1
0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-27 01:42:20 +00:00
Commit Graph

747 Commits

Author SHA1 Message Date
Andreas Rheinhardt
b800327f4c avformat/avformat: Add FFInputFormat, hide internals of AVInputFormat
This commit does for AVInputFormat what commit
59c9dc82f4 did for AVOutputFormat:
It adds a new type FFInputFormat, moves all the internals
of AVInputFormat to it and adds a now reduced AVInputFormat
as first member.

This does not affect/improve extensibility of both public
or private fields for demuxers (it is still a mess due to lavd).

This is possible since 50f34172e0
(which removed the last usage of an internal field of AVInputFormat
in fftools).

(Hint: tools/probetest.c accesses the internals of FFInputFormat
as well, but given that it is a testing tool this is not considered
a problem.)

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-07 08:53:31 -03:00
Anton Khirnov
1e7d2007c3 all: use designated initializers for AVOption.unit
Makes it robust against adding fields before it, which will be useful in
following commits.

Majority of the patch generated by the following Coccinelle script:

@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ...  };

with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
2024-02-14 14:53:41 +01:00
Anton Khirnov
08bebeb1be Revert "all: Don't set AVClass.item_name to its default value"
Some callers assume that item_name is always set, so this may be
considered an API break.

This reverts commit 0c6203c97a.
2024-01-20 10:34:48 +01:00
Paul Orlyk
57648ee091
avformat/rtsp: Support mode field of Transport header being sent in upper case
Fixes server compatibility issues with rtspclientsink GStreamer plugin

Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2024-01-16 01:34:57 +01:00
Andreas Rheinhardt
0c6203c97a all: Don't set AVClass.item_name to its default value
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-12-22 15:12:33 +01:00
Michael Niedermayer
e4d5ac8d7d
avformat/rtsp: Use rtsp_st->stream_index
Fixes: out of array access
Fixes: rtpdec_h264.c149/poc

Found-by: Hardik Shah of Vehere
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2023-10-27 18:10:47 +02:00
Andreas Rheinhardt
7001ff74ba avformat/aviobuf: Add ffio_init_(read|write)_context()
Most users of ffio_init_context() simply want to wrap
a buffer into an AVIOContext; they do not provide
function pointers at all.

Therefore this commit adds shortcuts for these two common
operations. This also allows to accept const data when reading
(i.e. the const is now cast away at a central place in
ffio_init_read_context() instead of at several callers).
This also allows to constify the data in ff_text_init_buf().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-09-07 00:41:45 +02:00
Andreas Rheinhardt
2b41463b87 avformat/internal: Don't include avcodec.h
The general demuxing API uses parsers and decoders. Therefore
FFStream contains pointers to AVCodecContexts and
AVCodecParserContext and lavf/internal.h includes lavc/avcodec.h.

Yet actually only a few files files really use these; and it is best
when this number stays small. Therefore this commit uses opaque
structs in lavf/internal.h for these contexts and stops including
avcodec.h.
This also avoids including lavc/codec_desc.h implicitly. All other
headers are implicitly included as now (mostly through codec.h).

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-09-26 03:02:50 +02:00
Michael Niedermayer
73c0fd27c5 avformat/rtsp: break on unknown protocols
This function needs more cleanup and it lacks error handling

Fixes: use of uninitialized memory
Fixes: CID700776

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2022-07-12 21:55:22 +02:00
Yubo Xie
7795f045a0 libavformat/rtsp: pkt_size option is not honored in rtsp
Signed-off-by: xyb <xyb@xyb.name>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
2022-04-27 20:47:59 +08:00
Martin Storsjö
a78f136f3f configure: Use a separate config_components.h header for $ALL_COMPONENTS
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.

