Commit graph

1627 commits

Author SHA1 Message Date
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