Fix some racyness as the test was checking the idle property
right after it got the preview message for video recordings. In some
conditions, it might happen that camerabin2 still hasn't decremented
the processing counter after posting the preview and/or the image/video-done
message and the test will get idle=false and fail.
The approach for checking for the idle property was a busy loop with
a sleep. Far from elegant, but good enough for these tests.
Adding some additional checks for custom camerabin2 messages
Also improves reliability of the tests by waiting for the preview-buffer message.
This message is generated from the preview pipeline, and might arrive later than the
image/video-done from the main camerabin2 pipeline, the tests need to wait for it even
after getting the image/video-done.
Allow context callbacks to perform custom buffer verification
and custom element setup. Also move some static tracking variables
into helper struct and expose the latter struct for use by unit test
(callbacks).
Use resource warning messages to notify camerabin2 that a capture
as aborted or couldn't be started, making it decrement the
processing counter and making the idle property more reliable.
Adds another test that checks that the idle property works
correctly when bogus start-capture calls are made.
This fails currently, but should remind us of fixing it in
the future by defining a proper error reporting from camera
sources to camerabin2
A camerabin2 tests depends on the commit
dcbba0932d for jpegenc on -good
that was released on 0.10.27.
This patch makes it check for this version before running the
test.
Increases timeout value so that tests won't fail with full
debug output.
Also removes buffer alloc from videotestsrc as this feature
isn't ready on camerabin2
Fixes capture location creation to use multifilesink location
like strings. E.g. using capture_%d instead of capture_0 to let
camerabin2 handle the %d replacing
Adds 'idle', a read-only boolean property that tells applications
if there is any capturing/saving/encoding going on in camerabin. If
not, it is safe to set it to NULL and release resources without
losing data.
Add "ready-for-capture" property to indicate if preparing a new
capture is possible.
"ready-for-capture" changes before the 'image-done' signal, so
the application can be notified that it can do a new capture
even before the previous one has finished encoding/saving.
Garbage after frames is now included in the frames because flacparse
has no easy way to detect the real end of a frame. Decoders are
expected to everything after the frame because only decoding the
bitstream will reveal the real end of the frame.
Fixes bug #631814.
The tests were done in 2 steps, first there was a suite
that generated the files (while checking that camerabin
was operating correctly). Then there was a second suite
that was run to check that all files were playable with
playbin2. Those second tests were not being run because
they were checking if camerabin was initialized, and it
never was as those tests didn't use a 'setup' function.
This commit refactors the tests by removing this second
suite and merging its validation with the first suite's
functions.
This commit is out of sequence, it really belongs between commits
1509833141 (Automatic update of
common submodule) and dd26f378b0
(configure: Fix the CELT checks to ...)
Adds block-after-capture property to block running viewfinder after capturing.
This property is useful if application wants to display capture preview and avoid
running viewfinder on background.
Based on a patch by Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
Adds a new property called viewfinder-filter to camerabin.
This property is used to add a filter to process the video
flow right before the viewfinder sink.
Also updates test to check property exists.
It's an external which lives in gstcheck.c. Redeclaring it makes some
compilers/architectures think the 'buffers' in the individual tests are
a different symbol... and therefore we end up comparing holodecks with
oranges.
Fixes build of faac due to functions not being static nor
having being declared in headers. (No previous prototype error)
Probably due to added -Wmissing-prototypes
Only put primary language into GST_TAG_LANGUAGE, and convert to lower case,
ie. only use "en" of "en_GB". This is per our tag documentation and hence
what apps expect. Also add example to kateenc property description so people
know a language code is wanted here.
katedec: Kate decoder (text only)
kateenc: Kate encoder (text and DVD SPU only)
katetag: Kate tagger
kateparse: Kate parser
tiger: Kate renderer using the Tiger rendering library
Fixes#525743.
Connect to the pad-removed signal of the ptdemux elements so that we remove the
ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
the sinkpads.
Fixes#561752
Sprinkle more logging to make it easier to follow. Specify a low framerate and
capture resolution to avoid tests timing out. Make the sinks sync to test closer
to reality. Fix Makefile to use uninstalled interface.
* decouple image capturing from image post-processing and encoding
* post image-captured message after image is captured
* post preview-image message with snapshot of captured image
Create output caps from input caps, so we maintain any fields we
might get on the input caps, such as codec_data or rate and channels.