Signed-off-by: Martin Storsjö <martin@martin.st>
2022-03-16 14:12:49 +02:00
Martin Storsjö
4eb9232c6e libavformat: Split version.h
Signed-off-by: Martin Storsjö <martin@martin.st>
2022-03-16 14:05:26 +02:00
Vittorio Giovara
620d151e5c rtp: convert to new channel layout API
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2022-03-15 09:42:36 -03:00
Andreas Rheinhardt
aeb5d943c6 avformat/rtsp: #if unused functions away, fix -O0 compilation
parse_rtsp_message() is only called if the rtsp demuxer is enabled
and so it is normally compiled away if said demuxer is disabled.
Yet this does not happen when compiling with -O0 and this leads
to a linking failure because parse_rtsp_message() calls functions
that may not be available if the rtsp demuxer is disabled.
Fix this by properly #if'ing the unused functions away.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-01-08 18:13:04 +01:00
Limin Wang
f1c8c25832 avformat/rtsp: fix the error code from ffurl_read_complete()
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-07 20:48:07 +08:00
Limin Wang
6d42af02f5 avformat/rtsp: add error code handling for ff_rtsp_skip_packet()
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-07 20:33:17 +08:00
Limin Wang
f210766a55 avformat/rtsp: free the alloc memory if failed
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-07 20:33:17 +08:00
Limin Wang
130e4c6f4c avformat/rtsp: remove redundant assignment
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-07 20:33:17 +08:00
Limin Wang
98054e4f01 avformat/rtsp: load the sdp file with avio_read_to_bprint()
this allows getting rid of the hardcoded max size of SDP.

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-12-04 08:26:30 +08:00
Andreas Rheinhardt
a4798a5d51 all: Use av_memdup() where appropriate
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-12-03 16:07:02 +01:00
Limin Wang
7bf4c06809 avformat/rtp: add localaddr for network interface selection
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2021-11-27 11:21:17 +08:00
Andreas Rheinhardt
6c8f96f18b avformat/rtsp: Use av_dict_set_int()
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-19 04:40:13 +02:00
Andreas Rheinhardt
40bdd8cc05 avformat: Avoid allocation for AVStreamInternal
Do this by allocating AVStream together with the data that is
currently in AVStreamInternal; or rather: Put AVStream at the
beginning of a new structure called FFStream (which encompasses
more than just the internal fields and is a proper context in its own
right, hence the name) and remove AVStreamInternal altogether.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-17 13:22:25 +02:00
Andreas Rheinhardt
45bfe8b838 avformat/avio: Move internal AVIOContext fields to avio_internal.h
Currently AVIOContext's private fields are all over AVIOContext.
This commit moves them into a new structure in avio_internal.h instead.
Said structure contains the public AVIOContext as its first element
in order to avoid having to allocate a separate AVIOContextInternal
which is costly for those use cases where one just wants to access
an already existing buffer via the AVIOContext-API.
For these cases ffio_init_context() can't fail and always returned zero,
which was typically not checked. Therefore it has been made to not
return anything.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-08-25 23:01:54 +02:00
Hayden Myers
9b4b0df470 libavformat/rtsp.c: Reply to GET_PARAMETER requests
Some encoders send GET_PARAMETER requests as a keep-alive mechanism.
If the client doesn't reply with an OK message, the encoder will close
the session.  This was encountered with the impath i5110 encoder, when
the RTSP Keep-Alive checkbox is enabled under streaming settings.
Alternatively one may set the X-No-Keepalive: 1 header, but this is more
of a workaround.  It's better practice to respond to an encoder's
keep-alive request, than disable the mechanism which may be manufacturer
specific.

Signed-off-by: Hayden Myers <hmyers@skylinenet.net>
Signed-off-by: Martin Storsjö <martin@martin.st>
2021-07-12 16:00:48 +03:00
Andriy Gelman
6df4bcf1d0 avformat/rtsp: Include rtcp in port range check
Currently it is only checked that the rtp port does not exceed rtp_port_max.

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2021-07-05 13:37:39 -04:00
Andriy Gelman
d1f78067a6 avformat/rtsp: Reindent after previous commit
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2021-07-05 13:37:26 -04:00
Andriy Gelman
02387de90e avformat/rtsp: Set port_off to zero for low min/max port range
Fixes:
$ ffmpeg -min_port 32000 -max_port 32001 -i rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov -f null -
[1]    303871 floating point exception (core dumped)

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2021-07-05 13:36:22 -04:00
Andriy Gelman
8257d6dda6 avformat/rtsp: Fix timeout option
92c40ef882 added a listen_timeout option
for sdp. This allowed a user to set variable timeout which was
originally hard coded to 10 seconds.

The commit used the initial_timeout variable to store the value. But
this variable is shared with rtsp where it's used to infer a "listen"
mode. Thus, the timeout value could not be set in rtsp, and the default
value (initial_timeout = -1) would give 100ms timeout.

