Commit graph

276 commits

Author SHA1 Message Date
Tim-Philipp Müller
d236f9185f .gitignore: ignore new appsrc unit test 2010-09-24 12:19:43 +01:00
Thiago Santos
24b523e129 appsrc: Do not override buffer caps if appsrc caps is null
Make appsrc not set caps on buffers when its own caps is NULL.

This avoids calling make_metadata_writable on all buffers and
prevents losing buffer caps in case we are not replacing it
with something meaningful.

https://bugzilla.gnome.org/show_bug.cgi?id=630353
2010-09-23 08:23:47 -03:00
Thijs Vermeir
2a868b70ec tests: use gst-check API in videotestsrc
use gst_check_drop_buffers in videotestsrc to
clear the global buffers list.
2010-09-16 15:07:15 +02:00
Thijs Vermeir
36e8018ca7 tests: Fix unit test of videotestsrc
Use UYVY for unit tests, it's exactly the same as Y422.
(which is currently disabled in videotestsrc)
2010-09-16 15:00:22 +02:00
Thijs Vermeir
78055cc852 tests: videotestsrc change the pattern property for the tests 2010-09-10 10:43:51 +02:00
Sebastian Dröge
acc7d51ce5 videoscale: Add some debug output to the videoscale negotiation test 2010-08-11 18:10:45 +02:00
Tim-Philipp Müller
14649643df checks: simplify GstBus usage in videoscale unit test
There's no need to run a main loop, add a bus watch and deal with
helper structs here just to wait for an EOS message.
2010-06-26 12:03:54 +01:00
Tim-Philipp Müller
3563c7c7a5 checks: speed up videoscale unit test a little
Use new gst_element_link_pads_full() function to link elements,
and disable all checks when linking (don't try this at home).

Down to 18s from 3m20s. Scary.
2010-06-26 11:45:34 +01:00
Stefan Kost
aef6123172 tests: use our own macros for the tests main function 2010-06-21 09:53:23 +03:00
Edward Hervey
d17d8e5319 Revert "audioresample: set pads as negotiable"
This reverts commit 5f74f3a82e.
2010-06-15 13:09:37 +02:00
Edward Hervey
be72e1506f Revert "audioconvert: set pads negotiable"
This reverts commit bbd7dee8f6.
2010-06-15 13:09:29 +02:00
Wim Taymans
bbd7dee8f6 audioconvert: set pads negotiable 2010-06-14 18:10:52 +02:00
Wim Taymans
5f74f3a82e audioresample: set pads as negotiable 2010-06-14 17:48:58 +02:00
Sebastian Dröge
39c39ff5ab videoscale: And only expect a single buffer in the unit test 2010-06-14 16:07:21 +02:00
Sebastian Dröge
b62f435fd0 videoscale: Only convert one buffer instead of five
Should make the unit test a lot faster.
2010-06-14 16:02:13 +02:00
Sebastian Dröge
8ae112058c videoscale: Add ffmpegcolorspace after videotestsrc for the unit test 2010-06-13 22:17:18 +02:00
Sebastian Dröge
303566654e videoscale: Add a unit test for checking if the negotiation works as expected 2010-05-13 11:16:02 +02:00
Benjamin Otte
3bd4aa26ff Add -Wwrite-strings to configure
Fixes for the code included
2010-03-16 17:41:51 +01:00
Benjamin Otte
5e21fa5e0e gst_element_class_set_details => gst_element_class_set_details_simple
Also change my email from the old university one to the current one.
2010-03-16 17:41:50 +01:00
Tim-Philipp Müller
444762a877 tests: fix videorate test
Fix up videorate test for latest videotestsrc changes: just check for
the important bits in the negotiated caps, not for exact equality with
our filter caps. Also don't leak the videorate element in the test.
2010-03-16 01:08:48 +00:00
Stefan Kost
412e5f5871 tests: update videorate test for videotestsrc changes
Add color-matrix to the caps we are comparing. Add logging og the caps in the
test.
2010-03-15 11:43:23 +02:00
Benjamin Otte
43b1683421 Add -Wmissing-declarations -Wmissing-prototypes to warning flags
Includes all the fixes necessary to make stuff compile again.
2010-03-11 13:50:31 +01:00
Benjamin Otte
9b5ef584e2 check: Ref buffers after setting caps on them
Reffing makes metadata unwritable, so we need to set the caps before.
2010-03-11 12:18:50 +01:00
Tim-Philipp Müller
c681ecab9f tests: fix typo in videorate unit test pipeline description
Two consecutive ! ! leave a 'Link without source' error in the debug log.
2010-03-10 00:59:10 +00:00
Tim-Philipp Müller
f7ca667189 tests: don't use deprecated functions in videorate unit test 2010-03-10 00:59:10 +00:00
Thiago Santos
e5f96a7a19 videorate: tests: New unit tests for upstream caps nego
Adds unit tests that check videorate's upstream caps
negotiation works properly (put passthrough caps
first)

