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1805 commits

Author SHA1 Message Date
Wim Taymans
a87811f49a basertppayload: small comment fix 2009-10-16 10:59:39 +02:00
Peter Kjellerstedt
7bca2a0019 rtp: Correct timestamping of buffers when buffer_lists are used 
The timestamping of buffers when buffer_lists are used failed if
a buffer did not have both a timestamp and an offset.
 2009-10-16 10:51:22 +02:00
Stefan Kost
f1c32d0fbb build: fix previous commit to fully accomodate the glib-gen.mak changes 
I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
 2009-10-16 10:56:56 +03:00
Stefan Kost
a89c1de0ea build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114 
The build rules in glib-gen.mak were using pattern rules in a non save way.
 2009-10-16 10:23:09 +03:00
Tommi Myöhänen
02cbde648c baseaudiosrc: fix timestamp comparission, Fixes #597407 2009-10-13 19:17:49 +03:00
Patrick Radizi
48a44f470b rtsp: handle socket errors 
gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
on a socekt. Fix this problem by checking for error on 'other' socket after poll
return.

Fixes #596159
 2009-10-12 15:48:46 +02:00
Wim Taymans
5dbaccabca audioclock: whitespace fixes 2009-10-12 15:47:28 +02:00
Mark Nauwelaerts
e18b42c0b6 tag: use BOM to recognize UTF-16/32 encoding and convert accordingly 2009-10-09 16:22:54 +02:00
Josep Torra
ccec231d2b audio: fix warnings building on macosx 2009-10-09 14:09:02 +02:00
Tim-Philipp Müller
92465ba8ac rtspconnection: we can use GLib 2.18 API unconditionally now 2009-10-07 10:32:17 +01:00
Tim-Philipp Müller
a52483e59e docs: clarify GstTuner docs in two places 2009-10-07 10:15:52 +01:00
Benjamin Otte
a27f439ab3 Update Since tags for NV12/NV21 
They are added in 0.10.26 now, not 0.10.25
 2009-10-07 09:58:27 +02:00
Benjamin Otte
1cf651f883 Add NV12 and NV21 formats 2009-10-07 09:54:07 +02:00
Benjamin Otte
92928134ca [video] Fix Y41B 
Chroma components should be aligned on 4byte boundaries.

https://bugzilla.gnome.org/show_bug.cgi?id=595849
 2009-10-07 09:54:07 +02:00
Sebastian Dröge
6d40818ec0 streamvolume: Define cbrt() if it's not available 
Fixes build on Win32, bug #597537.
 2009-10-07 07:28:15 +02:00
Wim Taymans
730eead9a9 rtsp: use CLOSE_SOCKET() instead of close() 
Use CLOSE_SOCKET instead of directly calling close() because it does the right
thing for windows.

Fixes #597539
 2009-10-06 22:37:00 +02:00
Sebastian Dröge
901dbc6ab4 cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc 2009-09-17 17:00:10 +02:00
Jonathan Matthew
6781c4c9c5 cddabasesrc: ignore URI fragments that look like device paths 
Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
worked before the fix for bug #321532.

Also adds a check for negative track numbers and some unit tests for URI
parsing.

Fixes bug #595454.
 2009-09-17 17:00:10 +02:00
Michael Smith
1f43f87023 vorbistag: don't ever return NULL in list of strings. 2009-09-15 15:55:34 -07:00
Sebastian Dröge
df9b8b57b3 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH 
This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
 2009-09-13 11:19:50 +02:00
Sebastian Dröge
6e23ea172f interfaces: API: Add GstStreamVolume interface 
Fixes bug #567660.
 2009-09-11 16:37:34 +02:00
Wim Taymans
8d2f20d1cb rtsp: properly fix the HTTP manual mode 
When we're not parsing HTTP, return EPARSE when we get an HTTP
message.
 2009-09-11 12:20:10 +02:00
Tim-Philipp Müller
794e03640d mixertrack: add READONLY and WRITEONLY flags 
Should really have been READABLE and WRITABLE, but those are hard to
add whilst maintaining backwards compatibility. See #343615.

API: GST_MIXER_TRACK_READONLY
API: GST_MIXER_TRACK_WRITEONLY
 2009-09-11 10:20:27 +01:00
Tim-Philipp Müller
e4e8417eeb ringbuffer: fix build against core that has debugging disabled 
The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
 2009-09-11 10:03:56 +01:00
Sebastian Dröge
445311bff4 fft: Mark one function as const and add notes that the structs should be private in 0.11 2009-09-11 07:22:15 +02:00
Stefan Kost
312d7d8014 ringbuffer: add human readable format names when logging 
Add string array with human readable names for format and type to be used in log
statements.
 2009-09-10 23:01:36 +03:00
Wim Taymans
e2e7ae0129 basertppay: don't print RTP timestamps as clocktime 
Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.

