- dist input files
- fix sample leak
- simplify check for elements
- only run mpg123 test if mpg123 is available and selected
- fix build in uninstalled setup
https://bugzilla.gnome.org/show_bug.cgi?id=686595
Before the element would post messages on the bus itself, now
the sinks do that based on the tag events they receive. But
since we don't have proper sink elements in these unit tests,
but just dangling pads, we have to post the tag messages the
test checks for ourselves.
Down from 52/55 failing to 7/52 failing.
Fix deinterlace unit test. Need to set right field on output caps.
Also remove right field (not old 0.10 "interlaced" boolean field)
from caps in unit test before comparing old and new.
Was waiting for a tag message on the bus, which would never
come, because elements don't post those themselves any more
but let sinks post them from tag events. Only that there are
no sinks in this unit test.
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.
rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes
videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink
work.
Need to add h263version field to input caps since the
payloader sink get_caps function will contain it in the
the caps, and the stricter caps subset check requires
this to be present in the input caps as well then.
Must flush after EOS before sending more buffers or
another EOS event, or the event or buffer will be
rejected. Also send a SEGMENT event at the start
of each stream for good measure.
Must flush after EOS before sending more buffers or
another EOS event, or the event or buffer will be
rejected. Also send a SEGMENT event at the start
of each stream for good measure.
This never really took off and is most likely completely
unused. If there is still a need for this, it should
probably be done differently, perhaps inside oggdemux/mux.
Or perhaps it should just be a guint64 channel mask, which would
be nicer in C, but more awkward for bindings (even more so since
we can't add a flags type for it, since that only supports guint
size flags). Fixes wavenc unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=669643
Discard caps event when checking for and counting various tag events,
and remove all testing of 8 bits depth in 16 bits width format since
it no longer exists.
Feed data into the pipeline using appsrc instead of fdsrc and
a pipe. Store unsigned byte values in guint8 instead of char.
Getting rid of the capsfilter also helps to avoid 'format is
not fully specified' warnings when pushing "video/x-h264" data
into rtph264pay with fully specified h264 caps in the sink template.
Most signal processing texts, including MATLAB, use the following convention for IIR filter coefficients:
a_0 y[n] + a_1 y[n-1] + ... + a_M y[n-M] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N]
Usually, a_0 is set to 1 because the coefficients can always be rescaled, giving
y[n] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] - a_1 y[n-1] - ... - a_M y[n-M]
The convention that was previously used by audiofxbaseiirfilter and derived class had the a and b coefficients swapped, and did not have the minus signs.
This change makes the audiofx plugin use the more common convention described above.
Some tests assumed that tag events would always pushed through
immediately, which isn't the case any longer, so push a newsegment
event and an empty buffer first.
Support for TAG_IMAGE and TAG_ATTACHMENT is commented out; this requires caps
on buffers which is gone from 0.11.
Segment handling in the demuxer is a bit complex; I added some FIXME comments
in places where I'm not yet sure if I ported correctly.
Conflicts:
ext/pulse/pulseaudiosink.c
ext/pulse/pulsesrc.c
gst/audioparsers/gstaacparse.c
gst/audioparsers/gstamrparse.c
gst/audioparsers/gstdcaparse.c
gst/audioparsers/gstflacparse.c
gst/effectv/gstradioac.c
gst/effectv/gstradioac.h
gst/effectv/gstripple.c
Some possible FIXMEs remaining in the audio parser getcaps functions.
Tests fail to build because g_mkdtemp is available from glib since
2.26.
This patch adds a condition around the redefinition of
g_mkdtemp on the tests to only build it if glib is older than
2.26.
When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading
requires each buffer to contain 6 blocks from each substream. This adds
code to collect all the frames needed to meet this requirement before
pushing out a buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=650313
This is needed for automatic transcoding using encodebin. Our typefinder
does not always add a variant to the found caps, and encodebin needs
an *exact* match to the caps on the source pad template, so we need
to add the variant-less video/quicktime caps to the template as well
for encodebin to be able to find it. Add unit test for this as well.
https://bugzilla.gnome.org/show_bug.cgi?id=642879
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.
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.
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.
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.
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.
Images might have framerate=0/1 in the caps, which caused an
assertion on deinterlace. I don't know of interlaced image formats
but deinterlace might be hardcoded on some generic pipelines and
it shouldn't assert.
The fix was to set field_duration to 0 if the input has a framerate
with a 0 numerator.
