Sebastian Dröge
d5a4ca9962
build: Make some more rules silent if requested
2010-03-09 21:01:38 +00:00
Benjamin Otte
ed3e1ab8b2
gstvideo: Fix typos in comments
2010-03-09 19:17:04 +01:00
Wim Taymans
92a474b18c
basedepay: clarify some documentation
2010-03-08 12:11:01 +01:00
Dake Gu
f37b42b40d
rtspconnection: fix handling of x-server-ip-address
...
Fix handling of x-server-ip-address.
2010-03-08 11:20:51 +01:00
Stefan Kost
ef09538785
make: fix copy and paste error in git rules (audio<->video)
2010-02-22 13:04:42 +02:00
Patrick Radizi
a8f51d61f7
rtspconnection: make sure not to dereference NULL username or password
...
Fixes #610268 .
2010-02-18 18:00:38 +00:00
Stefan Kost
54094cd9ce
examples: add video overlay examples for gtk, qt and qt graphics view
...
Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
adds all boilerplate to configure for using c++. The qt based examples are
optional like their gtk counterparts.
2010-02-17 09:48:10 +02:00
Wim Taymans
76f715cb8b
appsrc: fix Since tag
2010-02-12 18:00:40 +01:00
Tim-Philipp Müller
5a2ae53bae
riff: treat JUNQ chunks like JUNK chunks
2010-02-12 14:24:22 +00:00
Sebastian Dröge
8d7304b12c
appsrc: Update basesrc segment duration and post duration messages from the streaming thread
2010-02-12 14:37:03 +01:00
Stefan Kost
d0f2b5a1cb
tags: improve docs about determining the encoding
2010-02-12 14:21:11 +02:00
Stefan Kost
b330e9aedc
comment: fix wrong header comment
2010-02-12 14:21:11 +02:00
Stefan Kost
9334069fd2
riff: add a variant of the JUNK tag that several adobe products produce
...
JUNQ has same semantics as JUNK.
2010-02-12 14:21:11 +02:00
Wim Taymans
c94356ad9b
appsrc: add min-percent property
...
Emit need-data when the amount of data in the internal queue drops below
min-percent.
Fixes #608309
2010-02-12 12:34:07 +01:00
Wim Taymans
fac9346405
appsrc: cleanups
...
Avoid some typechecks.
Avoid dereferencing appsrc->priv all the time.
2010-02-12 12:34:07 +01:00
Wim Taymans
7cce982ee2
appsink: cleanups
...
Avoid some typecasting.
Avoid dereferencing appsink->priv all the time.
2010-02-12 12:31:49 +01:00
Wim Taymans
30fd219e63
rtsp: ignore \n and \r as the first line
...
Be more forgiving for bad servers and ignore \r and \n when we are looking for
the response/request line.
See #608417
2010-02-12 11:43:59 +01:00
Wim Taymans
be037e0dc8
rtsp: fail gracefully on bad Content-Length headers
...
Be careful when allocating the amount of bytes specified in the Content-Length
because it can be an insanely huge value. Try to allocate the memory but fail
gracefully with a nice error when the allocation failed.
2010-02-12 11:43:59 +01:00
Sebastian Dröge
b5fd5953d1
appsrc: Update segment duration and post a duration message if the duration changes
...
Fixes bug #609423 .
2010-02-12 11:00:08 +01:00
Tim-Philipp Müller
e6d868c31c
audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
2010-01-27 00:42:37 +00:00
Edward Hervey
c783ec3c4d
pbutils: Add description for Zip Block Motion Video
2010-01-23 15:35:05 +01:00
Edward Hervey
dde84e4c4b
riff: Add mapping for Zip Block Motion Video
2010-01-23 15:34:54 +01:00
Edward Hervey
52ec4f4394
riff: YUNV is a fourcc which is also used for YUY2 raw video
2010-01-23 15:26:37 +01:00
Edward Hervey
de736fb1d1
riff: vp61 and VP61 are also valid On2 VP6 fourcc
2010-01-23 15:13:45 +01:00
Edward Hervey
5dff488a26
riff: Add mapping for On2 VP5
2010-01-23 15:10:45 +01:00
Edward Hervey
b5367b89ed
riff: Add mapping for Sigma-Designs MPEG4
...
It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.
