Commit Graph

462 Commits

Author SHA1 Message Date
Martin Storsjö 6294d708b8 rtsp: Only set the ttl parameter if the server actually gave a value
Passing ttl=0 to the rtp/udp url contexts makes packets never
leave the host machine.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-09 15:04:32 +02:00
Tommy Winther 1a6b9a98ce rtsp: Handle requests from server to client
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.

Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-27 12:04:11 +02:00
Martin Storsjö c2ff63e3ac rtpenc: Move the trailing comma into FF_RTP_FLAG_OPTS
This simplifies adding more flags to the macro.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-23 15:27:42 +02:00
Martin Storsjö 298a587f44 rtp: Factorize the check for distinguishing RTCP packets from RTP
The binary doesn't change after this patch.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-16 17:45:33 +01:00
Martin Storsjö f3a094f2da sdp: Ignore RTCP packets when autodetecting RTP streams
The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-02-16 16:14:35 +01:00
Diego Biurrun a846202343 rtsp: Remove some unused variables from ff_rtsp_connect(). 2012-01-30 10:19:47 +01:00
Alex Converse 7181c4edee cosmetics: Remove extra newlines at EOF 2012-01-27 17:19:09 -08:00
Anton Khirnov adad5b88f8 lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft 2012-01-27 10:52:43 +01:00
Anton Khirnov 6e9651d106 lavf: remove AVFormatParameters from AVFormatContext.read_header signature 2012-01-27 10:51:57 +01:00
Dmitry Volyntsev 58f0978581 rtsp: Use a random offset for trying to open UDP ports for RTP
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.

This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-22 01:10:03 +02:00
Martin Storsjö dbb06b8c0d rtsp: Allow specifying the UDP port range via AVOptions
Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-22 01:10:02 +02:00
Dmitry Volyntsev bc495bad3d rtsp: Remove a leftover, currently pointless check
This check isn't relevant in the way the code currently works.

Also change a case of if (x == 0) into if (!x).

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-22 01:10:00 +02:00
Jean First 4be386b318 rtsp: Fix compiler warning for uninitialized variable
This one won't ever be used uninitialized in practice, but
the compiler doesn't realize it.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-01-04 22:15:42 +02:00
Anton Khirnov cd3716b9aa Replace all uses of av_close_input_file() with avformat_close_input(). 2011-12-12 20:34:38 +01:00
Anton Khirnov 3a7f7678eb lavf: deprecate av_close_input_stream().
And remove all its uses.
2011-12-12 20:21:47 +01:00
Martin Storsjö 30266038bd rtsp: Initialize the media_type_mask in the rtp guessing demuxer
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-12-02 11:52:47 +02:00
Anton Khirnov c3f9ebf743 lavf: make av_set_pts_info private.
It's supposed to be called only from (de)muxers.
2011-11-30 20:34:45 +01:00
Martin Storsjö 2583660664 rtpdec: Add an init function that can do custom codec context initialization
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-30 17:32:18 +02:00
Anton Khirnov ddffc2fdc3 avio: add support for passing options to protocols.
Not used anywhere yet, support for passing options from avio_open() will
follow.
2011-11-13 13:14:39 +01:00
Martin Storsjö 6f1b7b3944 avio: Add an AVIOInterruptCB parameter to ffurl_open/ffurl_alloc
Change all uses of these function to pass the relevant
callback on.
2011-11-13 13:12:17 +01:00
Martin Storsjö 9957cdbfd5 avformat: Use ff_check_interrupt 2011-11-13 13:08:13 +01:00
Martin Storsjö 6149485f6c http: Change the chunksize AVOption into chunked_post
The chunksize internal variable has two different uses - for
reading, it's the amount of data left of the current chunk
(or -1 if the server doesn't send data in chunked mode), where
it's only an internal state variable. For writing, it's used
to decide whether to enable chunked encoding (by default), by
using the value 0, or disable chunked encoding (value -1).