Fixes #594757
 2009-09-10 18:21:08 +02:00
Wim Taymans
ca3b91b2d0 rtsp: don't return EPARSE 
Don't blindly return EPARSE when http mode is disabled.
Restore old http mode after temporarily setting it to TRUE.
 2009-09-10 14:04:53 +02:00
Wim Taymans
35cddfb1e3 baseaudiosink: add ugly backward compat hack 
Check for pulsesink < 0.10.17 because it includes code that is now included in
baseaudiosink. Disable that code in baseaudiosink to be compatible with the
older version.
 2009-09-10 12:40:01 +02:00
Wim Taymans
06be2b8632 baseaudiosink: take clock time in setcaps 
Take the time of the clock so that the last_time field is set. This is important
for sinks that restart their internal ringbuffer after a caps change and need to
know the last know position.
 2009-09-09 18:26:03 +02:00
Wim Taymans
451789735c audioclock: add some more debug 2009-09-09 18:26:03 +02:00
Wim Taymans
fe47c6c4d5 baseaudiosink: correct for clock reset 
When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.

Fixes #594136
 2009-09-09 16:19:32 +02:00
Wim Taymans
47550f6984 baseaudiosink: whitespace fixes 2009-09-09 16:17:02 +02:00
Wim Taymans
70f01fd797 ringbuffer: add more debug 2009-09-09 16:16:40 +02:00
Wim Taymans
42fad5a166 whitespace fixes 2009-09-09 10:25:33 +02:00
Tim-Philipp Müller
265e125993 videosink: add "show-preroll-frame" property 
Add a property to disable rendering of video frames during preroll. This
will only work for videosinks that use the new ::show_frame() vfunc instead
of overriding basesink's preroll and render vfuncs directly.

API: GstVideoSink:show-preroll-frame
 2009-09-08 18:20:22 +01:00
Tim-Philipp Müller
e2b4187fe3 video: add GstVideoSinkClass::show_frame() 
Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
vfuncs and add some gtk-doc chunks.

API: GstVideoSinkClass::show_frame()
 2009-09-08 18:20:02 +01:00
Tim-Philipp Müller
3bbbea6212 navigation: don't do stuff inside g_return_val_if_fail() statements 
Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
 2009-09-08 16:00:47 +01:00
Havard Graff
a14e730aad navigation: Fix compiler warning with MSVC 
Fixes bug #594275.
 2009-09-08 15:54:57 +02:00
Havard Graff
f710bec408 basertpdepayload: fix event forwarding 2009-09-08 15:10:59 +02:00
Havard Graff
f0f72088bc rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB 
Fixes #594258
 2009-09-08 13:03:21 +02:00
Håvard Graff
058776bcf1 baseaudiosrc: improve slave skew resync 
The old one did the mistake of not actually advancing the ringbuffer, it just
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
extra delay in the pipeline.

Also make sure that the resync can never go back in time, producing the same
timestamps that has already been produced, as this can cause severe problems
for sinks and other synching mechanisms.

Fixes #594256
 2009-09-08 12:59:20 +02:00
Sebastian Dröge
40aba9e0dc introduction: Fix out-of-tree build 2009-09-05 13:46:58 +02:00
Sebastian Dröge
ab17f5d3fa rtsp: Fix introspection build by ordering sources/headers in dependency order 2009-09-05 13:13:23 +02:00
Sebastian Dröge
c53499c62b audio: Remove debug echo 2009-09-05 13:09:17 +02:00
Sebastian Dröge
93e19acfec audio: Fix build of introspection data by using dependency order for the headers/sources 2009-09-05 13:08:19 +02:00
Sebastian Dröge
7e90e0846c introspection: Strip Gst prefix from all types/functions 2009-09-05 12:31:47 +02:00
Sebastian Dröge
7794caf9f8 introspection: Fix build if gir-repository is not installed 2009-09-05 11:49:41 +02:00
Sebastian Dröge
740bcd9479 video: Add gobject-introspection support 2009-09-05 11:37:14 +02:00
Sebastian Dröge
0c0ba97689 tag: Add gobject-introspection support 2009-09-05 11:35:34 +02:00
Sebastian Dröge
31b8e7fcee sdp: Add gobject-introspection support 2009-09-05 11:34:11 +02:00
Sebastian Dröge
d91f5000e1 libs: Add nodist headers and sources to the introspection files 2009-09-05 11:31:48 +02:00
Sebastian Dröge
e13a186b56 rtsp: Add gobject-introspection support 2009-09-05 11:28:59 +02:00
Sebastian Dröge
8001b380b1 rtp: Add gobject-introspection support 2009-09-05 11:25:42 +02:00
Sebastian Dröge
6ebc9414b6 riff: Add gobject-introspection support 2009-09-05 11:23:13 +02:00
Sebastian Dröge
9942cd57ef pbutils: Add gobject-introspection support 2009-09-05 11:20:51 +02:00
Sebastian Dröge
666bdf9dad netbuffer: Add gobject-introspection support 2009-09-05 11:17:07 +02:00
Sebastian Dröge
df2235beb5 interfaces: Add gobject-introspection support 2009-09-05 11:15:05 +02:00
Sebastian Dröge
b357cb9d2a fft: Add gobject-introspection support 2009-09-05 11:09:45 +02:00
Sebastian Dröge
a5f7c699ca cdda: Add gobject-introspection support 
This is disabled for now until gobject-introspection is fixed
 2009-09-05 11:09:39 +02:00
Sebastian Dröge
403f353bba audio: Add gobject-introspection support 2009-09-05 11:09:33 +02:00
Sebastian Dröge
61ae0059a4 app: Add gobject-introspection support 2009-09-05 11:09:28 +02:00
Wim Taymans
7a7663476f audiortppay: add some debugging 2009-09-03 18:53:19 +02:00
Wim Taymans
c1db9ebb20 audiortppay: handle gaps 
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
 2009-09-03 17:59:00 +02:00
Wim Taymans
3a3c6f309c audiortppay: fix frame duration calculations 
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
 2009-09-03 17:59:00 +02:00
Wim Taymans
bfc19462bb rtppay: add some debugging 2009-09-03 17:59:00 +02:00
Wim Taymans
bb91a7b47c audiortppay: use offsets for RTP timestamps 
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
 2009-09-03 17:58:59 +02:00
Wim Taymans
c1ae0a2003 audiortppay: move function around 2009-09-03 17:58:59 +02:00
Wim Taymans
5808041f44 audiortppay: fix sample duration calculation 2009-09-03 17:58:59 +02:00
Wim Taymans
299ab7be0e audiortppay: more refactoring 
Unify the sample/frame buffer handling code by making the functions plugable.
 2009-09-03 17:58:59 +02:00
Wim Taymans
fb5037f727 audiortppayload: refactor some more 
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
 2009-09-03 17:58:59 +02:00
Wim Taymans
1c6b71af03 audiortppayload: refactor and cleanup 
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.