2010-01-23 15:04:35 +01:00
Edward Hervey
d714a5a68b
pbutils: Add description for LOCO Lossless codec
2010-01-23 14:35:28 +01:00
Edward Hervey
554b4a6c25
riff: Add mapping for LOCO Lossless codec
2010-01-23 14:35:16 +01:00
Edward Hervey
444e7a68aa
riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
2010-01-23 14:08:39 +01:00
Edward Hervey
da3dd574c1
pbutils: add description for Autodesk Animator codec
2010-01-23 13:50:26 +01:00
Edward Hervey
2795591247
riff: Add mapping for Autodesk Animator Codec
2010-01-23 13:50:09 +01:00
Edward Hervey
ab82529497
pbutils: Add description for y4m container
2010-01-21 13:47:30 +01:00
Olivier Crête
6c6d0e32cf
basertppayload: ptime/maxptime should be unsigned
...
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-21 10:46:31 +01:00
Olivier Crête
8d2ac0b2ec
basertppayload: ptime should be in nanoseconds
...
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-21 10:46:17 +01:00
Olivier Crête
ad399c8069
basertppayload: Reject empty caps
...
https://bugzilla.gnome.org/show_bug.cgi?id=607353
2010-01-19 13:29:19 +01:00
Sebastian Dröge
6dfc0270ec
audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
...
Fixes bug #607381 .
2010-01-19 09:26:37 +01:00
Edward Hervey
24f1a9a9b7
pbutils: Add description for MXF container format
2010-01-18 17:57:16 +01:00
Benjamin Otte
0994a5bff3
rtsp: Don't define h_error ourselves
...
It's included from netdb.h and that header might define it differently,
which can lead to build failures.
2010-01-13 23:06:40 +01:00
Tim-Philipp Müller
5e6162eebb
docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
...
Explain why the whole bus sync handler mess is needed. Add section about
how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
break issue and how to work around it (see #601809 ).
2010-01-10 23:50:02 +00:00
Tim-Philipp Müller
22ff20a574
docs: minor netbuffer documentation fix
2010-01-10 21:18:04 +00:00
Tim-Philipp Müller
17e1d8d20a
tag: fix up disting of lang-tables.c more correctly
...
lang-tables.c is included by lang.c and not really a proper source
file that should be compiled into its own object, so rename it to
lang-tables.dat and put it into EXTRA_DIST instead to ensure it
gets disted.
2010-01-07 15:49:53 +00:00
Christian Schaller
658388c57b
Add missing source file for tagger to Makefile and update spec file
2010-01-07 13:50:03 +00:00
Mark Yen
140283c12b
riff-media: handle 32 bit raw RGB video.
2010-01-06 18:31:22 -08:00
Wim Taymans
73d5ae1107
audiopayload: add support for buffer-lists
2010-01-06 13:39:14 +01:00
Olivier Crête
bc6179952b
basertpaudiopayload: Respect ptime if it is given
...
If the ptime is given in the caps, respect it and force the minimum
and maximum sizes to be exactly the requested ptime.
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Olivier Crête
a4b0f2a1bd
rtpbasepayload: Store ptime from caps
...
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Olivier Crête
21151ba940
basertppayload: Accept maxptime from caps
...
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Wim Taymans
f7070b6bc6
rtcpbuffer: add helper functions for SDES types
...
Add functions to convert SDES names to their types and back. Will be used later
to set SDES items using a GstStructure.
See #595265
2009-12-22 20:15:28 +01:00
Tim-Philipp Müller
98fc463f31
docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
2009-12-21 07:57:42 +00:00
Tim-Philipp Müller
848a7f2868
baseaudiosink: increase default drift tolerance to fix glitches with WMA
...
Increase default drift tolerance to 40ms to avoid glitches with decoders
or formats where there's a lot of timestamp jitter for some reason or
another (in this case: asf/wma), at least until we implement timestamp
smoothing.
2009-12-20 23:19:41 +00:00
Tim-Philipp Müller
b529a33105
docs: mention that gst_tag_get_language_name() may return NULL
2009-12-13 18:43:56 +00:00
Tim-Philipp Müller
4cb197999e
docs: misc. mixer docs improvements
2009-12-12 18:58:39 +00:00
Tim-Philipp Müller
f71c4167e0
docs: add short descriptions for API reference contents page
2009-12-12 18:17:32 +00:00
Tim-Philipp Müller
25227e16b5
tag: make internal language names table static
2009-12-12 17:43:26 +00:00
Tim-Philipp Müller
3361d3286d
tag: don't use GLib 2.22 API
...
g_mapped_file_unref() was introduced in GLib 2.22, but we depend
only on GLib 2.18, so use g_mapped_file_free() when compiling
against older GLib versions until we bump the GLib dependency.