This, while consistent, doesn't make much sense to expose
as an AVOption. This splits the usage of the internal variable
into two variables, chunksize which is used for reading (as
before), and chunked_post which is the user-settable option,
with the values 0 and 1, where 1 is default.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-10 13:21:26 +02:00
Martin Storsjö 196bf28c5d rtsp: Set http custom headers via the AVOption
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-10 10:51:35 +02:00
Martin Storsjö 4b3dc857e4 rtsp: Discard the dynamic handler, if it has an alloc function which failed
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-07 11:23:56 +02:00
Reimar Döffinger bb3244dee2 Replace all usage of strcasecmp/strncasecmp
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.

Instead use our own implementations that always treat the data
as ASCII.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-06 11:52:57 +02:00
Martin Storsjö d450cc4f4a rtsp: Disable chunked http post through AVOptions
This avoids having to use a private function.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-05 16:53:58 +02:00
John Brooks f011fcd67e rtsp: add allowed_media_types option
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-11-02 21:37:46 +02:00
Anton Khirnov 84ad31ff18 lavf: replace av_new_stream->avformat_new_stream part II.
Manual replacements are done in this commit.

In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
2011-10-19 17:02:11 +02:00
Martin Storsjö 51369f2891 rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 22:02:31 +03:00
Martin Storsjö 3a6765fb5d rtsp: Make the rtsp flags avoptions set via a define
This helps sharing these options with the sdp and rtp demuxers.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 22:02:30 +03:00
Martin Storsjö 9867aea524 rtsp: Remove the separate filter_source variable
Read it as a flag from the flags field instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:49 +03:00
Martin Storsjö eca4850c6d rtsp: Accept options via private avoptions instead of URL options
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.

This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:48 +03:00
Martin Storsjö 2c9aa0247d rtsp: Simplify AVOption definitions
Use defines for shortening common parts, omit the .dbl named
initializer (since it's the first element in the union).

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:47 +03:00
Martin Storsjö 17fff881e7 rtsp: Merge the AVOption lists
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-17 19:57:45 +03:00
Martin Storsjö 76b0d03d82 rtsp: Request that dynamic rate is disabled
DSS enables this automatically if streaming VOD over TCP. If
enabled, the server feeds packets faster than realtime, screwing
up RTCP NTP based timestamps.

Also, DSS doesn't indicate that this was indicated, if it was
enabled automatically (although if it was requested to be enabled,
a header saying that it was enabled is added, but this isn't
added if it is enabled automatically), making it even harder
to detect and work around properly without explicitly asking
for it to be disabled(/enabled, if we were able to support it).

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:47 +03:00
Martin Storsjö 30eae32530 rtsp: Parse the x-Accept-Dynamic-Rate header
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:45 +03:00
Martin Storsjö bfc6db4477 rtpdec: Add ff_ prefix to all nonstatic symbols
Signed-off-by: Martin Storsjö <martin@martin.st>
2011-10-12 14:48:12 +03:00
Diego Biurrun 76e25dbca6 rtsp: remove disabled code 2011-07-18 18:22:02 +02:00
Anton Khirnov dfc2c4d900 lavf: use designated initialisers for all (de)muxers.
It's more readable and less prone to breakage.
2011-07-17 06:58:37 +02:00
Mans Rullgard 0ebcdf5cda Do not include mathematics.h in avutil.h
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-07-03 21:42:06 +01:00
Diego Biurrun f75e3da535 RTSP: Doxygen comment cleanup
Do not use Doxygen for comments that apply to specific implementation
details; merge some duplicated Doxygen comment blocks.
2011-07-03 22:33:22 +02:00
Martin Storsjö d840733937 rtsp: Don't pass string pointer as format string to ff_url_join
In this case, the string that was passed couldn't contain
user-defined data and thus there was no risk for injection
bugs, but it's safer this way, if we later change the
content of the options string.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-06-16 17:40:28 +03:00
Anton Khirnov d2d67e424f Remove all uses of now deprecated metadata functions. 2011-06-08 07:43:45 +02:00
Diego Biurrun f190f676bc Replace custom DEBUG preprocessor trickery by the standard one. 2011-06-03 00:44:06 +02:00
Ilya 4515f9b58a rtsp: use strtoul to parse rtptime and seq values.
strtol could return negative values, leading to various error messages,
mainly "non-monotonically increasing dts".