API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
 2009-09-03 17:58:59 +02:00
Wim Taymans
50b9640d01 basertppay: add property to disable perfect RTP time 
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.

API: GstBaseRTPPayload::perfect-rtptime
 2009-09-03 11:29:23 +02:00
Wim Taymans
3a4edea56d basertppay: allow subclasses to influence RTP time 
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
 2009-09-03 11:15:20 +02:00
Wim Taymans
5a479669d4 basertppay: add macro to cast 2009-09-03 11:15:20 +02:00
Wim Taymans
bc3c8a1564 audiopayload: code cleanups 2009-09-03 11:15:20 +02:00
Wim Taymans
3c29efa692 audiortppayload: don't check adapter 
the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
 2009-09-03 11:15:20 +02:00
Jonas Holmberg
ec91d508af basertppayload: Make instance init faster by not reading /dev/urandom 3 times 
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.

Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.

Fixes bug #593284.
 2009-09-01 10:39:52 +02:00
Wim Taymans
008c760b6b cddabasesrc: safely handle the indexes 2009-08-28 19:06:57 +02:00
Wim Taymans
e40b262ab7 basertppayload: whitespace fixes. 2009-08-28 14:09:02 +02:00
Sebastian Dröge
72f3587f04 riff: Add support for AVF files 
AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.

Fixes bug #593117.
 2009-08-26 09:10:19 +02:00
Peter Kjellerstedt
8ce3612b71 rtsp: Mark Transport as supporting multiple values. 2009-08-24 14:39:16 +02:00
Peter Kjellerstedt
2882c22d95 rtsp: Added missing Since tags. 2009-08-24 13:58:50 +02:00
Eero Nurkkala
8ad8591e41 ringbuffer: Improve audiosink startup performance 
When we start the ringbuffer, immediatly continue processing samples if the
writer prepared some for us.

Fixes #545807
 2009-08-24 13:30:11 +02:00
Peter Kjellerstedt
066f9be5c9 rtsp: Added new API for sending using GstRTSPWatch. 
The new API to send messages using GstRTSPWatch will first try to send the
message immediately. Then, if that failed (or the message was not sent
fully), it will queue the remaining message for later delivery. This avoids
unnecessary context switches, and makes it possible to keep track of
whether the connection is blocked (the unblocking of the connection is
indicated by the reception of the message_sent signal).

This also deprecates the old API (gst_rtsp_watch_queue_data() and
gst_rtsp_watch_queue_message().)

API: gst_rtsp_watch_write_data()
API: gst_rtsp_watch_send_message()
 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
0af04aa4a8 rtsp: Made gst_rtsp_watch_queue_data() thread safe. 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
fb3b761af5 rtsp: Added gst_rtsp_connection_set_http_mode(). 
With gst_rtsp_connection_set_http_mode() it is possible to tell the
connection whether to allow HTTP messages to be supported. By enabling HTTP
support the automatic HTTP tunnel support will also be disabled.

