Commit graph

380 commits

Author SHA1 Message Date
Jan Schmidt
4b84d7552f check: Don't fail the basetime test when no audiosrc is available
On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
it isn't in gst-plugins-base. Just defer the test, instead of
failing it.
2009-10-15 10:28:39 +01:00
Tommi Myöhänen
5e8e7c0358 tests: new test for baseaudiosrc base_time comparison
This test reveals a bug in comparison operation between timestamp and
GstElement's base_time in GstBaseAudioSrc.
2009-10-13 19:17:49 +03:00
Jan Schmidt
34480029fb check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty. 2009-10-09 15:11:52 +01:00
Benjamin Otte
4db9487a1f tests/check/libs/video.c: Update strides for Y41B 2009-10-07 11:49:18 +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
Jan Schmidt
a9f815bd8d check: Improve audioresample test
Make the audioresample test work with CK_FORK=no, and
turn a g_print into a GST_INFO.
2009-09-11 21:44:18 +01:00
Sebastian Dröge
723b2baa5d volume: Implement GstStreamVolume interface 2009-09-11 16:37:35 +02:00
Sebastian Dröge
e22c843d0e audioresample: Add unit test for checking for timestamp drifts
This also checks for perfect timestamping and offsetting.
2009-08-26 09:10:18 +02:00
Sebastian Dröge
01408497a1 audioresample: Improve debugging a bit in the unit test 2009-08-26 09:10:18 +02:00
Tim-Philipp Müller
099989ff0f oggmux: don't drop the streamheader field from the output caps
Revert previous 'fix' for bug #588717 and fix it properly, whilst
maintaining the streamheader field on the output caps. Also make
sure we don't leak header buffers we couldn't push when downstream
is unlinked. Add unit test for the presence of the streamheader
field on the output caps and for the issue from bug #588717.
2009-08-20 13:14:19 +01:00
Sebastian Dröge
11ad341d35 streamheader: Fix caps leak in the vorbisenc unit test 2009-08-10 15:40:33 +02:00
Tim-Philipp Müller
cc6e70e8ec checks: fix stream header unit test hanging in gst_task_cleanup_all()
Set pipelines to NULL state and unref when done.
2009-08-10 14:14:30 +01:00
Tim-Philipp Müller
e199d7e1cd typefinding: fix detection of fLaC id packet in broken flac-in-ogg
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
2009-08-01 19:01:39 +01:00
Edward Hervey
9819a3519d tests/adder: Add stream consistency checking. Fixes #588748 2009-07-20 11:30:07 +02:00
Jan Schmidt
de02af8d4f adder: One more attempt to fix the adder test
Give up and discard and recreate the alsasrc after checking it can
be opened, due to some strange crash inside alsa when we don't.
2009-07-14 15:31:13 +01:00
Jan Schmidt
7753d46350 adder: Perform get_state() in the unit test
Wait for the alsasrc to return to NULL after setting it to PAUSED for
testing, otherwise it leads to segfaults later on.
2009-07-14 15:06:41 +01:00
Jan Schmidt
b26eae25d0 adder: Don't fail when alsasrc is unavailable
Make the liveadder test succeed silently when it can't be completed
either because alsasrc is unavailable, or because the device is
inaccessible.
2009-07-14 14:39:32 +01:00
Stefan Kost
4736429c59 adder: skip live-seek text if we have no audiosrc, add new test
The seek-test needs a real audiosrc. Also add a test that checks that adder is
reusable. Finaly handle warnings as warnings to fix a assertion.
2009-07-10 19:01:25 +01:00
Sebastian Dröge
399d4fcbe7 gio: Try to reuse the pipeline with the same stream objects 2009-07-08 17:19:05 +02:00
Stefan Kost
92ecca7f24 adder: make test more robust
Add audioconverts to the live-seeking test to make it negotiate.
2009-07-06 20:44:00 +01:00
Branko Subasic
55a5679d89 Added unit tests for buffer list support in appsink. 2009-06-29 11:59:47 +02:00
Stefan Kost
6688af35eb adder: test seek handling in adder
This tests seeking on an adder that has a normal and a live source connected.
Wheter the current behavior is the desired one needs to be discussed still
(see #586033)
2009-06-22 22:18:03 +03:00
Wim Taymans
66c388a0e0 rtp: add bufferlist support 2009-06-18 18:51:04 +02:00
Tim-Philipp Müller
40bea96ff6 subparse: recognise more subrip timestamp variants
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes #585197.
2009-06-10 14:41:41 +01:00
Tim-Philipp Müller
a18128a3f6 tests: fix audioresample unit test on big endian architectures
Don't hardcode endianness=1234 in the filtercaps, it will cause
pad link failures which will result in the test timing out.
2009-05-12 23:51:08 +01:00
Jan Schmidt
e25f281de8 check: Disable the playbin2 for this release, as it is a bit racy.
Disable the test, as per the discussion in #580120. Needs re-enabling
after the release, when playbin2 is fixed.
2009-04-24 18:13:22 +01:00
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
Jan Schmidt
a8e3b4cacb check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes. 2009-04-16 00:41:42 +01:00
Jan Schmidt
2f01e624f5 check: Fix the input uri in playbin2 test.
Don't try and use a random file in wim's home directory as a test input
2009-04-16 00:41:42 +01:00
Wim Taymans
4f89685217 check: add a unit test for playbin2
Add unit test for playbin2 and include the refcount test in #577794.
2009-04-10 13:44:40 +02:00
Wim Taymans
4cdfc4b900 check: fix appsink test
Fix the appsink test now that the method signature changed.
2009-04-10 12:27:53 +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
Jan Schmidt
df660e91c2 ignores: Ignore the videoscale check binary 2009-04-02 12:18:07 +01:00
Tim-Philipp Müller
d271c8de53 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
2009-04-01 15:36:38 +01:00
Sebastian Dröge
5545a9704e videoscale: Add some more unit tests 2009-03-28 12:48:04 +01:00
Sebastian Dröge
8bb44e0f32 videoscale: Add a lot of unit tests 2009-03-28 10:25:12 +01:00
Wim Taymans
eb7b313369 tests: fix include in the appsink test
Fix dist by doing the right include.
2009-03-17 11:03:57 +01:00
Jan Schmidt
8285d7fdb0 check: Ignore alsamixer in the states test too 2009-03-13 15:58:34 +00:00
Wim Taymans
661f2da6e0 Appsink: add padding for callbacks + docs
Add some padding to the callbacks structure just to be safe.

