We already have internally the information on what type of stream (audio,
video, container, subtitle, ...) a certain caps is.
Instead of forcing callers to specify which CODEC_TAG category a certain
caps is, use that information to make a smart choice.
Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
(if tag is specified it will be used, if caps is invalid it will be rejected,
...).
https://bugzilla.gnome.org/show_bug.cgi?id=702215
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
The array of factories should not contain a NULL element at the end
since the number of arguments is determined via G_N_ELEMENTS and the
NULL will be used as an argument to gst_element_factory_make() if
the other sources in the list weren't usable.
This allows getting a pad for a specific encoding profile, which can
be useful when there are several stream profiles of the same type.
Also update the encodebin unit tests so that we check that the returned
pad has the right caps.
https://bugzilla.gnome.org/show_bug.cgi?id=689845
This reverts commit adc9694ed7.
No need to restrict the conversion, we can handle interlace correctly. We
basically unpack each field, then convert each field to the target colorspace
and pack and interleave each field to the target format. We also disable any
fast path that can't deal with interlaced formats.
We were setting the query-func on the sink-pad, which got overwritten when
adding the new pad to collect pads. Instead register our query-func with the
collect pads object. This fixes filter caps. Add a test for it.
These override the variants without version suffix. Makes
'make check' work properly in environments that set the
suffixed variant for 1.0, such as jhbuild.
jhbuild already sets $GST_PLUGIN_PATH_1_0 which overrides $GST_PLUGIN_PATH. Set
both for the tests to see the locally built elements. Fixes 'make check' in
jhbuild.
A return value of FALSE here indicates that we don't have control-values. In
0.10 we were returning the default value of the property. Now we don't fill an
array with defaults in the ControlBinding, but leave it up to the element to
handle this case.
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)
+ Add tests that it behave correctly
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
There won't be a tag messages on the bus, because tags
are now sent downstream for sinks to post on the bus,
and there's no sink involved here that would do that.
Secondly, the audio decoder base class only sends the
tags out once it has received some non-header data as
input, which is not something we're providing here.
Fix invalid memory access caused by broken pointer arithmetic.
If we have a uint16_t *tmpbuf and add n * dest->stride to it, we
skip twice as much as we intended to because dest->stride is in
bytes and not in pixels. This made us write beyond the end of
our allocated temp buffer, and made the unit test crash.
Use multifdsink for pipes instead of multisocketsink,
to avoid "creating GSocket from fd 9: Socket operation
on non-socket "criticals from Gio. Test still fails,
but it fails in a different way now.
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
Since we now use videoconvert, which supports these.
Unfortunately videoscale still crashes with 64-bit formats
right now because of a too small temp buffer, but I'm sure
someone is going to fix this real soon now, just like the
other unit tests.
Bases on test app in bug #614296. Doesn't work reliably yet,
leads to not-negotiated errors sooner or later, even when
it's the same element being re-plugged.
Make the uri property getter return the next uri, like it was configured in the
setter.
Make a new current-uri and current-suburi property that reflects the currently
playing uri and suburi.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676665
Video base classes and theora plugin still needs to be ported again
Conflicts:
docs/libs/gst-plugins-base-libs-docs.sgml
docs/libs/gst-plugins-base-libs-sections.txt
docs/libs/gst-plugins-base-libs.types
ext/theora/gsttheoradec.c
ext/theora/gsttheoradec.h
ext/theora/gsttheoraenc.c
ext/theora/gsttheoraenc.h
gst-libs/gst/video/Makefile.am
gst-libs/gst/video/video.c
gst-libs/gst/video/video.h
gst/playback/gsturidecodebin.c
tests/check/libs/video.c
tests/check/pipelines/theoraenc.c
win32/common/libgstvideo.def
... such as setting input caps, ensuring to unmap and bearing in
mind that gst_buffer_make_writable usually does not provide
a separate memory copy/area.
gst_buffer_set_qdata() will leak the structure passed to it
when called incorrectly (e.g. on a non-metadata-writable buffer).