API: gst_rtsp_connection_set_http_mode()
 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
d5b4b5d8af rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context. 
If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
then just setup the base64 decoding context for the first connection.
 2009-08-24 13:19:46 +02:00
Peter Kjellerstedt
01d98fdb5d rtsp: Write as much as possible in gst_rtsp_source_dispatch(). 
Try to write as much as possible if there are multiple messages queued.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
e5ec74c7a9 rtsp: Add error_full callback to GstRTSPWatchFuncs. 
The error_full callback is similar to the error callback, but allows for
better error handling. For read errors a partial message is provided to
help an RTSP server generate a more correct error response, and for write
errors the write queue id of the failed message is returned.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
ab8bea4555 rtsp: Made read_line() support LWS. 
Rewrote read_line() to support LWS (Line White Space), the method used by
RTSP (and HTTP) to break long lines. Also added support for \r and \n as
line endings (in addition to the official \r\n).
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
607209f121 rtsp: Do not split headers which should not be split. 
From RFC 2068 section 4.2: "Multiple message-header fields with the same
field-name may be present in a message if and only if the entire
field-value for that header field is defined as a comma-separated list
[i.e., #(values)]." This means that we should not split other headers which
may contain a comma, e.g., Range and Date.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
08d3fe8561 rtsp: Parse WWW-Authenticate headers correctly. 
Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
allows commas both to separate between multiple challenges, and within the
challenges themself, we need to take some extra care to split these headers
correctly.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
efc8901a39 rtsp: Improve parse_line(). 
Make parse_line() handle keys with multiple values on one line correctly.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
db66ff4a62 rtsp: Rewrote setup_tunneling(). 
Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
coded strings and duplicates of the message parsing code.
 2009-08-24 13:19:45 +02:00
Peter Kjellerstedt
c18e2eec88 rtsp: Rewrote gen_tunnel_reply(). 
Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
than a hard coded string.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
e1b3393d6b rtsp: Ignore the Content-Length for POST requests. 
The Content-Length for POST requests with an x-sessioncookie header should
be ignored as the length is bogus and only there to fool proxies.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
11c8b811f3 rtsp: Normalize lines (remove extra whitespace) before parsing. 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
5716cd102a rtsp: Made parse_string() return a result. 
This will catch parsing errors when a too long string is received.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
fdd5a65632 rtsp: Improved parsing of messages. 
Do not abort message parsing as soon as there is an error. Instead parse
as much as possible to allow a server to return as meaningful an error as
possible.
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
ca154010fe rtsp: Added support for HTTP messages 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
dd7d0cfc45 rtsp: Added gst_rtsp_connection_create_from_fd(). 
API: gst_rtsp_connection_create_from_fd()
 2009-08-24 13:19:44 +02:00
Peter Kjellerstedt
814eaa728a rtsp: Add initial buffer support. 
The initial buffer contains data for a connection which should be used
before starting to actually read anything from the socket.
 2009-08-24 13:19:44 +02:00
Wim Taymans
2c08c76383 appsink: don't block in paused 
When we are asked to unlock we should either leave the render function or call
the wait_preroll method to release the stream lock.

Fixes #592657
 2009-08-24 13:16:39 +02:00
Peter Kjellerstedt
41f1d9a7d9 rtsp: Add support for the Authentication-Info header. 
The Authentication-Info header is defined in RFC 2617 (Digest Access
Authentication).
 2009-08-24 11:24:27 +02:00
Peter Kjellerstedt
3c4fa9274f rtsp: Avoid duplicated headers. 
Remove any existing Session and Date headers before adding new ones
when sending a request. This may happen if the user of this code reuses
a request (rtspsrc does this when resending after authorization fails).
 2009-08-19 09:31:51 +02:00
Peter Kjellerstedt
3b888cfe2a rtsp: Corrected the HTTP digest authorization computation. 
Do not use sizeof() on an array passed as an argument to a function and
expect to get anything but the size of a pointer. As a result only the
first 4 (or 8) bytes of the response buffer were initialized to 0 in
auth_digest_compute_response() which caused it to return a string which
was not NUL-terminated...
 2009-08-18 16:50:58 +02:00
Mark Nauwelaerts
87e6775844 riff: align API doc of gst_riff_parse_chunk with reality 2009-08-12 13:39:14 +02:00
Tim-Philipp Müller
cb19626c8c rtspconnection: don't use GLib-2.18 function 
g_checksum_reset() was added only in GLib 2.18, but we still require
only 2.16, so work around that if we only have 2.16. Fixes #591357.
 2009-08-10 20:18:24 +01:00
Sebastian Dröge
79ade6ad68 rtsp: Use GLib's GChecksum instead of our own MD5 implementation 2009-08-10 10:19:01 +02:00
Mart Raudsepp
689a4d4c10 navigation: Fix doc blurb typo for gst_navigation_send_key_event 2009-08-09 20:52:40 -04:00
Tim-Philipp Müller
0021e6b765 Revert inlines that cause compiler warnings and are not needed anyway 2009-08-08 17:51:10 +01:00
Edward Hervey
9329b8be72 gst-libs: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:57 +02:00
Wim Taymans
090808a295 baseaudiosrc: change default slave method 
Set the default slave method to the much better skew slaving algortihm.
 2009-08-06 12:58:58 +02:00
John Millikin
cd31b2e298 tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags 
Require latest core for this.

Fixes bug #590430.
 2009-08-06 06:43:38 +02:00
Sebastian Dröge
713f6ca8d5 cddabasesrc: Allow to specify the device name in the URI 
The allowed URI scheme is now:
cdda://(device#)?track

Also allow every combination of uppercase and lowercase
characters for the protocol part.

