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for rtp"
This reverts commit b71685865fe761925feedda3cd0b288224d9a509.
The commit lead to the use of an uninitialized variable.
Other issues were listed by Andreas Rheinhardt:
https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/259150.html
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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This particular function is only required to return nonzero on
errors, but use the common AVERROR() pattern for consistency.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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* commit '1e56173515826aa4d680d3b216d80a3879ed1c68':
rtsp: add pkt_size option
Merged-by: James Almer <jamrial@gmail.com>
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This allows users to specify an upper limit on the size of outgoing packets
when publishing via RTSP.
Signed-off-by: Martin Storsjö <martin@martin.st>
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stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
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This reverts commit 1ae8a1073ba8a3ceb9f51d9350785c27530aa469.
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stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
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This reverts commit f502bd5432c9d7a34392ec3147bc5b5e3a868d9c.
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stimeout option is already used in tcp transport, since
http is based on tcp, pass the option to http for tunneling
case.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Signed-off-by: Jun Li <junli1026@gmail.com>
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Fix ticket #7297
The current setting for send-expect-100 option is either
enabled if applicable or forced enabled, no option to force
disable the header. This change is to expand the option setting
to provide more flexibility, which is useful for rstp case.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Add https based tunneling for RTSP/RTP. Tested on Axis and Bosch cameras.
Https is widely used for security consideration.
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Reviewed-by: Lauri Kasanen
Reviewed-by: Tomas Härdin
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Fixes: OOM
Found-by: Michael Hanselmann <public@hansmi.ch>
Reviewed-by: Michael Hanselmann <public@hansmi.ch>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes: Infinite loop
Found-by: Michael Hanselmann <public@hansmi.ch>
Reviewed-by: Michael Hanselmann <public@hansmi.ch>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Fixes two warnings:
libavformat/rtpdec.c:155:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
libavformat/rtpdec.c:168:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
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The "timeout" option name inherently clashes with the meaning of the
HTTP libavformat protocol option with the same name. Rename it after a
deprecation period to make it compatible with the HTTP one.
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Signed-off-by: Marton Balint <cus@passwd.hu>
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* commit 'f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5':
rtsp: only break on parse_rtsp_message on error
Merged-by: James Almer <jamrial@gmail.com>
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Fix suggested by Luca Barbato.
This was causing spurious EOFs when using -rtsp_transport udp, as
reported in https://bugzilla.libav.org/show_bug.cgi?id=1103
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Make easier to handle the polling function before we implement
full threading support.
(cherry picked from libav commit ca960161f087ca38267b88ce90592010c59584f1)
Signed-off-by: James Almer <jamrial@gmail.com>
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* commit '79331df362fb05a0d04ca9489c87e5b80077a3f4':
rtsp: Lazily set up the pollfd array once
Merged-by: James Almer <jamrial@gmail.com>
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* commit '5263f464db5f2df74ddf712f6d1221b24475fa8e':
rtsp: Lazily allocate the pollfd array
Merged-by: James Almer <jamrial@gmail.com>
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* commit 'b9b82151a1aaa8bbf389853a6142c4e101d80b86':
rtsp: Move the pollfd setup out of the for loop
Merged-by: James Almer <jamrial@gmail.com>
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* commit '150e99d694f33ab9ad678834964909aa315d14a1':
rtsp: Factor out packet reading
Merged-by: James Almer <jamrial@gmail.com>
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Main use-case is proxying avio through a foreign I/O layer and a custom
AVIO context, without losing latency and performance characteristics.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Merged from Libav commit 173b56218f39c64.
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Main use-case is proxying avio through a foreign I/O layer and a custom
AVIO context, without losing latency and performance characteristics.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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* commit '67deba8a416d818f3d95aef0aa916589090396e2':
Use avpriv_report_missing_feature() where appropriate
Merged-by: Clément Bœsch <cboesch@gopro.com>
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Make easier to handle the polling function before we implement
full threading support.
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And use av_malloc_array.
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This was introduced in bc2a32969e.
The whole block that the statement was added to is only
relevant when used as a demuxer, but the other statements
there have had other if statements guarding them. Make
sure to only run this whole block if being used as a
demuxer.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
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Introduced in 00e122bc / bc2a3296
The whole block that the statement was added to is only
relevant when used as a demuxer, but the other statements
there have had other if statements guarding them. Make
sure to only run this whole block if being used as a
demuxer.
Fixes ticket #5844.
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Also set a default_whitelist for mmsh and ffrtmphttp.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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* commit '3fdffc032e8ea5676bc0c2551b900c0dc887835b':
rtsp: Use avcodec_descriptor_get instead of avcodec_find_decoder
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
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* commit 'bc2a32969eb4db17677971def5ad5b936d9d1648':
rtsp: Parse SSRC attributes in the SDP
Merged-by: Clément Bœsch <u@pkh.me>
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This is only used for logging a human readable codec name for
debugging.
Signed-off-by: Martin Storsjö <martin@martin.st>
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When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
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* commit 'fab8156b2f30666adabe227b3d7712fd193873b1':
avio: Copy URLContext generic options into child URLContexts
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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* commit '9200514ad8717c63f82101dc394f4378854325bf':
lavf: replace AVStream.codec with AVStream.codecpar
This has been a HUGE effort from:
- Derek Buitenhuis <derek.buitenhuis@gmail.com>
- Hendrik Leppkes <h.leppkes@gmail.com>
- wm4 <nfxjfg@googlemail.com>
- Clément Bœsch <clement@stupeflix.com>
- James Almer <jamrial@gmail.com>
- Michael Niedermayer <michael@niedermayer.cc>
- Rostislav Pehlivanov <atomnuker@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
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We cannot play multiple multicast streams with the same port at the
same time. This is because both rtp and rtcp port are opened in
read-write mode, so they will not bind to the multicast address. Try
to make rtp port as read-only by default to solve this bug.
Signed-off-by: Zhao Zhili <wantlamy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
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Should make the default behaviour safer for careless callers that open
random untrusted files.
Bug-Id: CVE-2016-1897
Bug-Id: CVE-2016-1898
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This way, the decisions about which protocols are available for use in
any given situations can be delegated to the caller.
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