This is expected, but we must avoid doing that in valgrind.
Now that we no longer support all methods for all formats, we
need to cater for that in the transform function: we can't
transform formats not supported by the currently-selected
mehod.
make check, folks. It's da bomb.
Rename the offset field in GstVideoFormatInfo to poffset to avoid confusion with
the offset of the plane in the buffer. The poffset is the offset in the plane
where the first byte of the component data can be found.
Properly implement the COMP_OFFSET calculations.
Fix YV12 and YVU9, simply use the same offsets as the regular I420 and YUV9
variants, we use the plane info to reorder components already.
Improve the unit test.
We have removed things like protocol=gdp in the tcp elements
in favour of explicit gdppay/depay elements, so there's no need
to keep a public API and library for now. We can still add it
back later. Someone needs to think hard about 0.11 and gdp
anyway one of these days.
When the payload for an Exif tag is less than or equal to 4 bytes,
the data is simply put into the offset field. Fix writing these
kinds of payloads on big endian systems (and possibly also on
little endian systems). The caller will have already formatted
the bytes in memory according to the writer's endianness, so just
write out the bytes as they are in this case. Fixes tags unit test
on big endian systems.
We used to add a trailing \n to the end of generated xmp packets.
Windows viewer was unhappy with it and we fixed it in
96d2120c2b
The problem is that this caused xmp generated before this fix
to not be recognized and parsed anymore. This patch makes it
recognize xmp with the trailing \n and without, fixing the
regression. Also adds tests for it.
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
Previously, the source posted a TAG message before buffers would
even be pushed towards the sink, so we'd get the TAG message before
any ASYNC_DONE message. Now the tags get sent downstream to the sink
to get posted there, and the tag event will get queued and handled
later after preroll has finished, so now we get the ASYNC_DONE
message before the TAG message.
If they go when calling snd_config_update_free_global, they're
not really bug leaks, but more like intentional ones we don't
want to get told about.
https://bugzilla.gnome.org/show_bug.cgi?id=615342
There's no code whatsoever that uses these macros. If anyone
ever feels the need to resurrect them, we should add them to
gstutils.h in core or libgstaudio or so.
This will cause not-linked errors that usually don't happen
because normal decoders/parsers will set srcpad caps before
allocating buffers from downstream.
Make the element_factory_make_or_warn utility function try parsing
the input string as a bin if element_factory_make() fails. This makes
the --audiosink/--videosink commandline options accept a pipeline
string.
Make sure decodebin1 doesn't try to plug the same parser twice
in a row (so we can change all parsers to accept parsed input as
well without breaking applications still using the old decodebin1
element).
The testchannels audio test program is using -lgstaudio-0.10 to link
with libgstaudio which won't use the gstaudio library that was just
built but the one from the system. This is an issue since it means
we won't be testing the code from the current source tree, and it
also breaks the build when building on a system which don't have
a libgstaudio yet.
https://bugzilla.gnome.org/show_bug.cgi?id=652100
The old v4l interface has been deprecated for years and even
been removed from the kernel headers. If anyone still needs
this plugin, they can resurrect it in gst-plugins-bad, there's
no reason for it to be in -base.
Before the behaviour was undefined and implemented differently by elements,
now core checks for this (and other problems) and returns NULL and an assertion.
GtkFunction is gone, and there's no update policies for
GtkRanges any more (but the default was continuous anyway,
so no need to set it to that mode explicitly).
https://bugzilla.gnome.org/show_bug.cgi?id=639215
The unit tests are riddled with g_assert() and friends, sometimes
containing functional code like set_state() calls in them even
(looking at you, pipeline/capsfilter-renegotiation). Make sure we
don't disable assert and cast checks for the unit tests even if
this has been specified for the rest of the code base, e.g. via
--disable-glib-asserts.