Fixes bug #321532.
 2009-08-06 06:43:34 +02:00
Philip Jägenstedt
1b4220bd03 appsrc: Clarify documentation about caps and linkage 
Fixes bug #589095.
 2009-08-06 06:43:34 +02:00
Olivier Crête
429d3555a2 audiofilter: Don't assert on slightly different caps 
Plugins should not assert on incompatible caps, caps negotiation will
fail anyway.
 2009-07-30 14:34:05 +01:00
Olivier Crête
4e88633de4 audiosink: Add stream-status messages 
Fixes #587695
 2009-07-20 12:54:37 +02:00
Olivier Crête
cc0da016f8 audiosrc: Add stream-status messages 
See #587695
 2009-07-20 12:54:37 +02:00
Tim-Philipp Müller
d53e754d42 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest 
Differentiate subtitle streams and lyrics/cracktastic/complex streams via
the category string in the headers. This seems like a useful distinction
to make, and also seems more future-proof. See #525743.
 2009-07-13 23:00:04 +01:00
Stefan Kost
cae6a55ba3 navigation: simplify docs 
Make short-desc short - its used in the toc. Strip uneeded markup.
 2009-07-13 21:54:47 +03:00
Jan Schmidt
85de44aa01 navigation: Add some partial documentation 
Add a general documentation blurb for the GstNavigation functionality.
Still lacks some example code and detail on how to implement it.
 2009-07-13 17:55:55 +01:00
Tim-Philipp Müller
f6a508d963 pbutils: add description for Siren codec and make two descriptions non-translatable 2009-07-13 17:52:39 +01:00
Elliott Sales de Andrade
132fb5c050 riff: add siren to the RIFF parser 
Add siren7 caps to the RIFF parser.
 2009-07-13 18:22:55 +02:00
David Schleef
530cb7268b basevideo: send basevideo back to remedial school 
Move basevideo classes and schroedinger plugin to -bad.
 2009-07-01 10:27:30 -07:00
Wim Taymans
6c28c3f139 netaddress: add constant for max len 2009-07-01 12:54:21 +02:00
Wim Taymans
8ef62de3f0 netbuffer: add gst_netaddress_to_string 
Add function to serialize a net address to a string.

API: GstNetAddress::gst_netaddress_to_string()
 2009-07-01 12:48:38 +02:00
Stefan Kost
0e967f1b14 multichannel: rewrite the new doc comment a bit 
Its part of the audio lib.
 2009-06-29 17:49:58 +03:00
Wim Taymans
8601862e27 ringbuffer: add vmethod to clear the ringbuffer 
Add a vmethod so that subclasses can be notified when they should clear the data
in the ringbuffer.
 2009-06-29 15:17:25 +02:00
Jan Schmidt
a9097080a3 riff-media: Fix the fourcc caps property for VC-1/WMVA 
The caps property for carrying fourccs is 'format', not 'fourcc'
 2009-06-29 14:01:33 +01:00
Wim Taymans
f5962f0a4f rtsp: include in.h for FreeBSD compat 
Fixes #586920
 2009-06-29 12:20:52 +02:00
Wim Taymans
3928dbbb45 appsink: add docs and signals 
Add docs for the new callback.
Add signals for the new buffer-list support.
 2009-06-29 12:14:43 +02:00
Branko Subasic
6518d283d5 Added buffer list support. 2009-06-29 11:59:47 +02:00
Branko Subasic
fb0fd53212 Added buffer list support. 2009-06-29 11:59:46 +02:00
Peter Kjellerstedt
8927dbc98b sdp: Include winsock2.h after defining WINVER. 
Similar to bug #587080.
 2009-06-29 09:36:27 +02:00
Peter Kjellerstedt
c398f2f376 rtsp: Moved a comment. 2009-06-29 09:31:40 +02:00
Stefan Kost
57a7d6f699 docs: add basic section docs for multichannel and relocate the ones for audio 
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
 2009-06-27 23:25:09 +03:00
Руслан Ижбулатов
07c237ad19 Define WINVER before including any win headers 
Fixes bug #587080.
 2009-06-27 14:02:50 +02:00
René Stadler
41b7504e9c riff: prevent crash if rounded up tag size exceeds data size 
When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
and an invalid read past the buffer data follows.
 2009-06-27 01:22:52 +03:00
Sebastian Dröge
939baee2bd basevideocodec: By default don't allow caps changes on the srcpad 
This fixed playback of Dirac files with schrodec when upstream wants
a different width/height, basevideocodec accepts this and then
pushes buffers with new caps but content of the old caps.
In the best case this will just result in wrong unit size and a
failure in basestransform elements.
 2009-06-26 15:20:09 +02:00
Tim-Philipp Müller
adff66fc83 pbutils: add description for multipart 
So we get slightly nicer error messages when multipartdemux is missing.
 2009-06-24 09:51:11 +01:00
Wim Taymans
85af9b82e8 basertppayload: add support for bufferlists 
Based on patch from Ognyan Tonchev.

