they are easier to understand. Also give the add a 'u' postfix to silence
a pre-c99 compiler warning.
Originally committed as revision 22965 to svn://svn.ffmpeg.org/ffmpeg/trunk
This patch also changes FF_NETERROR() to be an AVERROR(), i.e. it is always
negative, whereas it was previously positive.
Originally committed as revision 22887 to svn://svn.ffmpeg.org/ffmpeg/trunk
codec, so there's no reason to (invalidly) limit it to only Theora.
Also fixes issue 1880 (compilation error on -DDEBUG).
Originally committed as revision 22886 to svn://svn.ffmpeg.org/ffmpeg/trunk
we'd memset() the codec context to zero, thereby setting audio input to U8
and video to YUV420P. For most video encoders, that actually works, but for
most audio codecs, it doesn't. This patch changes defaults to those set by
avcodec_context_get_defaults() and have ffmpeg figure out the optimal encoding
format itself if not set explicitely (as it does for the non-ffserver-cases
also).
Originally committed as revision 22751 to svn://svn.ffmpeg.org/ffmpeg/trunk
but doesn't actually do that. What's worse, it creates timestamp adjustments
that are different per stream within a session, leading to a/v sync issues.
See discussion in thread "[FFmpeg-devel] rtp streaming x264+audio issues (and
some ideas to fix them)". Patch suggested by Luca Abeni <lucabe72 email it>.
Originally committed as revision 21857 to svn://svn.ffmpeg.org/ffmpeg/trunk
- the return value of url_open_dyn_*buf() is 0 on success, so using
if (!(res = url_open_dyn_*buf())) return res; is not going to work
- url_open_dyn_packet_buf actually writes the max_packet_size before
each piece of data. Feeding this to the ASF demuxer will never work.
Therefore, use url_open_dyn_buf() instead.
Originally committed as revision 21853 to svn://svn.ffmpeg.org/ffmpeg/trunk
the packet was filled in, leading to virtually random behaviour in the
decoder later on. Instead, return a negative value.
Originally committed as revision 21851 to svn://svn.ffmpeg.org/ffmpeg/trunk
not supported in FFmpeg. This will cause crashes later because the samplerate
is used to initialize the timebase.
Originally committed as revision 21741 to svn://svn.ffmpeg.org/ffmpeg/trunk
the system wants a length entry in all socket structures or not. Patch by
Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21351 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 1658 (the crasher), although the format itself is obviously
still unsupported.
Originally committed as revision 21078 to svn://svn.ffmpeg.org/ffmpeg/trunk
parts of FFmpeg. Also change a starting condition; while (condition) {
... bla = bla->next; } loop into a proper for() loop.
Originally committed as revision 21071 to svn://svn.ffmpeg.org/ffmpeg/trunk
"for (i = 1; i < num + 1; i++)" look like "for (i = 1; i <= num; i++)".
Programmers worldwide rejoiced and used the operand, e.g. in the thread
"[PATCH] remove ugliness in cel_filters.c", and He saw that it was good.
Originally committed as revision 20489 to svn://svn.ffmpeg.org/ffmpeg/trunk
(philip coombes zoneminder com), see "[PATCH]RTSP Basic Authentication"
thread on mailinglist.
Originally committed as revision 19905 to svn://svn.ffmpeg.org/ffmpeg/trunk
used to return packet data, which might update the flags/timestamp to be
used for the next packet data returned by the demuxer. However, that was
separated out into a new function, and the flags/timestamp are thus never
updated within ff_rm_parse_packet() anymore, and thus do not need to be
a pointer.
Originally committed as revision 19539 to svn://svn.ffmpeg.org/ffmpeg/trunk
implement RTCP/bye. See "[PATCH] rtsp.c: EOS support" thread from a few
months back.
Originally committed as revision 19517 to svn://svn.ffmpeg.org/ffmpeg/trunk
discussion in "[PATCH] RTSP-MS 14/15: ASF packet parsing" thread on mailinglist.
Originally committed as revision 19516 to svn://svn.ffmpeg.org/ffmpeg/trunk
AVFormatContext.packet_size. See "[PATCH] asf*.c/h: use
AVFormatContext->packet_size instead of own copy" thread on ML.