Original commit message from CVS:
* gst/selector/gstoutputselector.c:
* tests/icles/output-selector-test.c:
Use BOILERPLATE macro and update test to the latest api changes.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
Add a fixme comment.
* gst/selector/gstoutputselector.c:
Fix same leak as in input-selector.
* tests/icles/output-selector-test.c:
Improve the test.
Adds a tag to inform what mode was used by a camera to calculate
the picture capturing exposure
Also adds mapping to exif and tests
API: GST_TAG_CAPTURING_METERING_MODE
https://bugzilla.gnome.org/show_bug.cgi?id=631773
Adds new tag for tagging sharpness processing used
when capturing an image. Also maps it in the exif
tags.
Tests included.
API: GST_TAG_CAPTURING_SHARPNESS
https://bugzilla.gnome.org/show_bug.cgi?id=631773
Make appsrc not set caps on buffers when its own caps is NULL.
This avoids calling make_metadata_writable on all buffers and
prevents losing buffer caps in case we are not replacing it
with something meaningful.
https://bugzilla.gnome.org/show_bug.cgi?id=630353
Binding generators apparently need this as they can't really know
that the callback is guaranteed to be called exactly once and that
the user_data can be freed at the end of it.
And deprecate the gulong versions. This is to support platforms
where sizeof(unsigned long) < sizeof(void *). Fixes#627565.
API: Add gst_x_overlay_set_window_handle()
API: Deprecate: gst_x_overlay_set_xwindow_id()
API: Add gst_x_overlay_got_window_handle()
API: Deprecate: gst_x_overlay_got_xwindow_id()
API: Add GstXOverlay::set_window_handle()
API: Deprecate: GstXOverlay::set_xwindow_id()
Adds mappings from:
GST_TAG_CAPTURING_EXPOSURE_PROGRAM -> ExposureProgram
GST_TAG_CAPTURING_EXPOSURE_MODE -> ExposureMode
GST_TAG_CAPTURING_SCENE_CAPTURE_TYPE -> SceneCaptureType
GST_TAG_CAPTURING_GAIN_ADJUSTMENT -> GainControl
GST_TAG_CAPTURING_WHITE_BALANCE -> WhiteBalance
GST_TAG_CAPTURING_CONTRAST -> Constrast
GST_TAG_CAPTURING_SATURATION -> Saturation
Also renames gst_tag_image_orientation_from_exif_value and
gst_tag_image_orientation_to_exif_value to remove the 'gst'
prefix and not including in the win32 defs.
Tests included.
If we find a bag of tags of type string in the xmp packet, we
should concat them, this is not the ideal approach, but at
least works for now as we don't know what type of tag it
is (simple, structure, seq, alt or bag)
So people can check what version of the gst-plugins-base libs they're
building against or linked against.
API: GST_PLUGINS_BASE_VERSION_MAJOR
API: GST_PLUGINS_BASE_VERSION_MINOR
API: GST_PLUGINS_BASE_VERSION_MICRO
API: GST_PLUGINS_BASE_VERSION_NANO
API: GST_CHECK_PLUGINS_BASE_VERSION
API: gst_plugins_base_version()
API: gst_plugins_base_version_string()
Use new gst_element_link_pads_full() function to link elements,
and disable all checks when linking (don't try this at home).
Down to 18s from 3m20s. Scary.
Adds mappings for:
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
GST_TAG_GEO_LOCATION_MOVEMENT_SPEED
GST_TAG_GEO_LOCATION_ELEVATION
Does some refactoring in the code to reduce number of parameters
passed to functions
Tests included.
Adds GST_TAG_GEO_LOCATION_MOVEMENT_SPEED,
GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION and
GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION to xmp
mappings.
Tests included.
Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
This should make sure arguments are passed to the linker in the right
order, and makes LDFLAGS usable again.
Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
Fixes#615697.
