Ensures the test can run on systems without alsa (or any audio output for
that matter), and will avoid people running build slaves wondering what
the hell was beeping during the night :)
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.
This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive
Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.
It also drops some packets to activate restransmission and
check they are actually retransmited.
This unit test verifies that the rtxsend element correctly maintains
a buffer of already transmitted rtp packets and that it can
re-transmit all of them correctly on demand. It also verifies
that the limit of this buffer (max-size-packets property) is respected.
Several senders / one receiver
Similar than test_drop_one_sender but with multiple senders
mixed through the funnel element.
It drops some packets and checks that they are retransmited
correctly.
Test for one sender / one receiver
Build the pipeline
videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
and drop some buffers between rtprtxsend and rtprtxreceive
Then it checks that every dropped packet has been re-sent.
It also checks that not too much requests has been sent.
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
This test checks that when we have multiple internal sender sources
in rtpsession, SRs contain RBs for every other sender source, and that
they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
which is the max number of RBs that can fit in a SR.
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.
This reverts commit 26040ee38c
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
These new tests send a tag event before seding the buffer. Tested case are an
empty tag list, a tag list with orientation-180 set and an invalid orientation value.
https://bugzilla.gnome.org/show_bug.cgi?id=719497
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
Silencing this warning:
elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum <anonymous>’ [-Werror=enum-compare]
if (status != SOUP_STATUS_OK && !send_error_doc)
With gcc 4.8.2 (debian)
Store both DTS and PTS on buffers.
Make a queue for srcpad events.
Activate pads after linking so that we don't get RECONFIGURE events.
Add test for retransmission.
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
In 0.10 elements would post tag messages on the bus
directly, and rganalysis would only post a tag message
when it changed tags. In 1.0, only sinks post tag
messages when they receive the serialised tag event.
This means that we get an additional tag message on
the bus now where we didn't expect one before.
https://bugzilla.gnome.org/show_bug.cgi?id=695090
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
Previously we would skip level message when processing buffers > the requested
interval. Also the message frequency would contain quite some jitter due to only
considering them at the end of buffers.
Cleanup the tests while we're at it.
We're testing with an http server on localhost, but don't support
an exception list for the http_proxy, so just unset the environment
variable to make sure we can run this test properly even if the
environment has http_proxy set.
Also, don't skip all tests if there is an issue with the SSL server,
just run the non-SSL tests then.
https://jenkins.qa.ubuntu.com/view/Raring/view/JHBuild%20Gnome/job/jhbuild-amd64-gst-plugins-good/
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.
Also add unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=689336
This change enable automatic cropping using -1 set to left, top, right or
bottom property. In the case both side are set to automatic cropping, the
croping will be done equally on both side (in the odd case, right and
bottom cropping will be 1 pixel more).
https://bugzilla.gnome.org/show_bug.cgi?id=687761
- 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.
