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.
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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.
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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.
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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
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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
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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
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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().
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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
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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"
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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
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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
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* 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
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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
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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.
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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
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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
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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
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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()
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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
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Fixes bug #513373 .
2009-05-12 09:03:22 +02:00
Sebastian Dröge
e1875bf25f
interfaces: API: Add gst_color_balance_get_balance_type()
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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
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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"
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This reverts commit c4074a2ee4
.
2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c
Revert "[baseaudiosink] add docs for can-activate-pull"
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This reverts commit 416ce16f26
.
2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26
[baseaudiosink] add docs for can-activate-pull
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* 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
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* 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
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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
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Fixes #579267 .
2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f
baseaudiosink: don't unparent the ringbuffer
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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
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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
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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
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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)
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
2009-03-31 19:00:00 +02:00
Wim Taymans
8be68f983c
Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
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This reverts commit 79de0b8d67
.
2009-03-31 18:57:08 +02:00
Stefan Kost
79de0b8d67
rtsp: reset whole message (was sizeof pointer instead of sizeof type)
2009-03-31 12:27:09 +03:00
Jan Schmidt
43788e4796
doc: Fix a typo in the GstMixer docs
2009-03-31 00:58:24 +01:00
Wim Taymans
0d3d3026d2
rtsp: start CSeq counting from 1 instead of 0
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Start counting from 1 instead of 0 as this is what most other clients
seem to do.
2009-03-25 16:37:28 +01:00
Wim Taymans
1081ae59eb
rtsp: add ETag and If-Match headers
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Add new headers, we need them for RealMedia support.
2009-03-25 16:36:14 +01:00
Tim-Philipp Müller
0267e79778
audiosrc: improve 'Dropped n samples' warning message
2009-03-25 11:27:44 +00:00
Sebastian Dröge
108ead73c8
rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
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This also fixes another instance of CVE-2008-4316.
2009-03-17 22:53:44 +01:00
Wim Taymans
f4b7cbbf16
rtsp: fix resolving of hostnames
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We were returning a pointer to a stack variable with the resolved hostname,
which doesn't work.
return a copy of the resolved ip address instead.
Fixes #575256 .
2009-03-13 16:19:41 +01:00
Wim Taymans
91b2d71da0
appsrc: release lock in _eos flushing case
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Release the mutex when we are flushing in gst_app_src_end_of_stream()
Fixes #574964 .
2009-03-13 15:16:44 +01:00
Jan Schmidt
566583e871
vorbistag: Protect memory allocation calculation from overflow.
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Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
2009-03-12 15:02:07 +00:00
Wim Taymans
0e2157029e
rtsp: fix parsing of the timeout parameter
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--
2009-03-11 18:45:59 +01:00
Wim Taymans
b674584e97
rtsp: fix g_return condition
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when parsing a data message, we require a data message.
2009-03-11 17:29:41 +01:00
Wim Taymans
18f612ffa9
rtsp: free the right string.
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Free the key value before we remove the header item from the array. The item we
retrieved from the array is only valid until we remove it from the array.
2009-03-11 14:09:54 +01:00
Wim Taymans
16225d45be
rtsp: keep track of amount of decoded bytes
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Keep track of the actual amount of decoded bytes, which can be less than 3 when
we decode the last bits of a base64 message.
2009-03-11 14:09:54 +01:00
Wim Taymans
f964c0fc38
rtsp: only add ports when not using TCP
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Only add the port numbers in the transport string when we are using udp or
multicast.
2009-03-09 13:53:41 +01:00
Wim Taymans
bc54a5f9a0
rtsp: use gstreamer dump mem
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--
2009-03-09 13:53:15 +01:00
Wim Taymans
3a72044a22
rtsp: use glib base64 encoder
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--
2009-03-09 13:51:48 +01:00
Edward Hervey
a3c88fb32b
Riff: Add mapping for Fraps video codec.
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Found through insanity testrun. Confirmed mapping in libavformat.
2009-03-09 10:03:13 +01:00