Tell status in top comment. Use debug logging instead of print to be able to
see timing issue in debug log viewer. Add more commandline flags. Test reverse
playback.
We know our plugins and examples are independent of each other, so may
just as well build them in parallel. Makes the output a bit messy, but
that shouldn't be a problem and can easily be avoided with make -j1.
Don't make libgstinterfaces (and thus libgstaudio etc.) indirectly depend
on libgstvideo by using the GstVideoRectangle helper structure in the API,
which causes undesirable dependencies, esp. with the gobject-introspection
(people will point and laugh at us if they find out that libgstaudio
depends on libgstvideo). Instead, pass the x, y, width and height parameters
directly to the function.
Re-fixes #610249.
Adds a mapping to the _ELEVATION tag, this is a different
mapping as it has to be mapped into exif:GPSAltitude and
exif:GPSAltitudeRef at the same time. So we needed to refactor
a little more to be able to deserialize it properly.
Now, when parsing a xmp buffer into a taglist all tags are
added to a list before being parsed so that when one of the
altitude tags are found the deserialization function can search
for its complementary tag to do the correct parsing
Fixes#613690
Fix up videorate test for latest videotestsrc changes: just check for
the important bits in the negotiated caps, not for exact equality with
our filter caps. Also don't leak the videorate element in the test.
Right now this mostly demonstatest what not works. That is seeking with
start-type = NONE to only update the rate and playing backwards. Also
it shows that non-flushing seeks tend to lockup adder. Separate unit tests
for the issues follow.
XMP metadata can be embedded in many media container formats. Implement own
parser and formatter that can be used to convert between an xpacket and a
GstTagList. Add unit tests.
Also set a title in the qt examples like it is now done in the gtk example.
Fix the newly added find_video_sink in the gtk example and add similar function
to the qt examples.
Try to find a working videosink, don't hardcode xvimagesink. Also
add some borders to window and give it a title so that it's clear
that this is really a Gtk+ window and not a window created by the
videosink.
Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
adds all boilerplate to configure for using c++. The qt based examples are
optional like their gtk counterparts.
The only difference between the two unit tests right now is,
that the decodebin2 test resets the element to READY before trying
to reuse it instead of NULL. decodebin2 guarantees to be reusable
without going back to NULL.
Add some utility functions for language tags and ISO-639
codes. These are useful for both GUIs and elements. The
iso-codes package is used for language name translations
if available.
API: gst_tag_get_language_codes()
API: gst_tag_get_language_name()
API: gst_tag_get_language_code()
API: gst_tag_get_language_code_iso_639_1()
API: gst_tag_get_language_code_iso_639_2B()
API: gst_tag_get_language_code_iso_639_2T()
Add a new video event to mark the start or end of a still-frame
sequence, and a parser function to identify and extract info from
such events.
API: gst_video_event_new_still_frame()
API: gst_video_event_parse_still_frame()
Fixes: #601942
This completes the deprecated GTK API fix in commits 81a0a986 and
79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
co take a gpointer, not a GtkObject.
the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
path before the local ones... resulting in the example applications picking
up the system-wide libraries and not the (potentially modified) uninstalled
libraries
Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
worked before the fix for bug #321532.
Also adds a check for negative track numbers and some unit tests for URI
parsing.
Fixes bug #595454.
Revert previous 'fix' for bug #588717 and fix it properly, whilst
maintaining the streamheader field on the output caps. Also make
sure we don't leak header buffers we couldn't push when downstream
is unlinked. Add unit test for the presence of the streamheader
field on the output caps and for the issue from bug #588717.
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
Keep the pipeline paused when we detect download buffering. The user has to
manually start the pipeline for now because we can't estimate when the buffering
will finish or when we have underrun.
This tests seeking on an adder that has a normal and a live source connected.
Wheter the current behavior is the desired one needs to be discussed still
(see #586033)
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes#585197.