Remove the now invisible marshaller methods from the docs.

Fix a comment in the unit test.
2009-02-26 11:42:44 +01:00
Sebastian Dröge
f14015567b Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref) 2009-02-22 19:20:40 +01:00
Edward Hervey
83fe624025 tests: Fix indentation 2009-02-22 13:43:35 +01:00
Wim Taymans
e5d8551552 Add method to install callbacks on appsink
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes #571299.

Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.

Add a unit test for appsink.

Clean up some of the appsink docs.

API: GstAppSink::gst_app_sink_set_callbacks()
2009-02-19 10:44:31 +01:00
Tim-Philipp Müller
95d6fb0501 pbutils: remove duplicate detail strings when calling the external codec installer
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
2009-02-02 17:34:23 +00:00
Sebastian Dröge
5dfcb63252 Rename files and types from speexresample to audioresample
Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
2009-01-23 12:33:41 +01:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00
Edward Hervey
c5ae184910 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
2008-12-12 19:41:28 +00:00
Wim Taymans
93e5a373ea tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
Pushing 10 buffers is enough to run the test.
2008-12-11 10:33:48 +00:00
Olivier Crete
3c9df39c15 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-08 12:08:32 +00:00
Sebastian Dröge
153406eef5 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-decodebin.xml:
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gnomevfs.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-queue2.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-uridecodebin.xml:
* docs/plugins/inspect/plugin-video4linux.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst/speexresample/gstspeexresample.c: (plugin_init):
* gst/speexresample/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(GST_START_TEST), (test_pipeline):
Rename the moved speexresample to audioresample, integrate into the
build system and remove the old audioresample from the build system.
Fixes bug #558124, #385061, #346218, #116051.
2008-11-27 16:57:09 +00:00