Set channels and rate on the output caps if we don't have input caps
or they don't contain such fields. We do this partly because we can,
but also because some muxers need this information. Tagreadbin will
also be happy about this.
Don't allow setting filename via img-done signal parameter but force app
use filename property. Don't stop capture when setting filename property.
Update check unit test based on the change.
The video was recorded for too long for the test timeouts. Also the verification
suite did not had custom timouts at all. Also split the verification for images
and video to get better reporting.
Use playbin2 for validation. Use tmp_dir for capturing. Wait with g_cond for
burst capture finish. Cleanup some g_object_set. Add some logging to ease
tracing.
Original commit message from CVS:
* gst/mxf/mxfaes-bwf.c: (mxf_bwf_handle_essence_element),
(mxf_aes3_handle_essence_element):
* gst/mxf/mxfalaw.c: (mxf_alaw_handle_essence_element):
* gst/mxf/mxfd10.c: (mxf_d10_picture_handle_essence_element),
(mxf_d10_sound_handle_essence_element):
* gst/mxf/mxfdemux.c: (gst_mxf_demux_pad_init),
(gst_mxf_demux_choose_package),
(gst_mxf_demux_handle_header_metadata_update_streams),
(gst_mxf_demux_pad_next_component),
(gst_mxf_demux_handle_generic_container_essence_element),
(gst_mxf_demux_parse_footer_metadata),
(gst_mxf_demux_handle_klv_packet), (gst_mxf_demux_src_query):
* gst/mxf/mxfdv-dif.c: (mxf_dv_dif_handle_essence_element):
* gst/mxf/mxfjpeg2000.c: (mxf_jpeg2000_handle_essence_element):
* gst/mxf/mxfmetadata.c: (mxf_metadata_sequence_init),
(mxf_metadata_structural_component_init),
(mxf_metadata_generic_picture_essence_descriptor_init):
* gst/mxf/mxfmpeg.c: (mxf_mpeg_video_handle_essence_element),
(mxf_mpeg_audio_handle_essence_element):
* gst/mxf/mxfparse.h:
* gst/mxf/mxfup.c: (mxf_up_handle_essence_element):
* gst/mxf/mxfvc3.c: (mxf_vc3_handle_essence_element):
* tests/check/elements/mxfdemux.c: (_sink_chain):
Implement support for OP2a/b/c and OP3a/b/c, i.e. tracks with
more than a single component. This currently only works for
the case where the components are stored in playback order
in the file.
Set some more default/distinguished values for the structural
metadata.
Make some types more strict by choosing the correct subclasses.
Set DISCONT flag on buffers after a component switch.
Take the last partition from the random index pack for the footer
partition of the header partition doesn't reference the footer
partition. This gives us the final structural metadata for
some more files in the beginning.
Original commit message from CVS:
* tests/check/elements/mxfdemux.c: (_sink_event):
* tests/check/elements/mxfdemux.h:
Make sure the main loop is already running when handling the EOS
event in pull mode. This works around a race condition that can
happen if the element goes into PLAYING, handles everything and
sends EOS before the main loop is started.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/mxfdemux.c: (_pad_added), (_sink_chain),
(_sink_event), (_create_sink_pad), (_create_src_pad_push),
(_src_getrange), (_src_query), (_create_src_pad_pull),
(GST_START_TEST), (mxfdemux_suite):
* tests/check/elements/mxfdemux.h:
Add push and pull mode unit test for mxfdemux.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts 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.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
Add unit tests for new parsers.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/qtmux.c: (setup_src_pad),
(teardown_src_pad), (setup_qtmux), (cleanup_qtmux),
(check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main):
Add unit test for qtmux.
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.
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.
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.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* ext/x264/gstx264enc.c:
* tests/check/Makefile.am:
* tests/check/elements/x264enc.c: (setup_x264enc),
(cleanup_x264enc), (GST_START_TEST), (x264enc_suite), (main):
Add documentation and unit test for x264enc.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_finalize), (gst_audio_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_set_property), (gst_interleave_get_property),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_collected):
* gst/interleave/interleave.h:
Allow setting channel positions via a property and allow using the
channel positions on the input as the channel positions of the output.
Fix some broken logic and memory leaks.
* tests/check/Makefile.am:
* tests/check/elements/interleave.c: (src_handoff_float32),
(sink_handoff_float32), (GST_START_TEST), (interleave_suite):
Add unit tests for checking correct handling of channel positions.
Original commit message from CVS:
* gst/interleave/deinterleave.c:
Add another example launch line.