Signed-off-by: Anton Khirnov <anton@khirnov.net>
2011-05-24 19:11:28 +02:00
Martin Storsjö 0b4949b518 rtsp: Only do keepalive using GET_PARAMETER if the server supports it
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.

Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-05-11 10:42:34 +03:00
Martin Storsjö 9261e6cf3f rtp: Rename the open/close functions to alloc/free
This avoids clashes if we internally want to override the global
open function.

Signed-off-by: Martin Storsjö <martin@martin.st>
2011-04-24 00:05:37 +03:00
Stefano Sabatini 59d96941f0 avio: remove AVIO_* access symbols in favor of new AVIO_FLAG_* symbols
Make AVIO_FLAG_ access constants work as flags, and in particular fix
the behavior of functions (such as avio_check()) which expect them to
be flags rather than modes.

This breaks API.
2011-04-19 19:47:58 +02:00
Anton Khirnov f87b1b373a avio: AVIO_ prefixes for URL_ open flags. 2011-04-07 18:07:16 +02:00
Anton Khirnov 1869ea03b7 avio: make url_get_file_handle() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov e52a9145c8 avio: make url_close() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov 925e908bc7 avio: make url_write() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov dce3756459 avio: make url_read_complete() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov bc371aca46 avio: make url_read() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov 0589da0aa5 avio: make url_open() internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov 62eaaeacb5 avio: make url_connect internal. 2011-04-04 17:45:20 +02:00
Anton Khirnov 5652bb9471 avio: make url_alloc internal. 2011-04-04 17:45:19 +02:00
Anton Khirnov 6dc7d80de7 avio: avio_ prefix for url_close_dyn_buf 2011-04-03 22:47:05 +02:00
Martin Storsjö 895678f823 rtsp: Specify unicast for TCP interleaved streams, too
According to the RFC, the default is multicast if nothing is
specified, which doesn't make sense for TCP.

According to a bug report, some Axis camera models give a
"400 Bad Request" error if this is omitted.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-03-21 20:58:33 +01:00
Mans Rullgard 2912e87a6c Replace FFmpeg with Libav in licence headers
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-03-19 13:33:20 +00:00
Nicolas George c76374c6db Use AVERROR_EXIT with url_interrupt_cb.
Functions interrupted by url_interrupt_cb should not be restarted.
Therefore using AVERROR(EINTR) was wrong, as it did not allow to distinguish
when the underlying system call was interrupted and actually needed to be
restarted.

This fixes roundup issues 2657 and 2659 (ffplay not exiting for streamed
content).

Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-03-15 08:09:19 -04:00
Anton Khirnov 22a3212e32 avio: rename url_fopen/fclose -> avio_open/close.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 10:18:55 -05:00
Martin Storsjö 28c4741a66 libavformat: Remove FF_NETERRNO()
Map EAGAIN and EINTR from ff_neterrno to the normal AVERROR()
error codes. Provide fallback definitions of other errno.h network
errors, mapping them to the corresponding winsock errors.

This eases catching these error codes in common code, without having
to distinguish between FF_NETERRNO(EAGAIN) and AVERROR(EAGAIN).

This fixes roundup issue 2614, unbreaking blocking network IO on
windows.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-23 07:21:31 -05:00
Anton Khirnov b7effd4e83 avio: avio_ prefixes for get_* functions
In the name of consistency:
get_byte           -> avio_r8
get_<type>         -> avio_r<type>
get_buffer         -> avio_read

get_partial_buffer will be made private later

get_strz is left out becase I want to change it later to return
something useful.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-21 11:23:22 -05:00
Anton Khirnov e731b8d872 avio: move init_put_byte() to a new private header and rename it
init_put_byte should never be used outside of lavf, since
sizeof(AVIOContext) isn't part of public ABI.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:31 -05:00
Anton Khirnov ae628ec1fd avio: rename ByteIOContext to AVIOContext.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-20 08:37:15 -05:00
Anton Khirnov 9fcae9735e Replace remaining uses of parse_date with av_parse_time.
Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-16 23:39:57 +00:00
Martin Storsjö 2c35a6bde9 rtsp: udp_read_packet returning 0 doesn't mean success
If udp_read_packet returns 0, rtsp_st isn't set and we shouldn't
treat it as a successfully received packet (which is counted and
possibly triggers a RTCP receiver report).