Repaint the window black on expose, as this looks nicer when resizing or using
the expander. Also show time after slider, as this saves a whole line (nice on
small displays).
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes#579463.
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
Add some padding to the callbacks structure just to be safe.
Remove the now invisible marshaller methods from the docs.
Fix a comment in the unit test.
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes#571299.
Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.
Add a unit test for appsink.
Clean up some of the appsink docs.
API: GstAppSink::gst_app_sink_set_callbacks()
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes#567636.
Original commit message from CVS:
* tests/examples/app/appsrc_ex.c: (main):
Some comments.
When pulling a buffer we can get NULL when the element is EOS, don't try
to unref this NULL buffer.
Original commit message from CVS:
* configure.ac:
* tests/examples/seek/Makefile.am:
* tests/icles/Makefile.am:
Make the seek and colorkey examples depend on gtk+-x11 as they use
GDK_WINDOW_XID.
Fixes the build with gtk+-quartz.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes#564482.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_event):
Remove erroneous gst_buffer_ref().
* tests/check/libs/rtp.c: (GST_START_TEST):
Don't forget to unref the buffer once you're done with it.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek), (stop_cb),
(skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
(main):
Hook up the SKIP seek flag.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes#563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
* tests/icles/Makefile.am:
* tests/icles/test-colorkey.c:
Allow setting colorkey if possible. Implement property probe interface
for optional X features (autopaint-colorkey, double-buffer and
colorkey). Fixes#554533
Original commit message from CVS:
* tests/icles/.cvsignore:
update ignore file.
* tests/icles/Makefile.am:
* tests/icles/test-box.c: (make_pipeline), (main):
Add another interactive command line experimentation suite for
dynamically boxing/cropping/saling an input video.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
* sys/xvimage/xvimagesink.h:
Add a "draw-border" property that can be set to false to disable
drawing borders.
* tests/icles/test-colorkey.c:
* tests/icles/Makefile.am:
Add new test application for the colorkey handling.
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
Original commit message from CVS:
* tests/check/elements/audioresample.c: (setup_audioresample),
(fail_unless_perfect_stream), (test_perfect_stream_instance),
(test_discont_stream_instance):
Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Add debugging for coherence.
Original commit message from CVS:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/codec-select.c: (make_encoder),
(make_pipeline), (do_switch), (my_bus_callback), (main):
Add example app that dynamically switches between 3 'encoders'.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-scale.c: (make_pipeline), (main):
Add dynamic rescaling tests for the new basetransform.
Original commit message from CVS:
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
* tests/check/pipelines/streamheader.c: (streamheader_suite):
Enable unit tests on PPC again as the bugs are now fixed.
Original commit message from CVS:
* tests/examples/seek/Makefile.am:
* tests/examples/seek/seek.c:
Remove libgstvideo usage. Use gtk_get_option_group instead of
gtk_init().
Original commit message from CVS:
* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
(gst_ximagesink_setcaps):
* sys/ximage/ximagesink.h:
When the caps change, make sure to re-draw borders in
force-aspect-ratio=true mode.
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
Don't clear the border_draw flag until we actually draw the border.
* tests/check/Makefile.am:
Ignore alsasink/src during the states test too, so it doesn't fail
when running without access to the sound device.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Free and clear the seek element list so that we don't use invalid
references when seeking after recreating a gst-launch line.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Original commit message from CVS:
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
Do not try to run a test which requires vorbisenc unless we have
actually built it.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
Original commit message from CVS:
* ext/pango/Makefile.am:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
Use gstvideo functions to calculate strides and plane offsets. Fixes
rendering issue ('ghost' images of the text on the chroma planes)
with widths or heights that are not multiples of 8 (#506659 and
probably also #485729).
* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
(main):
Test with odd height/width too.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
Original commit message from CVS:
* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
Remove wrong_channels_identification_header unit test as we now
support 7 (and more channels).