This was attempted to be fixed in c8101aabee,
which changed the meaning of initial_timeout = -1 to be an infinite
timeout. However, it did not address the issue that the timeout could
still not be set. Being able to set the timeout is useful because it
allows to automatically reconfigure from a udp to tcp connection in the
lower transport.

In this commit this is fixed by using the stimeout variable to
store the timeout value.

Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2021-07-05 12:49:55 -04:00
James Almer
b9c5fdf602 avformat: move AVStream.{parser,need_parsing} to AVStreamInternal
Those are private fields, no reason to have them exposed in a public
header.

Signed-off-by: James Almer <jamrial@gmail.com>
2021-05-07 09:27:21 -03:00
Andreas Rheinhardt
bc70684e74 avformat: Constify all muxer/demuxers
This is possible now that the next-API is gone.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 11:48:06 -03:00
Andreas Rheinhardt
6f34f03190 avformat/rtsp: Remove deprecated old options, rename stimeout->timeout
Deprecated in ff46124b0d.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:09 -03:00
Jiangjie Gao
3e9284fccb avformat/rtsp: support buffer_size and pkt_size options for RTP
And forward it to the underlying UDP protocol.

Fixes ticket .

Signed-off-by: Jiangjie Gao <gaojiangjie@live.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
2021-03-19 23:13:26 +01:00
Andreas Rheinhardt
e9513052b5 avformat/rtsp: Fix build failure when RTP demuxers are disabled
rtsp.c uses a check of the form "if (CONFIG_RTSP_DEMUXER && ...) {}"
with the intent to make the code compilable even though the part guarded
by this check contains calls to functions that don't exist when the RTSP
demuxer is disabled. Yet even then compilers still need a declaration of
all the functions in the dead code block and error out if not (due to
our usage of -Werror=implicit-function-declaration) and no such
declaration exists for a static function in rtsp.c. Simply adding a
declaration leads to a "used but never defined" warning, therefore this
commit resorts to an #if.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2021-02-03 21:36:47 +01:00
tpol
71ce5c32f0 avformat/rtsp: correctly set media control uri with mpegts
Fixes 

Currently the media control uri is not correctly assigned when mpegts is
signalled in the media description.

The code checks whether at least one AVStream has been setup before
assigning to the media's uri. With mpegts the AVStreams are setup when
parsing packets and so the media's uri is skipped. This is fixed by
using rt->nb_rtsp_streams in the check which counts all medias in the
sdp.

Reviewed-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2021-01-17 21:36:16 -05:00
Aman Karmani
d20f059fb9 avformat/rtsp: add satip_raw flag to receive raw mpegts stream
This can be used to receive the raw mpegts stream from a SAT>IP
server, by letting avformat handle the RTSP/RTP/UDP negotiation
and setup, but then simply passing the MP2T stream through
instead of demuxing it further.

For example, this command would demux/remux the mpegts stream:

    SATIP_URL='satip://192.168.1.99:554/?src=1&freq=12188&pol=h&ro=0.35&msys=dvbs&mtype=qpsk&plts=off&sr=27500&fec=34&pids=0,17,18,167,136,47,71'
    ffmpeg -i $SATIP_URL -map 0 -c copy -f mpegts -y remux.ts

Whereas this command will simply write out the raw stream, with
the original PAT/PMT/PIDs intact:

    ffmpeg -rtsp_flags satip_raw -i $SATIP_URL -map 0 -c copy -f data -y raw.ts

Signed-off-by: Aman Karmani <aman@tmm1.net>
2020-12-28 14:08:44 -08:00
Aman Karmani
98b76bb11f avformat/rtsp: add support for satip://
The SAT>IP protocol[1] is similar to RTSP. However SAT>IP servers
are assumed to speak only MP2T, so DESCRIBE is not used in the same
way. When no streams are active, DESCRIBE will return 404 according
to the spec (see section 3.5.7). When streams are active, DESCRIBE
will return a list of all current streams along with information
about their signal strengths.

Previously, attemping to use ffmpeg with a rtsp:// url that points
to a SAT>IP server would work with some devices, but fail due to 404
response on others. Further, if the SAT>IP server was already
streaming, ffmpeg would incorrectly consume the DESCRIBE SDP response
and join an existing tuner instead of requesting a new session with
the URL provided by the user. These issues have been noted by many
users across the internet[2][3][4].