This fixes issue 2612.
2011-02-17 00:37:00 +01:00
Martin Storsjö b2dd842d21 rtsp/rdt: Assign the RTSPStream index to AVStream->id
This is used for mapping AVStreams back to their corresponding
RTSPStream. Since d9c0510, the RTSPStream pointer isn't stored in
AVStream->priv_data any longer, breaking this mapping from AVStreams
to RTSPStreams.

Also, we don't need to clear the priv_data in rdt cleanup any longer,
since it isn't set to duplicate pointers.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-11 16:58:19 -05:00
Martin Storsjö b22dbb291d Use avformat_free_context for cleaning up muxers
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:39:55 -05:00
Martin Storsjö 1338dc0823 libavformat: Use avcodec_copy_context for chained muxers
This avoids having the chained AVStream->codec point to the same
AVCodecContext owned by the outer AVStream. The downside is that
changes to the AVCodecContext made after calling av_write_header
cannot be detected automatically within the chained muxer.

This avoids having to manually unlink the chained AVStream->codec
by setting it to null before freeing the chained muxer via generic
freeing functions.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
2011-02-04 11:28:07 -05:00
Martin Storsjö ce41c51b0c Free AVStream->info in chained muxers
This fixes memory leaks in the RTSP muxer and RTP hinting in the
mov muxer present since SVN rev 25418.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 01:03:31 +01:00
Martin Storsjö d9c0510e22 rtsp: Don't store RTSPStream in AVStream->priv_data
For mpegts in RTP, there isn't a direct mapping between RTSPStreams
and AVStreams, and the RTSPStream isn't ever stored in
AVStream->priv_data, which was earlier leaked. The fix for this
leak, in ea7f080749, lead to
double frees for other, normal RTP streams.

This patch avoids storing RTSPStreams in AVStream->priv_data, thus
avoiding the double free. The RTSPStreams are always available via
RTSPState->rtsp_streams anyway.

Tested with MS-RTSP, RealRTSP, DSS and mpegts/RTP.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2011-02-03 00:49:15 +01:00
Luca Barbato ea7f080749 Free the RTSPStreams in ff_rtsp_close_streams
This plugs a small memory leak

Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-02-01 20:40:16 +01:00
Luca Barbato dfd2a005eb Replace dprintf with av_dlog
dprintf clashes with POSIX.1-2008
2011-01-29 23:55:37 +01:00
Luca Barbato f81c7ac70a rtsp: make ff_sdp_parse return value forwarded
the sdp demuxer did not forward it at all while the rtsp demuxer assumed
a single kind of error
2011-01-28 15:45:19 +01:00
Luca Barbato a8475bbdb6 os: replace select with poll
Select has limitations on the fd values it could accept and silently
breaks when it is reached.
2011-01-28 15:45:19 +01:00
Diego Elio Pettenò c6610a216e Prefix all _demuxer, _muxer, _protocol from libavformat and libavdevice.
This also lists the objects from those two libraries as internal (by adding
the ff_ prefix) so that they can then be hidden via linker scripts.
2011-01-26 22:10:09 +00:00
Diego Elio Pettenò 57c4d01ec9 Make ff_rtsp_send_cmd_with_content_async static to rtsp.c.
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-25 22:10:36 +01:00
Martin Storsjö 2762a7a28b rtspdec: Retry with TCP if UDP failed
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:49:36 +01:00
Martin Storsjo aeb2de1c82 rtsp: Use ff_rtsp_undo_setup in the cleanup code in ff_rtsp_make_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:43 +01:00
Martin Storsjo 93e7490ee0 rtsp: Split out a function undoing the setup made by ff_rtsp_make_setup_request
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:39 +01:00
Martin Storsjo fef5649a82 rtsp: Make make_setup_request a nonstatic function
Signed-off-by: Janne Grunau <janne-ffmpeg@jannau.net>
2011-01-24 22:46:36 +01:00
Martin Storsjö a3b058b7ba rtsp: Properly fail if unable to open an input RTP port
Originally committed as revision 26285 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-09 10:47:53 +00:00
Martin Storsjö a92c30d76e rtsp: Allow requesting of filtering of source packets
If filtered, only packets from the right source address and port
are received.