Originally committed as revision 19270 to svn://svn.ffmpeg.org/ffmpeg/trunk
value. Currently, this is only used in qcp.c and mpegenc.c, and they are
practically unaffected by this change.
See "[PATCH] make packet_size in AVFormatContext unsigned" thread on ML.
Originally committed as revision 19250 to svn://svn.ffmpeg.org/ffmpeg/trunk
(max packet size should fit in 13 bits as a kB value, so 1<<23, plus the
header which is 10 bytes), and as per mpegenc.c internal assumptions
(min packet size is 20 bytes). See "[PATCH] make packet_size in
AVFormatContext unsigned" thread.
Originally committed as revision 19215 to svn://svn.ffmpeg.org/ffmpeg/trunk
not a stream error. This also fixes an infinite loop on EOS during ASF file
playback, and Michael claims he "likes the patch" (ref: [PATCH] RTSP-MS
15/15: move packet_time_start zero value assignment in asf.c" mailinglist
tread).
Originally committed as revision 18552 to svn://svn.ffmpeg.org/ffmpeg/trunk
thread "[PATCH] RTSP-MS 15/15: move packet_time_start zero value
assignment in asf.c".
Originally committed as revision 18531 to svn://svn.ffmpeg.org/ffmpeg/trunk
Real wants OPTIONS) while the connection is idle, otherwise it will
be aborted after a short period (usually a minute). See the thread
"[PATCH] rtsp.c: keep-alive" on the mailinglist.
Originally committed as revision 18525 to svn://svn.ffmpeg.org/ffmpeg/trunk
qualification task, see "RTP/Vorbis payload implementation (GSoC qual
task)" thread on mailinglist.
Originally committed as revision 18509 to svn://svn.ffmpeg.org/ffmpeg/trunk
buffer size of the fmtp parameter buffer. For Vorbis RT(S)P, these
contain full Vorbis headers, which can be up to 12kb each, formatted
in base64, so 16kb total. Patch required for proper Vorbis/RTP playback,
submitted as GSoC qualification task in the thread "RTP/Vorbis payload
implementation (GSoC qual task)" by Colin McQuillan m.niloc googlemail
com.
Originally committed as revision 18508 to svn://svn.ffmpeg.org/ffmpeg/trunk
Patch (GSoC qualfication task) by Zhentan Feng spyfeng gmail com, see
"[FFmpeg-devel] Add waveformat extensible support in wav muxer (SoC
qualification task)" thread on mailinglist.
Originally committed as revision 18372 to svn://svn.ffmpeg.org/ffmpeg/trunk
See "[FFmpeg-devel] [PATCH] asf fix" thread. Patch contributed by
David DeHaven <dave sagetv com>.
Originally committed as revision 18213 to svn://svn.ffmpeg.org/ffmpeg/trunk
redir_isspace(char) to check if '\0' is a space or not. Therefore, we now
use memchr(), since then we can give the length of the string (i.e. the
length excluding the terminating '\0'). Fixes issue 919, see also the
follow-ups in the "[PATCH] rtsp.c small cleanups" mailinglist thread.
Originally committed as revision 18177 to svn://svn.ffmpeg.org/ffmpeg/trunk
statement (get_word_sep()) already does that all by itself. See summary in
"[PATCH] rtsp.c small cleanups" thread on mailinglist.
Originally committed as revision 18128 to svn://svn.ffmpeg.org/ffmpeg/trunk
same line as the if. See summary in "[PATCH] rtsp.c small cleanups" thread on
mailinglist.
Originally committed as revision 18127 to svn://svn.ffmpeg.org/ffmpeg/trunk
in a stream (e.g. malicious input, broken file, etc.). See summary in "[PATCH]
rtsp.c small cleanups" thread on mailinglist.
Originally committed as revision 18126 to svn://svn.ffmpeg.org/ffmpeg/trunk
it is '\0' rather than its content (char *p; if (p == '\0') instead of if
(*p == '\0')). See summary in "[PATCH] rtsp.c small cleanups" thread on
mailinglist.
Originally committed as revision 18125 to svn://svn.ffmpeg.org/ffmpeg/trunk
Ronald S. Bultje
