Mercurial > libavformat.hg
view oggparsevorbis.c @ 4723:a2390c6a35e6 libavformat
Fix index generation in the way that it was supposed to be used. See the
discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading
code".
Over time, this code broke somewhat, e.g. seq was never actually written
into (and was thus always 1, therefore the seq condition was always true),
whereas it was supposed to be set to the sequence number of the video slice
in case the video frame is divided over multiple RM packets (slices). The
problem of this is that packets other than those containing the beginning
of a video frame would be indexed as well.
Secondly, flags&2 is supposed to be true for video keyframes and for these
audio packets containing the start of a block. For some codecs (e.g. AAC),
that is every single packet, whereas for others (e.g. cook), that is the
packet containing the first of a series of scrambled packets that are to be
descrambled together. Indexing any of the following would lead to incomplete
and thus useless frames. Problem here is that flags would be reset to 2 to
indicate that the first packet is ready to be returned, and in addition if
no data was left to be returned (which is always true for the first packet),
then we wouldn't actually write the index entry anyway.
All in all, the idea was good and it probably worked at some point, but that
is long ago. This patch should at the very least make it likely for this code
to be executed again at the right times, i.e. the way it was originally
intended to be used.
author | rbultje |
---|---|
date | Sun, 15 Mar 2009 20:14:25 +0000 |
parents | ad9324c36a3f |
children | 304a0ea063f0 |
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/** Copyright (C) 2005 Michael Ahlberg, Måns Rullgård Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. **/ #include <stdlib.h> #include "libavutil/avstring.h" #include "libavutil/bswap.h" #include "libavcodec/bitstream.h" #include "libavcodec/bytestream.h" #include "avformat.h" #include "oggdec.h" int vorbis_comment(AVFormatContext * as, uint8_t *buf, int size) { 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 = bytestream_get_le32(&p); if (end - p < s) return -1; p += s; n = bytestream_get_le32(&p); while (p < end && n > 0) { const char *t, *v; int tl, vl; s = bytestream_get_le32(&p); if (end - p < s) break; t = p; p += s; n--; v = memchr(t, '=', s); if (!v) continue; tl = v - t; vl = s - tl - 1; v++; if (tl && vl) { 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]); tt[tl] = 0; memcpy(ct, v, vl); ct[vl] = 0; av_metadata_set(&as->metadata, tt, ct); av_freep(&tt); av_freep(&ct); } } 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); return 0; } /** Parse the vorbis header * Vorbis Identification header from Vorbis_I_spec.html#vorbis-spec-codec * [vorbis_version] = read 32 bits as unsigned integer | Not used * [audio_channels] = read 8 bit integer as unsigned | Used * [audio_sample_rate] = read 32 bits as unsigned integer | Used * [bitrate_maximum] = read 32 bits as signed integer | Not used yet * [bitrate_nominal] = read 32 bits as signed integer | Not used yet * [bitrate_minimum] = read 32 bits as signed integer | Used as bitrate * [blocksize_0] = read 4 bits as unsigned integer | Not Used * [blocksize_1] = read 4 bits as unsigned integer | Not Used * [framing_flag] = read one bit | Not Used * */ struct oggvorbis_private { unsigned int len[3]; unsigned char *packet[3]; }; static unsigned int fixup_vorbis_headers(AVFormatContext * as, struct oggvorbis_private *priv, uint8_t **buf) { int i,offset, len; unsigned char *ptr; len = priv->len[0] + priv->len[1] + priv->len[2]; ptr = *buf = av_mallocz(len + len/255 + 64); ptr[0] = 2; offset = 1; offset += av_xiphlacing(&ptr[offset], priv->len[0]); offset += av_xiphlacing(&ptr[offset], priv->len[1]); for (i = 0; i < 3; i++) { memcpy(&ptr[offset], priv->packet[i], priv->len[i]); offset += priv->len[i]; } *buf = av_realloc(*buf, offset + FF_INPUT_BUFFER_PADDING_SIZE); return offset; } static int vorbis_header (AVFormatContext * s, int idx) { struct ogg *ogg = s->priv_data; struct ogg_stream *os = ogg->streams + idx; AVStream *st = s->streams[idx]; struct oggvorbis_private *priv; if (os->seq > 2) return 0; if (os->seq == 0) { os->private = av_mallocz(sizeof(struct oggvorbis_private)); if (!os->private) return 0; } if (os->psize < 1) return -1; priv = os->private; priv->len[os->seq] = os->psize; 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) { const uint8_t *p = os->buf + os->pstart + 7; /* skip "\001vorbis" tag */ unsigned blocksize, bs0, bs1; if (os->psize != 30) return -1; 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; st->time_base.num = 1; st->time_base.den = st->codec->sample_rate; } else if (os->buf[os->pstart] == 3) { if (os->psize > 8) vorbis_comment (s, os->buf + os->pstart + 7, os->psize - 8); } else { st->codec->extradata_size = fixup_vorbis_headers(s, priv, &st->codec->extradata); } return os->seq < 3; } const struct ogg_codec ff_vorbis_codec = { .magic = "\001vorbis", .magicsize = 7, .header = vorbis_header };