See #585559
 2009-06-19 15:52:34 +02:00
Wim Taymans
f5c8055edf rtpbuffer: use new convenience functions 
New core convenience functions makes the list getters and setters trivial.
Maybe even too trivial...
 2009-06-19 15:33:04 +02:00
Wim Taymans
457d39075c rtp: cleanups, add _list_get_seq() too 
Clean up the docs a little.
Add missing _list_get_seq method.
Add new symbols to the docs
 2009-06-18 19:04:52 +02:00
Wim Taymans
e2ccc1ee39 rtp: cleanups 
Add Since tags to docs
Move some code around
Add win32 symbols
 2009-06-18 18:51:04 +02:00
Wim Taymans
66c388a0e0 rtp: add bufferlist support 2009-06-18 18:51:04 +02:00
Wim Taymans
f385081c92 rtp: pass data to macros instead of GstBuffer 2009-06-18 18:50:35 +02:00
Peter Kjellerstedt
4fd61fbaa4 rtsp: Made the parsing of the RTSP URL scheme more generic. 2009-06-17 18:34:57 +02:00
Peter Kjellerstedt
726a47f777 rtsp: Added gst_rtsp_watch_queue_data(). 
gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
but allows for queuing any data block for writing (much like
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)

API: gst_rtsp_watch_queue_data()
 2009-06-17 18:34:33 +02:00
Peter Kjellerstedt
595f8b6d00 rtsp: Only extract the session ID from RTSP responses. 2009-06-17 18:02:18 +02:00
Peter Kjellerstedt
ddbeb44f14 rtsp: Added support for parsing IPv6 addresses in RTSP URLs. 2009-06-17 18:00:17 +02:00
Peter Kjellerstedt
95a606a0bb rtsp: Use getaddrinfo() to support both IPv4 and IPv6. 2009-06-17 17:59:47 +02:00
Peter Kjellerstedt
e1a4c8871a rtsp: Improved base64 decoding in fill_bytes(). 
The base64 decoding in fill_bytes() expected the size of the read data to
be evenly divisible by four (which is true for the base64 encoded data
itself). This did not, however, take whitespace (especially line breaks)
into account and would fail the decoding if any whitespace was present.
 2009-06-17 17:53:54 +02:00
Wim Taymans
ffd90dda89 audiosrc: fix get_offset 
When we need to jump to the most recently captured sample, jump to where the
next sample will be written instead of to some old data.

Fixes #581460
 2009-06-17 14:00:23 +02:00
Wim Taymans
57a13f28de audiosink: free the ringbuffer when going to NULL 
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
 2009-06-17 13:18:18 +02:00
Wim Taymans
e4492c24ea audio: correctly handle short read/writes 2009-06-17 13:17:30 +02:00
René Stadler
2c5f455423 baseaudiosrc: add some extra logging for buffer timestamps 2009-06-17 12:36:50 +02:00
Sebastian Dröge
a64caea0bd videofilter: Add a default get_unit_size function 
This returns the correct values for all formats that are handled by
GstVideoFormat and makes all the custom get_unit_size functions in
many elements unnecessary.
 2009-06-16 19:38:17 +02:00
Wim Taymans
33837d420c rtsp: add Timestamp header field 
fixes #585994
 2009-06-16 18:57:20 +02:00
Tim-Philipp Müller
70089160f8 audiosink, audiosrc: do the class_ref()s in the right class_init functions 
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
 2009-06-16 14:14:26 +01:00
Tim-Philipp Müller
3767cb6005 audiosink,audiosrc: ref the audio ring buffer class and type in class_init 
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
 2009-06-15 15:39:09 +01:00
Wim Taymans
a5491ba218 audiosrc: return FALSE when receiving a SEEK event 
When receiving a seek event, return FALSE as we don't implement seeking.
 2009-06-15 12:57:39 +02:00
Peter Kjellerstedt
73dd8236ce rtsp: Use a more consistent naming of GstRTSPRec variables. 2009-06-15 09:28:34 +02:00
Peter Kjellerstedt
ff38999c8b rtsp: Call message_sent() callback for all sent messages. 
Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
 2009-06-15 09:28:13 +02:00
Wim Taymans
a9c82f9472 ringbuffer: handle border cases in resampler 2009-06-11 19:13:28 +02:00
Wim Taymans
8bbf2e8a32 docs: fix typo 2009-06-11 12:39:19 +02:00
Wim Taymans
69b7fb3845 baseaudiosink: reset accum when dropping samples 
When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
 2009-06-11 12:38:35 +02:00
Jan Schmidt
c1bc55a4f5 docs: Fix a couple of warnings from the docs build. 2009-06-11 11:16:15 +01:00
Tim-Philipp Müller
249d9b4aa1 Don't include config.h multiple times when build audio testchannel app. 
Fixes build problem on win32 (#585075).
 2009-06-10 21:37:29 +01:00
Wim Taymans
e01fab3ace rtsp: add some more docs 2009-06-09 22:00:53 +02:00
Peter Kjellerstedt
263c5b227b rtsp: Avoid a compiler warning. 2009-06-09 18:24:55 +02:00
Peter Kjellerstedt
dfc57e3f8a rtsp: Updated documentation for GstRTSPResult. 
Moved GST_RTSP_ELAST to be last in the documentation to match the actual
enum values.
 2009-06-09 18:23:28 +02:00
Peter Kjellerstedt
9c40eeeb4c rtsp: Plug a memory leak. 
Free memory related to any partially read and/or written RTSP messages.
 2009-06-09 16:28:20 +02:00
Wim Taymans
38e59ec75d baseaudiosink: no need to cause discont when clipping 
Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
 2009-06-09 12:09:15 +02:00
Wim Taymans
cb4952fc2e audiosink: don't align when we clip 
Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
 2009-06-08 17:26:59 +02:00
Edward Hervey
ee3b251234 pbutils: Add description for hdv/aux-* formats. 2009-06-08 10:25:00 +02:00
Tim-Philipp Müller
5da78c8489 libgsttag: don't extract genres from empty ID3v1 tags 
If we don't have any other info, don't try to interpret the
genre field. In particular we don't want to interpret a genre
of 0 as 'Blues' if no other fields are set and the entire tag
is just empty.
 2009-06-06 12:04:12 +01:00
Peter Kjellerstedt
2dbd8702dd rtsp: Fixed a typo. 2009-06-05 14:06:17 +02:00
Peter Kjellerstedt
de18ad458f rtsp: Remove an unused variable. 2009-06-05 14:05:54 +02:00
Peter Kjellerstedt
b0a9848524 rtsp: Removed duplicate initialization of conn->writefd. 2009-06-05 13:59:14 +02:00
Peter Kjellerstedt
0167e3589d rtsp: Use #defined status codes. 2009-06-05 13:55:08 +02:00
Peter Kjellerstedt
c1a6644a18 rtsp: Correct gen_tunnel_reply(). 
Prevent gen_tunnel_reply() from generating an incomplete response
in case an error response code is given.
 2009-06-05 13:53:29 +02:00
Wim Taymans
59d9833924 rtsp: add G_LIKELY because we can 2009-06-02 12:10:39 +02:00
Peter Kjellerstedt
d8e0b5a4da rtsp: Avoid compiler warnings with -Wextra. 2009-06-01 09:59:22 +02:00
Peter Kjellerstedt
848b834cb9 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined. 2009-06-01 09:58:27 +02:00
Peter Kjellerstedt
e69c3a4f70 sdp: Remove an unused variable. 2009-06-01 09:43:04 +02:00
Wim Taymans
dcc42d5f92 netbuffer: also note the order of IP4 addresses 
IP4 addresses are also stored in network byte order. Make a note of this in the
docs.
 2009-05-27 11:08:37 +02:00
Tim-Philipp Müller
6292ff4ae0 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14" 
This reverts commit 418760cf74.