* gst/interleave/interleave.c: (interleave_24),
(gst_interleave_finalize), (gst_interleave_base_init),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_change_state), (__remove_channels),
(__set_channels), (gst_interleave_sink_getcaps),
(gst_interleave_set_process_function),
(gst_interleave_sink_setcaps), (gst_interleave_sink_event),
(gst_interleave_src_query_duration), (gst_interleave_src_query),
(forward_event_func), (forward_event), (gst_interleave_src_event),
(gst_interleave_collected):
* gst/interleave/interleave.h:
Major rewrite of interleave using GstCollectpads. This new version
also supports almost all raw audio formats and has better caps
negotiation. Fixes bug #506594.
Also update docs and add some more examples.
* tests/check/elements/interleave.c: (interleave_chain_func),
(GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
(interleave_suite):
Add some more extensive unit tests for interleave.
Original commit message from CVS:
* tests/check/elements/deinterleave.c: (GST_START_TEST):
Set keep-positions property to TRUE for the 8 channel test to ensure
that the original channel position is set on the output.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c: (deinterleave_24),
(gst_deinterleave_finalize), (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
(gst_deinterleave_set_process_function),
(gst_deinterleave_sink_setcaps), (__remove_channels),
(__set_channels), (gst_deinterleave_getcaps),
(gst_deinterleave_process), (gst_deinterleave_chain),
(gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Add support for all raw audio formats and provide better negotiation
if the caps are changing.
Don't allow changes of the channel positions and set the position of
the corresponding channel on the src pad caps.
General cleanup and smaller bugfixes.
* tests/check/elements/deinterleave.c: (float_buffer_check_probe):
Check the channel positions on the output buffer caps.
Original commit message from CVS:
* tests/check/Makefile.am:
Add deinterleave unit test to VALGRIND_TO_FIX, since it causes
weird invalid free errors in valgrind/libc after _exit for some
reason.
* tests/check/elements/deinterleave.c: (pads_created),
(set_channel_positions), (src_handoff_float32_8ch),
(float_buffer_check_probe),
(pad_added_setup_data_check_float32_8ch_cb),
(make_fake_src_8chans_float32), (GST_START_TEST),
(deinterleave_suite):
Add some more deinterleave unit test bits I had locally.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.h:
* gst/interleave/plugin.h:
Split definitions into separate header files for better documentation
generation.
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
(gst_deinterleave_process):
Don't use alloca, allow caps changes as long as the number of channels
does not change, don't use g_warning, return NOT_NEGOTIATED as early
as possible and some other cleanup.
* gst/interleave/interleave.c: (gst_interleave_base_init),
(gst_interleave_class_init):
Do some random cleanup.
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(deinterleave_chain_func), (deinterleave_pad_added),
(deinterleave_suite):
Add unit tests for the deinterleave element.
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.
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.
Original commit message from CVS:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST):
Only check if the generated fingerprints are valid Base64. The
fingerprints are different when running on different architectures
which is a) no problem because the fingerprints are tolerant enough
and b) is caused by libofa. Fixes bug #528266.
Original commit message from CVS:
* tests/check/elements/souphttpsrc.c: (got_buffer),
(souphttpsrc_suite):
Increase the timeout for the internet tests to 250 seconds
and check for NULL caps instead of just crashing.
The real fix would be to implement an shoutcast server for the unit test
instead of relying on a working internet connection.
Fixes bug #521749.
Original commit message from CVS:
* tests/check/elements/ofa.c: (GST_START_TEST):
Also check that we have processed at least 135 seconds of audio
until we stop and calculated a fingerprint.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST),
(ofa_suite), (main):
Add simple unit tests for the OFA plugin.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb),
(gst_soup_http_src_chunk_allocator),
(gst_soup_http_src_got_chunk_cb),
(gst_soup_http_src_uri_get_protocols):
Don't autoplug souphttpsrc for dav/davs. This is better handled by
GIO and GnomeVFS as they provide authentication.
Don't leak the icy caps if we already set them and get a new
icy-metaint header.
Try harder to set the icy caps on the output buffer to have correct
caps for the first buffer already.
* tests/check/elements/souphttpsrc.c: (got_buffer),
(GST_START_TEST):
Check that we get a buffer with application/x-icy caps if iradio-mode
is enabled and we have an icecast URL.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_input_selector_class_init),
(gst_input_selector_init), (gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_get_property),
(gst_input_selector_request_new_pad),
(gst_input_selector_release_pad),
(gst_input_selector_push_pending_stop),
(gst_input_selector_switch):
* gst/selector/gstinputselector.h:
Various cleanups.