Fixes #608025
2010-02-22 17:27:52 -03:00
Sebastian Dröge
40a841e377 playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2 2010-02-15 10:23:13 +01:00
Sebastian Dröge
b326b77ffa playbin2: Enable all unit tests
They're all working and valgrind clean now.
2010-02-15 08:28:24 +01:00
Sebastian Dröge
ad9830f547 tests: Add decodebin2 test to .gitignore 2010-02-15 06:50:29 +01:00
Sebastian Dröge
93d7bd2c43 decodebin2: Add simple unit test, mainly a copy of the decodebin unit test
The only difference between the two unit tests right now is,
that the decodebin2 test resets the element to READY before trying
to reuse it instead of NULL. decodebin2 guarantees to be reusable
without going back to NULL.
2010-02-15 01:21:14 +01:00
Sebastian Dröge
52a34b65b5 playbin: Fix indention in the unit test 2010-02-14 22:16:31 +01:00
Wim Taymans
59ace1b9ee adder: use collectpads clipping function
Install a clipping function in the collectpads and use the audio clipping helper
function to perform clipping to the segment boundaries.

Fixes #590265
2009-12-24 16:30:23 +01:00
Stefan Kost
3d73a7458a adder: make events succeed, if they succed on atleast one pad 2009-11-19 21:28:23 +02:00
Iago Toral
f63643bd54 subparse: Add support for DKS subtitle format
Fixes bug #598936.
2009-10-22 10:02:11 +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
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
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
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
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
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
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
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
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
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
Sebastian Dröge
ecf6fe6455 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.
2008-11-25 16:37:50 +00:00
Jon Trowbridge
0bdeaae59e gst/volume/gstvolume.*: Cleanup volume, define and use default values.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_update_volume), (gst_volume_set_volume),
(gst_volume_get_volume), (gst_volume_set_mute),
(gst_volume_class_init), (gst_volume_init),
(volume_process_double), (volume_process_float),
(volume_process_int32), (volume_process_int32_clamp),
(volume_process_int24), (volume_process_int24_clamp),
(volume_process_int16), (volume_process_int16_clamp),
(volume_process_int8), (volume_process_int8_clamp), (volume_setup),
(volume_transform_ip), (volume_set_property),
(volume_get_property):
* gst/volume/gstvolume.h:
Cleanup volume, define and use default values.
Recalculate new volume and mute setup before processing. Fixes #561789.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add controller unit test. Patch by: Jonathan Matthew
Fix bogus test that messed with basetransform's internal state.
2008-11-24 12:03:11 +00:00
Sebastian Dröge
f31ea1e221 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
2008-11-22 15:02:15 +00:00
Sebastian Dröge
b2cbf8f91d tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
2008-10-30 14:46:31 +00:00
Sebastian Dröge
d80b5c4aae Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/arch.h:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(_gcd), (gst_speex_resample_transform_size),
(gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_query), (gst_speex_resample_set_property):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_wrapper.h:
* tests/check/elements/speexresample.c: (setup_speexresample),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance):
Add support for double samples as input and refactor the usage
of the different compilation flavors of the speex resampler.
2008-10-30 12:43:44 +00:00
Sebastian Dröge
f5b4fa17ff gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
2008-10-29 12:11:20 +00:00
Sebastian Dröge
60bf63486b Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
2008-10-10 17:13:40 +00:00
Sebastian Dröge
b735321f58 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
2008-10-10 15:32:10 +00:00
Stefan Kost
1875564b65 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
2008-09-03 10:12:04 +00:00
Edward Hervey
162cb885c6 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (setup_audioresample),
(fail_unless_perfect_stream), (test_perfect_stream_instance),
(test_discont_stream_instance):
Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Add debugging for coherence.
2008-09-01 16:05:45 +00:00
Sebastian Dröge
59a0c5373d tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Enable unit tests on PPC again as the bugs are now fixed.
2008-06-30 09:46:15 +00:00
Sebastian Dröge
fdd708c418 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
2008-05-30 08:42:17 +00:00
Sebastian Dröge
ca7a0b8e9e tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).
2008-05-29 19:37:47 +00:00
Sebastian Dröge
45ef6b5e13 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously	conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 11:34:09 +00:00
Tim-Philipp Müller
206f91995b Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
2008-05-25 20:51:35 +00:00
Wim Taymans
c6b54c3d02 Don't use bad gst_element_get_pad().
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/decodetest.c: (new_decoded_pad_cb):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
(cleanup_decodebin):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(connect_element), (gst_decode_group_control_demuxer_pad):
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
(mute_group_type):
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
(gst_play_bin_set_property), (handoff), (gen_video_element),
(gen_text_element), (gen_audio_element), (gen_vis_element),
(remove_sinks), (add_sink), (setup_sinks):
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
(gst_play_sink_request_pad):