To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.

If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.

Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-06 15:22:58 +00:00
Martin Storsjö 29db7c3af4 rtsp: Parse RTP-Info headers
Originally committed as revision 26236 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-05 21:23:42 +00:00
Martin Storsjö d2995eb910 rtsp: Store the Content-Base header value straight to the target
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.

Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:11:12 +00:00
Martin Storsjö 77223c5388 rtsp: Pass the method name to ff_rtsp_parse_line
Originally committed as revision 26191 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:10:12 +00:00
Martin Storsjö acc9ed1450 rtsp: Pass RTSPState to ff_rtsp_parse_line, instead of HTTPAuthState
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.

Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:07:56 +00:00
Martin Storsjö 3df54c6bf2 rtsp: Add a method parameter to ff_rtsp_read_reply
Originally committed as revision 26189 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-02 10:06:21 +00:00
Martin Storsjö 3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-01 22:27:16 +00:00
Martin Storsjö 9e99f84f7d rtsp: Check if the rtp stream actually has an RTPDemuxContext
For example MS-RTSP doesn't have RTPDemuxContexts for all streams.

This fixes issue 2448.

Originally committed as revision 26107 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-27 09:56:19 +00:00
Martin Storsjö 8c579c1c60 rtsp: Require the transport reply from the server to match the request
This fixes a crash if we requested TCP interleaved transport, but the
server replies with transport data for UDP. According to the RFC, the
server isn't allowed to respond with another transport type than the
one requested.

Originally committed as revision 26077 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-23 15:05:24 +00:00
Martin Storsjö bbd8f5477d rtsp: Don't set the RTP time base from the sample rate if no sample rate is set
This also reverts SVN rev 26016, which incorrectly overwrote the time base
with 90 kHz for all streams, regardless of what was set by the SDP parsing.

The stream that triggered the fix in 26016 still works after this commit.

Originally committed as revision 26022 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-15 21:06:25 +00:00
Martin Storsjö 86b6e387cc rtsp/rtpdec: Set the AVStream time_base directly in rtsp when it is known
This fixes cases where the RTP time base and the sample rate of the stream
differ. Previously, the AVStream time_base was unconditionally set to
the sample rate (which initially was set to one value when parsing the
rtpmap field in the SDP, but later overridden by an a=SampleRate field).

Additionally, this makes the code actually use the stream time base set
in rtpmap for video codecs, instead of hardcoding it to always be 90 kHz.

Originally committed as revision 25908 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:29:44 +00:00
Martin Storsjö bb776f3b00 rtsp: Parse RealRTSP sample rate declarations from the SDP
The RTP time base can be different from the actual content sample rate.

Originally committed as revision 25907 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-07 13:28:45 +00:00
Martin Storsjö 6a7e31a901 rtsp: Look for RTP payload handlers for static payload types, too
Originally committed as revision 25893 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:44 +00:00
Martin Storsjö 003eb64217 rtsp: Factorize code for initializing the rtp payload handler
Originally committed as revision 25892 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-05 19:41:09 +00:00
Martin Storsjö 0b6a7ff4b4 rtsp: Do a forgotten reindenting
Originally committed as revision 25839 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-28 21:17:39 +00:00
Martin Storsjö dd22cfb101 rtsp: Parse and use the Content-Base reply header, if present
This fixes playing RTSP urls with query parameters.

Originally committed as revision 25755 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-11-15 15:08:53 +00:00