Added tags to the pads.
Select active pad based on the pad object instead of its name.
Fix refcount in set_active_pad.
Add property to get the number of pads.
* gst/selector/gstoutputselector.c:
(gst_output_selector_class_init),
(gst_output_selector_set_property),
(gst_output_selector_get_property):
Various cleanups.
Select the active pad based on the pad object instead of its name.
Fix locking when setting the active pad.
* gst/selector/gstselector-marshal.list:
* tests/check/elements/selector.c: (cleanup_pad),
(selector_set_active_pad), (run_input_selector_buffer_count):
Fixes for pad instead of padname for pad selection.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
(gst_soup_http_src_init), (gst_soup_http_src_dispose),
(gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
(gst_soup_http_src_create):
* ext/soup/gstsouphttpsrc.h:
* tests/check/elements/souphttpsrc.c: (run_test), (GST_START_TEST),
(souphttpsrc_suite):
Add support for specifying a list of cookies to be passed in
the HTTP request. Fixes bug #518722.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
* gst/selector/gstinputselector.h:
Added "select-all" property to make it work like aggregator in 0.8.
* gst/selector/gstoutputselector.c:
Fix resend-latest behavoiur.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/selector.c:
Add unit tests for selector.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
Use g_file_[sg]et_contents() instead of using stdio functions.
Should be less error prone.
* tests/check/elements/multifile.c:
Create a temporary directory using standard functions instead of
creating a directory in the current dir.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Remove equalizer plugin docs
* tests/check/Makefile.am:
Add GST_OPTION_CFLAGS, to get -Werror -Wall into the tests as for
other modules.
* tests/check/elements/multifile.c:
* tests/check/elements/rganalysis.c:
* tests/check/elements/rglimiter.c:
Fix compiler warnings from -Wall -Werror
Original commit message from CVS:
* configure.ac:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-xingheader.xml:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c:
* tests/check/elements/xingmux_testdata.h:
Remove the xingmux plugin, as the element has moved into
mpegaudioparse in -ugly.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
Add documentation for the xingheader plugin.
* tests/check/elements/xingmux.c: (GST_START_TEST):
Set element state to PLAYING before doing something else.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c: (setup_xingmux),
(cleanup_xingmux), (GST_START_TEST), (xingmux_suite), (main):
* tests/check/elements/xingmux_testdata.h:
Add simple unit test for the xingmux element.
* gst/xingheader/gstxingmux.c: (generate_xing_header),
(gst_xing_mux_finalize), (xing_reset):
Fix a memleak and invalid seek tables with less than 100 MP3 frames.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/equalizer.c: (setup_equalizer),
(cleanup_equalizer), (GST_START_TEST), (equalizer_suite), (main):
Add some minimal tests for the equalizer plugin.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_request_new_pad):
Let's not call every request pad we create "sink%d", that'll
create problems if there's to be more than one pad. Fixes#490682.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/interleave.c:
Add unit test for the above.
Original commit message from CVS:
Patch by: Thomas Green <tom78999 gmail com>
* ext/neon/gstneonhttpsrc.c:
With libneon 2.6, we need to set the NE_SESSFLAG_ICYPROTO
flag if we want ICY streams to be handled too, otherwise
libneon will error out with a 'can't parse reponse' error.
Fixes#474696.
* tests/check/elements/neonhttpsrc.c:
Unit test for the above by Yours Truly.
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST),
(bpwsinc_suite):
* tests/check/elements/lpwsinc.c: (GST_START_TEST),
(lpwsinc_suite):
Also test everything in 32 bit float mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/bpwsinc.c: (setup_bpwsinc),
(cleanup_bpwsinc), (GST_START_TEST), (bpwsinc_suite), (main):
Add unit tests for bpwsinc, testing fundamental functionality again.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/lpwsinc.c: (setup_lpwsinc),
(cleanup_lpwsinc), (GST_START_TEST), (lpwsinc_suite), (main):
Add unit tests for lpwsinc, testing fundamental functionality.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/timidity.c (GST_START_TEST, timidity_suite,
main):
Add typefind test for midi.
