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).