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
(cb_newpad):
* gst/playback/test6.c: (new_decoded_pad_cb):
* tests/check/elements/audioconvert.c: (GST_START_TEST):
* tests/check/elements/audiorate.c: (test_injector_chain),
(do_perfect_stream_test):
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
* tests/check/elements/gnomevfssink.c:
* tests/check/elements/textoverlay.c:
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
* tests/check/pipelines/oggmux.c: (test_pipeline):
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
* tests/examples/seek/seek.c: (make_mod_pipeline),
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline),
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
(update_fill), (msg_buffering):
Don't use bad gst_element_get_pad().
2008-05-21 16:36:50 +00:00
Sebastian Dröge
74d46a9977 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.
2008-05-21 07:46:02 +00:00
Sebastian Dröge
d03bbd1e3e gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:39:56 +00:00
Sjoerd Simons
fd84ec0ca3 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
2008-05-13 10:59:49 +00:00
Sjoerd Simons
09163ca363 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Tim-Philipp Müller
fd54092a2a gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller  <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
2008-05-06 12:12:16 +00:00
Sebastian Dröge
9333eb4899 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
2008-05-05 12:33:05 +00:00
Tim-Philipp Müller
1157de776a tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
2008-05-03 16:00:04 +00:00
Tim-Philipp Müller
84c8a873f9 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c:
Make test compile without unused function/variable warnings on PPC.
2008-03-23 13:12:41 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
Tomasz Sałaciński
6ab3a0e0c0 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
2008-02-23 09:51:26 +00:00
Jan Schmidt
f0ca338866 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
2008-01-12 23:24:02 +00:00
Tim-Philipp Müller
5850d56021 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_primary_decoder):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
(cddabasesrc_suite):
Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
deprecated in the future (see #498924).
2007-12-21 22:46:56 +00:00
Robin Stocker
7bbbf15ad8 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes #502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
2007-12-08 18:38:39 +00:00
Thijs Vermeir
c0a5c0655a tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
2007-11-02 14:59:06 +00:00
Tim-Philipp Müller
b0cf257c79 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
2007-10-31 17:32:22 +00:00
Tim-Philipp Müller
998c07972c tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily.  Looks like a problem with the 'jess'
visualisation.
2007-09-12 17:35:52 +00:00
Tim-Philipp Müller
ffdd6b3665 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
2007-09-12 17:15:12 +00:00
Sebastian Dröge
6fa7788c5d Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
2007-09-09 04:08:48 +00:00
Sebastian Dröge
6d7debb0bb gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
2007-09-05 21:20:12 +00:00
Michael Smith
1b7a0df57e gst/audiorate/gstaudiorate.c: Debug output fixes.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Debug output fixes.
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
(GST_START_TEST):
Change the number of buffers used; 500 is too many and leads to
timeouts.
2007-08-10 13:55:44 +00:00
Michael Smith
9f9e76bc99 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If we have a large (> 1 second) discontinuity, push a series of
smaller buffers rather than a single very large buffer. Avoids
unreasonably large single buffer allocations when encountering a
large gap.
* tests/check/elements/audiorate.c: (GST_START_TEST),
(audiorate_suite):
Add a test for this.
2007-08-09 15:44:02 +00:00
Tim-Philipp Müller
79b4be47a8 Fix 'make check' build against core CVS.
Original commit message from CVS:
* configure.ac:
* tests/check/elements/volume.c: (GST_START_TEST):
Fix 'make check' build against core CVS.
2007-07-11 23:12:12 +00:00
Sebastian Dröge
dbb857b93b gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_dithering_get_type),
(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
(gst_audio_convert_init), (gst_audio_convert_set_caps),
(gst_audio_convert_set_property), (gst_audio_convert_get_property):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_noise_shaping),
(gst_audio_quantize_free_noise_shaping),
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither),
(gst_audio_quantize_setup_quantize_func),
(gst_audio_quantize_setup), (gst_audio_quantize_free):
* gst/audioconvert/gstaudioquantize.h:
Implement dithering and noise shaping in audioconvert. By default now
TPDF dithering (and no noise shaping) will be used when converting
from a higher bit depth to 20 bit depth or smaller, otherwise
everything will be as it is now.
For the last audioconvert in a pipeline it would make sense to
use some kind of noise shaping, enabling it by default for all
conversions would give undesired results though. Fixes #360246.
* tests/check/elements/audioconvert.c: (setup_audioconvert),
(GST_START_TEST):
Adjust unit test for the new audioconvert.
2007-06-28 20:37:58 +00:00
Tim-Philipp Müller
087f644cfe ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
Skip empty buffers, but not empty header buffers. That way the original
vorbisdec unit test still passes (#451145); also, take into account
that those empty packets might carry a granulepos.
* tests/check/Makefile.am:
* tests/check/elements/vorbisdec.c:
(_create_codebook_header_buffer), (_create_audio_buffer),
(GST_START_TEST), (vorbisdec_suite):
Add unit test that sends an empty packet.
2007-06-27 10:14:03 +00:00