Original commit message from CVS:
* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
(gst_y4m_encode_setcaps):
* tests/check/elements/y4menc.c: (GST_START_TEST):
Plug some leaks; try to make build bot happy again.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes#421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.
Original commit message from CVS:
Patch by: Lutz Mueller <lutz topfrose de>
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_class_init),
(gst_neonhttp_src_init), (gst_neonhttp_src_set_property),
(gst_neonhttp_src_set_uri), (gst_neonhttp_src_set_proxy),
(gst_neonhttp_src_send_request_and_redirect),
(gst_neonhttp_src_uri_set_uri):
* ext/neon/gstneonhttpsrc.h:
Simplify _set_uri() and _set_proxy() and remove the unused ishttp
member (#388050).
* tests/check/elements/neonhttpsrc.c: (GST_START_TEST):
Fix bogus URI to something that actually exists, otherwise we just
bypass the test (and also to something that doesn't redirect, since
neonhttpsrc doesn't seem to handle this very gracefully yet)
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* configure.ac:
* sys/Makefile.am:
* sys/directsound/Makefile.am:
* sys/directsound/gstdirectsoundsink.c:
(gst_directsoundsink_reset):
Add directsoundsink to build and dist it, so it gets built when
compiling with MingW on win32 and the required headers and libraries
are available (fixes: #392638). Also simplify DirectDraw check a bit.
* tests/check/elements/.cvsignore:
Fix CVS ignore for neonhttpsrc test binary.
Original commit message from CVS:
* tests/check/elements/videocrop.c: (GST_START_TEST),
(videocrop_test_cropping_init_context):
When we can't create an element needed for the test, print a message
detailing which element it actually is that's missing (#390673).
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_create),
(send_request_and_redirect):
Fix minor mem leak in redirect code.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/neonhttpsrc.c: (handoff_cb),
(GST_START_TEST), (neonhttpsrc_suite):
* tests/check/gst-plugins-bad.supp:
Add super-basic unit test for #384140.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_set_wp_config):
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_create_src_pad):
* gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_create_pads):
* tests/check/elements/wavpackparse.c: (wavpackparse_found_pad):
Activate pads before adding them to running element.
Original commit message from CVS:
* configure.ac:
Bump requirements of -base (videocrop test case needs this).
* gst/videocrop/gstvideocrop.c:
Document sloppy handling of subsampled chroma planes if
left/top cropping is an odd number.
* tests/check/elements/videocrop.c: (handoff_cb),
(videocrop_test_cropping_init_context),
(videocrop_test_cropping_deinit_context),
(videocrop_test_cropping), (check_1x1_buffer), (GST_START_TEST),
(videocrop_suite), (main):
Add another unit test that crops the input to 1x1 (and checks
that that pixel has the expected values in a number of formats).
Original commit message from CVS:
* tests/check/elements/videocrop.c: (handoff_cb),
(buffer_probe_cb), (test_caps_transform), (test_passthrough),
(notgst_value_list_get_nth_int), (videocrop_suite):
More tests: check passthrough mode and caps transform in
both directions with fixed values, ranges and lists.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
I forgot to include the file containing the #define :)
Now includes "config.h"
Original commit message from CVS:
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
proxying get/set caps is the wrong thing to do, since we really
do change caps quite fundamentally
* tests/check/elements/gdpdepay.c:
* tests/check/elements/gdppay.c:
remove declaration of buffers, it's already done in gstcheck.h
Original commit message from CVS:
* tests/check/elements/audioresample.c: (test_reuse),
(audioresample_suite):
Add test case for bug #342789 fixed below.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (gst_audioresample_init),
(audioresample_start), (audioresample_stop),
(gst_audioresample_set_property), (gst_audioresample_get_property):
Implement GstBaseTransform::start and ::stop so that audioresample
can clear its internal state properly and be reused insted of
causing non-negotiated errors with playbin under some circumstances
(#342789).
* tests/check/elements/audioresample.c: (setup_audioresample),
(cleanup_audioresample):
Need to set element state here so that ::start and ::stop are
called.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (set_channel_positions),
(get_float_mc_caps), (get_int_mc_caps):
* tests/check/elements/audioresample.c:
* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
* tests/check/elements/videorate.c:
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
* tests/check/elements/volume.c:
* tests/check/elements/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
Don't busy-wait in tests; this was causing test timeouts very
frequently when running under valgrind.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
(gst_gdp_pay_get_property):
add crc-header and crc-payload properties
don't error out on some things that are recoverable
* tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
add test for crc