We now require GLib 2.16.
 2009-05-26 18:21:31 +01:00
Wim Taymans
796f8e2f76 netbuffer: document that the port is network order 
Document the fact that we store the port number in network order in
GstNetAddress and that the caller should byteswap appropriately.
 2009-05-26 15:39:18 +02:00
Andy Wingo
c7ca6abe53 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-05-26 13:17:44 +02:00
Bastien Nocera
9c508ba458 cddabasesrc: Remove copy of sha1 digest 
Remove our copy of sha1 digest now that we depend on glib 2.16.
Fixes #536313
 2009-05-26 11:11:03 +02:00
Tim-Philipp Müller
5fa9a8f4d0 video: don't expose internal gst_adapter_get_buffer() helper function 
If it's really needed it should go into GstAdapter in core.
 2009-05-25 00:19:25 +01:00
David Schleef
538c1cde31 basevideo: Fix memleak 2009-05-22 21:29:51 -07:00
David Schleef
35aae561e8 basevideo: Add preset interface to encoder 2009-05-22 17:34:56 -07:00
Wim Taymans
81170c4989 audiosink: improve debug message 2009-05-21 10:48:49 +02:00
Michael Smith
35a9de28f4 gstid3tag: Don't extract a track number unless present. 
In ID3v1, a track number is present only if byte 125 is null AND
byte 126 is non-null. If the track number is not present, don't add
a track number tag with value 0.
 2009-05-19 18:12:18 -07:00
Wim Taymans
243d366b34 videoutils: remove adapter methods 
Remove adapter methods now that they are in core.
 2009-05-20 00:48:40 +02:00
Wim Taymans
c68a361e31 audiosink: return the return value of wait_preroll 
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
 2009-05-19 17:17:37 +02:00
David Schleef
17f3810f7b video: remove // comments 2009-05-15 16:21:15 -07:00
David Schleef
45cf881f39 video: Add Y444, v210, v216 formats 2009-05-15 16:18:59 -07:00
David Schleef
4ec34e83d5 video: Copy BaseVideo classes from Schroedinger 2009-05-15 16:18:58 -07:00
Tim-Philipp Müller
f2031e1313 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000 2009-05-15 20:50:06 +01:00
Wim Taymans
b9723f6e1c audioclock: make our internal time monotonic 
Make the internal time increase monotonically.
 2009-05-13 21:38:56 +02:00
Sebastian Dröge
ab75db1653 propertyprobe: Fix typo in the docs 2009-05-12 15:53:07 +02:00
Wim Taymans
0a09632396 rtpdepay: add some more comments 2009-05-12 10:39:49 +02:00
Wim Taymans
d655120ee6 audioclock: make sure values are ever increasing 2009-05-12 10:39:41 +02:00
Sebastian Dröge
24dd91b1f0 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
e057414049 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
59aa1251d9 interfaces: API: Add gst_mixer_get_mixer_type() 
This is a convenience function that returns the mixer_type
of the interface struct.
 2009-05-12 09:03:24 +02:00
Sebastian Dröge
29b063b39b interfaces: Add docs for gst_color_balance_get_balance_type() 2009-05-12 09:03:24 +02:00
Sebastian Dröge
9fc4d195e1 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists 2009-05-12 09:03:22 +02:00
John Millikin
ef473dd0ae vorbistag: Store cover art in vorbiscomments 
Fixes bug #513373.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
e1875bf25f interfaces: API: Add gst_color_balance_get_balance_type() 
This is a convenience function that returns the balance_type
of the interface struct.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
b6c3567b41 interfaces: Separate struct definitions from typedefs 2009-05-12 09:03:22 +02:00
Tim-Philipp Müller
279b996d20 pbutils: add description for APE tag caps 2009-05-12 01:59:01 +01:00
Tim-Philipp Müller
3d33e2a873 tagdemux: cache events from upstream and re-send them once we have a source pad 
Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
Fixes #580318.
 2009-05-12 01:15:21 +01:00
Michael Smith
8f6399f109 riff: support UYVY raw 4:2:2 in riff. 2009-05-11 14:04:16 -07:00
Andy Wingo
9f74ce745f Revert "add can-activate-pull property to baseaudiosink" 
This reverts commit c4074a2ee4.
 2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c Revert "[baseaudiosink] add docs for can-activate-pull" 
This reverts commit 416ce16f26.
 2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26 [baseaudiosink] add docs for can-activate-pull 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
  can-activate-pull.
 2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-04-28 18:28:50 +02:00
Tim-Philipp Müller
8efe6108c4 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core 
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463.
 2009-04-19 18:15:28 +01:00
Tim-Philipp Müller
418760cf74 rtspconnection: don't use GLib-2.16 API, we require only 2.14 
Fixes #579267.
 2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f baseaudiosink: don't unparent the ringbuffer 
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
 2009-04-17 11:03:32 +02:00
Olivier Crete
d927114ef8 RTCP: don't fail when retrieving invalid PT 
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.