2009-12-12 17:41:44 +00:00
Tim-Philipp Müller
088c7c07a2
tag: add some utility functions for language codes and tags
...
Add some utility functions for language tags and ISO-639
codes. These are useful for both GUIs and elements. The
iso-codes package is used for language name translations
if available.
API: gst_tag_get_language_codes()
API: gst_tag_get_language_name()
API: gst_tag_get_language_code()
API: gst_tag_get_language_code_iso_639_1()
API: gst_tag_get_language_code_iso_639_2B()
API: gst_tag_get_language_code_iso_639_2T()
2009-12-12 15:48:37 +00:00
Sebastian Dröge
51e2cafe0e
audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
...
...and fix code style a bit.
2009-11-26 10:38:29 +01:00
Sebastian Dröge
3949cba47d
audiofilter: Add _CAST variants of the cast macros
2009-11-26 10:38:28 +01:00
Wim Taymans
75c5aed1ba
audiosink: add adjustement when slaving
...
Our calibration against the pipeline clock is done with the adjusted
ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
when reusing audio sinks after switching clocks and slaving methods in a
pipeline.
2009-11-25 10:26:16 -06:00
Stefan Kost
9e8db533a1
debug: fix format string that was missing a var
2009-11-21 17:47:26 +02:00
Wim Taymans
0e6b9e596d
baseaudiosink: fix initial calibration
...
When we are calibrating the internal clock against the external clock take into
account the time offset applied to our internal clock because we will subtract
that in the render_function again.
2009-11-18 17:11:03 +01:00
Mark Nauwelaerts
0fb680f680
baseaudiosrc: fix 'uninitialized' compiler warning
2009-11-18 12:37:44 +01:00
Jan Schmidt
36711ab477
video: Add functions to create/parse still frame events.
...
Add a new video event to mark the start or end of a still-frame
sequence, and a parser function to identify and extract info from
such events.
API: gst_video_event_new_still_frame()
API: gst_video_event_parse_still_frame()
Fixes : #601942
2009-11-18 00:10:57 +00:00
Sreerenj B
f3b3dd33f3
rtsp: avoid crashing on SIGPIPE
...
Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
Fixes #601772
2009-11-13 11:18:46 +01:00
Jan Schmidt
8c76ae5fa9
appsrc: Clear the EOS state on a seek.
...
Allow seeking back into the stream after it hits EOS.
2009-11-10 13:56:01 +00:00
Sebastian Dröge
27d4f9dca3
cddabasesrc: Never return a negative track number in get_uri()
2009-11-09 18:12:15 +01:00
Sebastian Dröge
acaeed6131
cddabasesrc: Don't set the track to 1 every time a device is set
...
Fixes bug #601104 .
2009-11-09 18:12:15 +01:00
Wim Taymans
4f3f9a1054
basesrc: fix startup position in the ringbuffer
...
When we start and we need to produce the first sample, go to the next sample
that will be written into the ringbuffer instead of trying to go to sample 0.
We relied on rather small ringbuffer sizes to correctly go to the current
sample, which breaks whith large buffers.
Fixes #600945
2009-11-06 12:22:00 +01:00
Wim Taymans
d8942e2850
baseaudiosink: make drift tolerance configurable
...
Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
drift or timestamp drift instead of relying on the latency-time value for clock
drift and 500ms for timestamp drift.
Remove warning about discont timestamp and simply resync. The warning is in some
cases not correct and is triggered more frequently now that we lower the
tolerance value.
2009-11-04 16:16:31 +01:00
Stefan Kost
f3db4e01b5
rtp: dump packets which we reject
2009-10-28 11:30:58 +02:00
Tim-Philipp Müller
6f4c1ac583
Remove GST_DEBUG_FUNCPTR where they're pointless
...
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
2009-10-28 00:59:35 +00:00
Olivier Crête
e27c24b200
rtpaudiopayload: Only sent exact multiple of the frame size
...
Also align the maximum size with the frame size, not only the minimum
2009-10-23 13:56:05 +03:00
Tim-Philipp Müller
65765dffbf
.gitignore: update after files got renamed
2009-10-17 21:11:10 +01:00
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