This patch also adds checks for this situation on the unit tests.
https://bugzilla.gnome.org/show_bug.cgi?id=641400
First bring down everything to NULL before attempting to unlink
or unref anything.
Avoids the tests just hanging there for ever waiting to acquire a
lock that doesn't exist anymore.
If setting the pipeline to PLAYING before issuing the seek, buffers
are already arriving at the sink before the seek is handled and
will have the wrong timestamps and everything.
Fixes bug #625547.
So matroska's Block structure has no keyframe flag, only the SimpleBlock has it.
To detect keyframes in Blocks, it is just the BlockGroup container that needs
to have a ReferenceBlock attached if it is a delta frame in video.
deinterlace now always adds the interlaced field to the output caps,
if it wasn't present in the input caps the output caps will still
contain interlaced=false.
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.
Now the alpha is multiplied with the already existing alpha
value instead of simply ignoring it and the luma/chroma values
are kept, even if the output is 100% transparent.
Make sure we set a base_time on the element.
Fix the timeout to at least twice the jitterbuffer latency.
Enable previously failing tests.
Remove impossible checks.
CHange the backwards test to always send first buffer first to have a define
basetime. Add another test that sends buffers backwards to assert that only
first sent buffer is keep and used as basetime. Disabled those tests still,
as its not passing/failing consitently and file a bug for jitterbuffer.
The souphttpsrc test wasn't compiling. The soup-misc.h header is needed for
soup_ssl_supported.
Fix the y4menc test to use a 'progressive' header for the test data now that
the element outputs correct interlacing info.
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
flvmux only accepts raw audio in little endian, but audiotestsrc
produces audio in the native endianness, which makes linking
between audiotestsrc and flvmux fail on big endian machines. Add
an audioconvert element in between the two to fix this.
Reversing the unsynchronisation seems to work slightly differently
for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
sizes in the frame header, so the unsynchronisation is applied to
the whole frame data including all the frame headers. v2.4 frames
have sync-safe sizes, however, so the unsynchronisation only needs
to be applied to the actual frame data, and it seems that's what's
being done as well. So we need to undo the unsynchronisation on a
per-frame basis for v2.4 tags for things to work properly.
Fixes extraction of coverart/images from APIC frames in ID3 v2.4
tags (#588148).
Add unit test for this as well.
Request pads are removed by the element instance in PAUSED->READY
so we need to re-request pads for every run and link them again.
Last fix for bug #590447.
Move unit test data into the directory where it belongs and make in particular
the flacdec unit test cd into the directory with the test files instead of making
assumptions about the current working directory in that unit test. As a side effect
of movng those files, there's only one EXTRA_DIST in tests/check/Makefile.am now,
which is likely to work better than having two. Hopefully fixes#582753.
The test creates buffers with non-silence, sets the GAP
flag on it and expects rganalysis to ignore the content and assume silence.
That's not the way how GAP buffers should be used, if the GAP flag is set
elements *can* assume that they only contain silence but they're not *required*
to assume that. The GAP flag must only be set on silence buffers.
Fixes bug #582252.
Cast the new value for the "delay" property to GstClockTime.
Integers without type are passed to vararg functions with
an integer type that can hold a pointer.
A recent commit added video/x-raw-gray support to videocrop. However
this lets the videocrop unit test fail. Because videotestsrc can't
generate this format.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
some additional filtering to prevent a metallic-sounding
result.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps):
Some cleanups, refactoring and minor enhancements in caps handling.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_init), (gst_matroska_pad_reset),
(gst_matroska_pad_free), (gst_matroska_mux_reset),
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad):
* tests/check/elements/matroskamux.c: (teardown_src_pad):
Only remove, release or reset what is appropriate upon state change.
Original commit message from CVS:
* tests/check/elements/videocrop.c: (check_1x1_buffer):
Update the unit test for the new color values for BT.601 red.
Fixes bug #563510.
Original commit message from CVS:
* tests/check/elements/souphttpsrc.c: (GST_START_TEST),
(run_server):
The ports in libsoup are unsigned integers and not signed
integers.
Original commit message from CVS:
* tests/check/elements/icydemux.c: (icydemux_found_pad):
Add some refcount check
* tests/check/elements/rtp-payloading.c: (rtp_pipeline_run):
Don't ignore the result of write(), fixes a compiler warning for me.
* tests/icles/videobox-test.c: (main):
Make the output a little more pretty.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another URL.
* tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags):
* tests/check/elements/rglimiter.c: (GST_START_TEST):
Add some more debug info.
Original commit message from CVS:
* tests/check/elements/cmmldec.c: (GST_START_TEST):
* tests/check/elements/rtp-payloading.c: (rtp_pipeline_create),
(rtp_pipeline_run):
* tests/check/elements/souphttpsrc.c: (souphttpsrc_suite):
Don't use declarations after statements.
Original commit message from CVS:
* tests/check/elements/avimux.c: (check_avimux_pad):
Adjust avimux unit test according to increased streamheader size.
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:
* docs/plugins/.cvsignore:
* tests/check/elements/.cvsignore:
Ignore some more generated things
* tests/check/Makefile.am:
Ignore OSS elements in the state changes test too.
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:
* tests/check/elements/multifile.c:
Include stdlib.h and unistd.h for mkdtemp. Some platforms have it
declared in the former, some have it declared in the latter.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable some more elements in the state test.
Add a define so the soup test can find the test files
it needs at runtime.
* tests/check/elements/souphttpsrc.c: (run_server):
Add a define so the soup test can find the test files
it needs at runtime.
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/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/gdkpixbufsink.c:
Add unit test for gdkpixbufsink element.
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:
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:
patch by: Wim Taymans <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_create):
* tests/check/elements/multifile.c: (GST_START_TEST):
Close some memory leaks spotted by the unit test. Fixes bug #515697.
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/spectrum/Makefile.am:
Fix includes order
* tests/check/Makefile.am:
Exclude v4l2src from the states test - it takes too long to start.
* tests/check/elements/spectrum.c:
Make the test run properly with CK_FORK=no
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:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
* gst/audiofx/audiowsinclimit.h:
* tests/check/Makefile.am:
* tests/check/elements/audiowsincband.c:
* tests/check/elements/audiowsinclimit.c:
Move the lpwsinc and bpwsinc elements from gst-plugins-bad into
the audiofx plugin, and rename to audiowsinclimit and audiowsincband
respectively.
Fixes: #467666
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_chain):
* tests/check/elements/icydemux.c:
Return GST_FLOW_NOT_NEGOTIATED if we get a buffer without
caps, and add a somewhat useful debug message. Plus test.
Original commit message from CVS:
* tests/check/Makefile.am:
Add rtp-payloading test to VALGRIND_TO_FIX.
* tests/check/elements/rtp-payloading.c:
Add semicolons after GST_TEST_END so gst-indent gets the
formatting right; make test less verbose in general, but
more verbose in the error case (which should probably
make the test fail anyway).
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rtp-payloading.c:
Add add testsuite for the rtp-payloader that tries simulating
dataflow. Needs more test data.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix up some CFLAGS sets.
Don't include gconfvideosrc in the states test.
* tests/check/elements/autodetect.c: (GST_START_TEST):
Add some error strings to fail_unless arguments to fix some weird
compiler errors on Solaris.
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:
* tests/check/elements/icydemux.c: (GST_START_TEST), (icydemux_suite):
Don't use deprecated GST_PLUGIN_DEFINE_STATIC.
Original commit message from CVS:
* tests/check/elements/avisubtitle.c: (check_correct_buffer):
Small unit test fix (has no practical impact at the moment,
since we're only feeding utf8 and hence just create a sub-
buffer for the output).
Original commit message from CVS:
* gst/avi/Makefile.am:
* gst/avi/gstavi.c:
* gst/avi/gstavisubtitle.c:
* gst/avi/gstavisubtitle.h:
* tests/check/Makefile.am:
* tests/check/elements/avisubtitle.c:
* win32/common/config.h:
Add avi subtitle element for bug #442034. Need seeking support
and more support for character conversion.
Original commit message from CVS:
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
Don't check the caps of the output buffer if they're equal some
other caps. The caps can change in a backward compatible way
and did at this point.
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:
Based on patch by: Jason Kivlighn <jkivlighn gmail com>
* gst/id3demux/id3v2frames.c:
Extract license/copyright URIs from ID3v2 WCOP frames
(Fixes#447000).
* tests/check/elements/id3demux.c:
* tests/files/Makefile.am:
* tests/files/id3-447000-wcop.tag:
Add simple unit test.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gsttaglibmux.c:
* tests/check/elements/apev2mux.c:
Update my mail address.
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:
* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
(audiochebyshevfreqband_suite):
* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
(audiochebyshevfreqlimit_suite):
Also test 32 bit float mode and the type 2 variants of the filters.