Fixes #579192.
 2009-04-17 10:53:10 +02:00
Wim Taymans
f83f57b648 app: add trivial cast macros 
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.

Fixes #579130
 2009-04-16 12:14:43 +02:00
Sebastian Dröge
a6cf0c8f06 video: Fix typo in the docs 2009-04-15 15:35:59 +02:00
Sebastian Dröge
a1d8cfde9d video: Add support for YVYU YUV colorspace 2009-04-15 14:53:47 +02:00
Tim-Philipp Müller
75acca2835 docs: fix hyperlink and move fft attribution to the right place 2009-04-15 00:19:19 +01:00
Stefan Kost
ab24d9d65c log: use G_GUINT64_FORMAT instead of llu 2009-04-15 00:02:39 +03:00
Josep Torra
71ab187355 RTSP: add missing headers for WMS RTSP 
Add missing headers related to Windows Media RTSP extension.
Fixes #578942
 2009-04-14 18:31:52 +02:00
Tim-Philipp Müller
9f23b82b2c Give credit to Mark Borgerding (kissfft author) 
and add myself to AUTHORS as well. Fixes #575638.
 2009-04-14 17:11:19 +01:00
Johann Prieur
86edcadc43 RTCP: add beginnings of Feedback messages 
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
 2009-04-14 16:45:20 +02:00
Wim Taymans
dffd1bcc97 baseaudiosrc: adjust the internal timestamp 
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
 2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9 baseaudiosink: use new clock time methods 
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.

When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
 2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823 audioclock: add methods for the internal offset 
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().

Add a debug category and some debug lines to the audio clock.

API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
 2009-04-14 13:08:52 +02:00
Wim Taymans
251f152c20 baseaudiosink: use the internal clock time 
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
 2009-04-10 21:50:55 +02:00
Martin Samuelsson
ee03bf5379 appsink: make callbacks return GstFlowReturn 
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827.
 2009-04-09 23:46:17 +02:00
Wim Taymans
e6798c5cce ringbuffer: allow for custom commit functions 
Allow subclasses to override the commit method.
 2009-04-09 18:04:44 +02:00
Wim Taymans
cae2981f83 baseaudiosink: fix a small glitch after pause 
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
 2009-04-08 18:06:54 +02:00
Stefan Kost
ff9ee1dc5a audiofilter: don't leak pad-template 
gst_element_class_add_pad_template() does not take ownership.
 2009-04-07 22:39:07 +03:00
Edward Hervey
2555eeb737 navigation/v4l: Don't use g_return_val_if_fail for computed/used values. 2009-04-04 16:28:14 +02:00
Wim Taymans
88110ea67e rtsp: use fully qualified urls when using a proxy 
Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
 2009-04-02 22:28:55 +02:00
Jan Schmidt
033e654172 navigation: Extend the navigation interface 
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
 2009-04-02 12:21:18 +01:00
Wim Taymans
eed784b372 rtsp: fix little typo in the comments 2009-04-01 09:03:35 +02:00
Tim-Philipp Müller
fc8c5cba15 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe 
People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
 2009-03-31 18:30:57 +01:00
Tim-Philipp Müller
dfe96ce618 rtsp: clear the entire builder structure 
And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
 2009-03-31 18:30:48 +01:00
Tim-Philipp Müller
dd9f077177 docs: fix typo in gst_rtsp_message_unset() API docs 2009-03-31 18:30:48 +01:00
Wim Taymans
8b37dc3eb8 rtsp: add support for proxies 
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
 2009-03-31 19:00:00 +02:00