This commit adds proper spec-compliant support for SAT>IP, including:

- support for the satip:// psuedo-protocol[5]
- avoiding the use of DESCRIBE
- parsing and consuming the com.ses.streamID response header
- using "Transport: RTP/AVP;unicast" because the optional "/UDP"
  suffix confuses some servers

This patch has been validated against multiple SAT>IP vendor devices:

- Telestar Digibit R2
  (https://telestar.de/en/produkt/digibit-r1-2/)
- Kathrein EXIP 418
  (https://www.kathrein-ds.com/en/produkte/sat-zf-verteiltechnik/sat-ip/227/exip-418)
- Kathrein EXIP 4124
  (https://www.kathrein-ds.com/en/products/sat-if-signal-distribution/sat-ip/226/exip-4124)
- Megasat MEG-8000
  (https://www.megasat.tv/produkt/sat-ip-server-3/)
- Megasat Twin
  (https://www.megasat.tv/en/produkt/sat-ip-server-twin/)
- Triax TSS 400
  (https://www.conrad.com/p/triax-tss-400-mkii-sat-ip-server-595256)

[1] https://www.satip.info/sites/satip/files/resource/satip_specification_version_1_2_2.pdf
[2] https://stackoverflow.com/questions/61194344/does-ffmpeg-violate-the-satip-specification-describe-syntax
[3] https://github.com/kodi-pvr/pvr.iptvsimple/issues/196
[4] https://forum.kodi.tv/showthread.php?tid=359072&pid=2995884#pid2995884
[5] https://www.satip.info/resources/channel-lists/
2020-12-28 14:08:44 -08:00
Andriy Gelman
207658112b avformat/rtsp: set AV_OPT_FLAG_DEPRECATED on deprecated options
Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2020-12-18 16:14:17 -05:00
Limin Wang
95d12da559 avformat/rtsp: prefer to use MAX_URL_SIZE for url and command buffer
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-12-05 09:00:53 +08:00
Andriy Gelman
4fe9e2fc16 avformat/rtsp: don't forget to call ff_network_close() on error
In sdp_read_header() some ff_network_close() calls were missed.

Also in rtp_read_header() update comment to explain why a single
call to ff_network_close() is enough to cover all cases even if
sdp_read_header() returns an error.

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2020-11-21 10:15:52 -05:00
Andriy Gelman
78537aa52f avformat/rtsp: set return variable in error path
In this error path ret still stores the number of bytes read in
ffurl_read().

Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2020-11-21 10:15:46 -05:00
Limin Wang
5bb313e723 avformat/rtsp: av_rescale -> av_rescale_q
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-14 09:09:28 +08:00
Limin Wang
aa1fab6934 avformat/rtsp: check return value of ffurl_read_complete
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-11 18:32:56 +08:00
Limin Wang
001ccbc5cc avformat/rtsp: prefer to use variable instead of type
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-11 18:32:56 +08:00
Limin Wang
33f6bb7828 avformat/rtsp: move SDP_MAX_SIZE macro definition to header file
move comments for the size of SDP_MAX_SIZE here:
Some SDP lines, particularly for Realmedia or ASF RTSP streams,
contain long SDP lines containing complete ASF Headers (several
kB) or arrays of MDPR (RM stream descriptor) headers plus
"rulebooks" describing their properties. Therefore, the SDP line
buffer is large.
The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
in rtpdec_xiph.c.

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-11 18:32:56 +08:00
Limin Wang
5439460016 avformat/rtsp: 16384 -> SDP_MAX_SIZE
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-11 18:32:56 +08:00
Limin Wang
c8101aabee avformat/rtsp: support infinite initial_timeout for rtsp option
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-11-07 10:13:29 +08:00
Limin Wang
784ce1c294 avformat/rtsp: reuse POLLING_TIME and remove POLL_TIMEOUT_MS
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-10-22 20:53:56 +08:00
Limin Wang
92c40ef882 avformat/rtsp: support for listen_timeout option for sdp
Now the listen timeout is hardcoded(10s).
How to test(30s timeout):
./ffprobe  -listen_timeout 30 -protocol_whitelist rtp,udp,file -i test.sdp

Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
2020-10-22 20:53:56 +08:00
Andriy Gelman
0d156eb58a avformat/rtsp: allocate correct max number of pollfds
There is one general rtsp connection plus two connections per stream (rtp/rtcp).

Reviewed-by: Zhao Zhili <zhilizhao@tencent.com>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
2020-10-08 23:18:18 -04:00