* commit 'e02dcdf6bb6835ef4b49986b85a67efcb3495a7f':
rtsp: Allow $ as interleaved packet indicator before a complete response header
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
Some RTSP servers ("HiIpcam/V100R003 VodServer/1.0.0") respond to
our keepalive GET_PARAMETER request by a truncated RTSP header
(lacking the final empty line to indicate a complete response
header). Prior to 764ec70149, this worked just fine since we
reacted to the $ as interleaved packet indicator anywhere.
Since $ is a valid character within the response header lines,
764ec70149 changed it to be ignored there. But to keep
compatibility with such broken servers, we need to at least
allow reacting to it at the start of lines.
Signed-off-by: Martin Storsjö <martin@martin.st>
packets are queued due to packet reordering until the queue reach its
maximal size or max delay is reached.
This commit adds a warning trace when max delay is reached.
Signed-off-by: Eloi BAIL <eloi.bail@savoirfairelinux.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '764ec70149728be82304c163ccc4e280f1629201':
rtsp: Only interpret $ as interleaved packet indicator at the start of replies
Merged-by: Michael Niedermayer <michael@niedermayer.cc>
Allow $ as character anywhere within normal RTSP replies - both
within the lines, and as the first character of RTSP header lines.
(The existing old comment indicated that an inline packet could
start at any line within a RTSP reply header, but that doesn't
sound valid to me, and I'm not sure if the existing code
handled that correctly either.)
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'b90adb0aba073f9c1b4abca852119947393ced4c':
rtsp: Make sure we don't write too many transport entries into a fixed-size array
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This removes the error logging added in 4e54432164.
This avoids warnings about "Invalid interval start specification 'now'"
for live rtsp streams.
We only try to parse some of the many valid values for time ranges
in RTSP - the other ones are fully valid but not interesting for the
use case in rtsp.c, so we shouldn't warn about them.
(Parsing the time ranges is needed to allow seeking, but e.g. setting
the current realtime clock for the start time doesn't make sense.
av_parse_time has got a different mode for parsing absolute times
as well, which can handle the special case "now", but that doesn't
make much sense for this particular use in rtsp.c.)
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts commit 26524e3581.
If we want the T.140 codec to have the AV_CODEC_ID_TEXT codec id,
its type needs to be AVMEDIA_TYPE_SUBTITLE, so, keep interpreting
the text media type as AVMEDIA_TYPE_SUBTITLE.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '8e32b1f0963d01d4f5d4803eb721f162e0d58d9a':
libavformat: Use ffio_free_dyn_buf where applicable
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '078d43e23a7a3d64aafee8a58b380d3e139b3020':
rtpdec: Free depacketizers if the init function failed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bb4a310bb85f43e62240145a656b1e5285b14239':
rtpdec: Don't free the payload context in the .free function
Conflicts:
libavformat/rtpdec_latm.c
libavformat/rtpdec_mpeg4.c
libavformat/rtpdec_mpegts.c
libavformat/rtpdec_xiph.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e72605f80bf5cbe32053a554ccc137e0a99cf3dd':
rtpdec: Allow allocating and freeing the private data without explicit functions
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b7a4c319fda22aa91ce29692d728ec6103b514f6':
rtpdec: Allow setting the need_parsing field in RTPDynamicProtocolHandler
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Many of these functions were named foo_free_context, and since
the functions no longer should free the context itself, only
allocated elements within it, the previous naming was slightly
misleading.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from how it is handled in codecs/demuxers/muxers
though (where the close function isn't called if the open function
failed), but since the number of depacketizers that have an .init
function is quite limited, this is easy to change.
The main point is that if the init function failed, we shouldn't
try to use that depacketizer at all - this makes sure that the
parse function doesn't need to check for the things that were
initialized in the init function.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes it more consistent with depacketizers that don't have any
.free function at all, where the payload context is freed by the
surrounding framework. Always free the context in the surrounding
framework, having the individual depacketizers only free any data
they've specifically allocated themselves.
This is similar to how this works for demuxer/muxers/codecs - a
component shouldn't free the priv_data that the framework has
allocated for it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '74d318f138f2a3f1b2fe81aea826d80d1e60f54c':
rtsp: Fix the indentation of a linewrapped statement
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '26524e358147aade6e9dd18fff42d61b966bbc70':
rtsp: Interpret the text media type as AVMEDIA_TYPE_DATA
See: afb0e5a810
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows getting rid of quite a bit of boilerplate in depacketizers.
The default value (initializing need_parsing to 0, aka
AVSTREAM_PARSE_NONE) is the same as it is initialized to by default
in AVStream.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'cdcc370293a159c321e41af7f0eef141c62d698d':
rtsp: punch holes again after pause
See: 22bb5bd7a3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.
Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.
Signed-off-by: Martin Storsjö <martin@martin.st>
