits.h"
+#include "libavcodec/bytestream.h"
#include "avformat.h"
-#include "bitstream.h"
-#include "bswap.h"
-#include "ogg2.h"
-#include "avstring.h"
+#include "oggdec.h"
-extern int
+int
vorbis_comment(AVFormatContext * as, uint8_t *buf, int size)
{
- uint8_t *p = buf;
+ const uint8_t *p = buf;
+ const uint8_t *end = buf + size;
unsigned s, n, j;
if (size < 8) /* must have vendor_length and user_comment_list_length */
return -1;
- s = AV_RL32(p);
- p += 4;
- size -= 4;
+ s = bytestream_get_le32(&p);
- if (size - 4 < s)
+ if (end - p < s)
return -1;
p += s;
- size -= s;
- n = AV_RL32(p);
- p += 4;
- size -= 4;
+ n = bytestream_get_le32(&p);
- while (size >= 4) {
- char *t, *v;
+ while (p < end && n > 0) {
+ const char *t, *v;
int tl, vl;
- s = AV_RL32(p);
- p += 4;
- size -= 4;
+ s = bytestream_get_le32(&p);
- if (size < s)
+ if (end - p < s)
break;
t = p;
p += s;
- size -= s;
n--;
v = memchr(t, '=', s);
v++;
if (tl && vl) {
- char tt[tl + 1];
- char ct[vl + 1];
+ char *tt, *ct;
+
+ tt = av_malloc(tl + 1);
+ ct = av_malloc(vl + 1);
+ if (!tt || !ct) {
+ av_freep(&tt);
+ av_freep(&ct);
+ av_log(as, AV_LOG_WARNING, "out-of-memory error. skipping VorbisComment tag.\n");
+ continue;
+ }
for (j = 0; j < tl; j++)
tt[j] = toupper(t[j]);
memcpy(ct, v, vl);
ct[vl] = 0;
- // took from Vorbis_I_spec
- if (!strcmp(tt, "AUTHOR") || !strcmp(tt, "ARTIST"))
- av_strlcpy(as->author, ct, sizeof(as->author));
- else if (!strcmp(tt, "TITLE"))
- av_strlcpy(as->title, ct, sizeof(as->title));
- else if (!strcmp(tt, "COPYRIGHT"))
- av_strlcpy(as->copyright, ct, sizeof(as->copyright));
- else if (!strcmp(tt, "DESCRIPTION"))
- av_strlcpy(as->comment, ct, sizeof(as->comment));
- else if (!strcmp(tt, "GENRE"))
- av_strlcpy(as->genre, ct, sizeof(as->genre));
- else if (!strcmp(tt, "TRACKNUMBER"))
- as->track = atoi(ct);
- else if (!strcmp(tt, "ALBUM"))
- av_strlcpy(as->album, ct, sizeof(as->album));
+ av_metadata_set(&as->metadata, tt, ct);
+
+ av_freep(&tt);
+ av_freep(&ct);
}
}
- if (size > 0)
- av_log(as, AV_LOG_INFO, "%i bytes of comment header remain\n", size);
+ if (p != end)
+ av_log(as, AV_LOG_INFO, "%ti bytes of comment header remain\n", p-end);
if (n > 0)
av_log(as, AV_LOG_INFO,
"truncated comment header, %i comments not found\n", n);
* [framing_flag] = read one bit | Not Used
* */
-typedef struct {
+struct oggvorbis_private {
unsigned int len[3];
unsigned char *packet[3];
-} oggvorbis_private_t;
+};
static unsigned int
-fixup_vorbis_headers(AVFormatContext * as, oggvorbis_private_t *priv,
+fixup_vorbis_headers(AVFormatContext * as, struct oggvorbis_private *priv,
uint8_t **buf)
{
int i,offset, len;
memcpy(&ptr[offset], priv->packet[i], priv->len[i]);
offset += priv->len[i];
}
- *buf = av_realloc(*buf, offset);
+ *buf = av_realloc(*buf, offset + FF_INPUT_BUFFER_PADDING_SIZE);
return offset;
}
static int
vorbis_header (AVFormatContext * s, int idx)
{
- ogg_t *ogg = s->priv_data;
- ogg_stream_t *os = ogg->streams + idx;
+ struct ogg *ogg = s->priv_data;
+ struct ogg_stream *os = ogg->streams + idx;
AVStream *st = s->streams[idx];
- oggvorbis_private_t *priv;
+ struct oggvorbis_private *priv;
if (os->seq > 2)
return 0;
if (os->seq == 0) {
- os->private = av_mallocz(sizeof(oggvorbis_private_t));
+ os->private = av_mallocz(sizeof(struct oggvorbis_private));
if (!os->private)
return 0;
}
priv->packet[os->seq] = av_mallocz(os->psize);
memcpy(priv->packet[os->seq], os->buf + os->pstart, os->psize);
if (os->buf[os->pstart] == 1) {
- uint8_t *p = os->buf + os->pstart + 11; //skip up to the audio channels
+ const uint8_t *p = os->buf + os->pstart + 7; /* skip "\001vorbis" tag */
+ unsigned blocksize, bs0, bs1;
if (os->psize != 30)
return -1;
- st->codec->channels = *p++;
- st->codec->sample_rate = AV_RL32(p);
- p += 8; //skip maximum and and nominal bitrate
- st->codec->bit_rate = AV_RL32(p); //Minimum bitrate
+ if (bytestream_get_le32(&p) != 0) /* vorbis_version */
+ return -1;
+
+ st->codec->channels = bytestream_get_byte(&p);
+ st->codec->sample_rate = bytestream_get_le32(&p);
+ p += 4; // skip maximum bitrate
+ st->codec->bit_rate = bytestream_get_le32(&p); // nominal bitrate
+ p += 4; // skip minimum bitrate
+
+ blocksize = bytestream_get_byte(&p);
+ bs0 = blocksize & 15;
+ bs1 = blocksize >> 4;
+
+ if (bs0 > bs1)
+ return -1;
+ if (bs0 < 6 || bs1 > 13)
+ return -1;
+
+ if (bytestream_get_byte(&p) != 1) /* framing_flag */
+ return -1;
st->codec->codec_type = CODEC_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_VORBIS;
return os->seq < 3;
}
-ogg_codec_t vorbis_codec = {
+const struct ogg_codec ff_vorbis_codec = {
.magic = "\001vorbis",
.magicsize = 7,
.header = vorbis_header
projects / frescor / ffmpeg.git / blobdiff