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:
* configure.ac:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
Use the new buffer clipping function from gstaudio here and
require gst-plugins-base CVS.
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
For framed Wavpack buffers we require a valid timestamp.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/audiopanorama.c: (GST_START_TEST):
* tests/check/elements/videocrop.c: (GST_START_TEST):
* tests/check/elements/videofilter.c:
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
error flags are included and it errors out on compiler warnings
for CVS builds; remove unused variables in various unit tests.
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
Double-check that RGB input caps are really RGBA caps (apparently
the core doesn't always catch it if those caps aren't a subset of
our template caps, also see #421543). Fixes#429319 in a way.
Also, don't leak the pad template in the transform_caps function.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/alphacolor.c: (setup_alphacolor),
(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
(GST_START_TEST), (alphacolor_suite):
Add some basic unit tests for alphacolor.
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:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_new):
Actually translate sunaudio mixer track labels instead of just
marking the strings as translatable (#377306); clean up weird
label string mapping code that serves no apparent purpose. Also
set the 'untranslated-label' property when creating mixer tracks
if the GstMixerTrack base class supports this.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/sunaudio.c: (GST_START_TEST),
(sunaudio_suite):
Very minimalistic unit test for sunaudiomixer element (compiles, but not
actually tested on a system where sunaudiomixer is available).
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_sink_activate):
Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
caps passed to it (previouslly one code path assumes it takes ownership
while another one assumes it doesn't).
* configure.ac:
* tests/files/Makefile.am:
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
Add directory where data for unit tests can be stored.
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
(read_tags_from_file), (run_check_for_file),
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
Add unit test for id3demux, and in particular for bug #407349. Only
testing pull-mode for now; push mode doesn't work yet because the test
files are smaller than ID3_TYPE_FIND_MIN_SIZE.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes#397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes#396057
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
* tests/check/elements/avimux.c: (teardown_src_pad):
Add support for more than one audio stream; write better AVIX
header; refactor code a bit; don't announce vorbis caps on our audio
sink pads since we don't support it anyway. Closes#379298.
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:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_request_new_pad):
Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
* tests/check/elements/matroskamux.c: (setup_src_pad),
(setup_sink_pad), (GST_START_TEST):
Activate pads before using them.
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:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
Activate pad before adding it to the already-running element.
* tests/check/elements/icydemux.c: (icydemux_found_pad):
Activate newly-created pad too.
Original commit message from CVS:
* tests/check/Makefile.am:
Disable autodetect test temporarily, so that the build bots
update -bad and the ranks of unreliable video sinks in there.
* tests/check/elements/autodetect.c: (GST_START_TEST):
Skip test if no usable videosink is found.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
Make static pad templates static to appease valgrind's leak
detector.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/autodetect.c: (GST_START_TEST),
(autodetect_suite):
Add simple test for the ghostpad lockup on shutdown fixed in core
CVS (audio bit disabled because it would need dozens of alsa
suppressions and I'm too lazy to add those now).
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
(gst_icydemux_typefind_or_forward):
* gst/icydemux/gsticydemux.h:
When we merge/collect multiple incoming buffers for typefinding
purposes, keep an initial 0 offset on the first outgoing buffer
as well (otherwise id3demux won't work right). Fixes#345449.
Also Make buffer metadata writable before setting buffer caps.
* tests/check/elements/icydemux.c: (typefind_succeed),
(cleanup_icydemux), (push_data), (GST_START_TEST),
(icydemux_suite):
Small test case for the above.
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:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
Do some extra sanity checks.
Fixes#350340.
* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
Check if clip->start_time is valid before adding the clip to the
track list.
Reset enc->preamble going from PAUSED to READY.
Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
only used for EOS.
Only post an error message if we were the one that created the fatal
GstFlowReturn value.
* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
Parse the seconds field of the npt-sec time format using %llu rather than
%d and check that the value scaled by GST_SECOND doesn't overflow.
Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
Lookup a clip's track with clip->track rather than clip->id which
makes no sense.
Identify a clip by its track and start time and not its xml id.
do some more input checking and make sure we don't do undefined shifts.
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
(cmml_tag_message_pop), (check_headers), (push_clip_full),
(push_clip), (push_empty_clip), (check_output_clip),
(GST_START_TEST), (cmmldec_suite):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
(check_headers), (push_clip), (check_clip_times), (check_clip),
(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
Added some more checks.