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes#533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_new_caps),
(gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
(gst_video_format_get_pixel_stride),
(gst_video_format_get_component_width),
(gst_video_format_get_component_height),
(gst_video_format_get_component_offset), (gst_video_format_get_size),
(gst_video_format_convert):
Add guards to these functions to ensure sane input values.
* tests/check/libs/video.c:
Fix unit test not to create caps with width=0 and height=0.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
Original commit message from CVS:
* tests/examples/seek/seek.c: (update_fill), (set_update_fill),
(play_cb), (pause_cb), (stop_cb), (msg_state_changed),
(msg_buffering), (main):
Add basic download reports to seek using the new buffering API.
Original commit message from CVS:
* tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
(msg_buffering), (connect_bus_signals), (main):
Add statusbar.
Add buffering support with feedback in the statusbar.
Original commit message from CVS:
* tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (main):
Rearrange some buttons in playbin2 and make some other boxes insensitive
when needed.
Add language codes to subtitle selection boxes when we gind the right
tags for the streams.
Original commit message from CVS:
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
(gst_video_format_from_rgba32_masks):
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
(#522635).
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
Add unit test for the RGB caps parsing and creation, checking for
internal consistency of the new API and consistency of the API with
the old GST_VIDEO_CAPS_* defines.
Original commit message from CVS:
* tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
(fourcc_list_struct), (fourcc_list), (fourcc_get_size),
(paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
(paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
(paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
(paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
(gst_video_format_is_packed), (video_format_is_packed):
Add unit test that makes sure that the strides, offsets and
sizes returned for the various YUV formats by the new video API
match the old reference implementation in videotestsrc.
Original commit message from CVS:
* tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
(init_visualization_features), (vis_combo_cb), (shot_cb), (main):
Add combo box for visualisations, populate it with a factory list
of all visualisation plugins, configure vis plugin instance in
playbin2.
Original commit message from CVS:
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
(subparse_suite):
Forward slashes at the beginning and end of a line also signify
italics (Fixes: #518162).
Original commit message from CVS:
* tests/check/gst-plugins-base.supp:
Add a suppression for a cached value in GIO that wasn't moved
while moving gio from -bad to -base.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
(create_element), (gst_play_frame_conv_convert):
* gst/playback/gstscreenshot.h:
Add screenshot conversion code from totem.
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
(gst_play_bin_class_init), (gst_play_bin_convert_frame),
(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
Implement frame property to get a color-unconverted snapshot.
Implement convert-frame action signal to get a converted snapshot image.
Configure connection speed in uridecodebin.
Document some more properties.
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_get_last_frame):
* gst/playback/gstplaysink.h:
Use last-buffer property of the video sink to get a video snapshot.
* tests/examples/seek/seek.c: (shot_cb), (main):
Add snapshot button for playbin2 and use the frame property to save the
frame as a png in the current directory.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
(no_more_pads_cb):
Get the object data correct so that we can remove our channels
correctly.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Add option to disable async behaviour in the sinks when possible. This
makes it possible to avoid an audio queue when dealing with
visualisations.
Add option to add a queue for the audio path.
* tests/examples/seek/seek.c: (clear_streams), (update_streams),
(main):
Disable the vis checkbox to match the defaults of playbin2.
Only get the stream info when we need to.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
(gen_vis_chain), (gst_play_sink_reconfigure),
(gst_play_sink_request_pad):
Move tee in front of the audio and vis pipelines.
Add queue for audio for now.
Add visualisation support.
* tests/examples/seek/seek.c: (main):
Visualisation is by default disabled.
Original commit message from CVS:
2008-02-12 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
some seek flags when changing rate.
Original commit message from CVS:
* ext/gio/gstgiostreamsink.c:
* ext/gio/gstgiostreamsrc.c:
Add documentation and example code for giostreamsink/giostreamsrc.
* tests/check/pipelines/gio.c: (GST_START_TEST):
Ask the GMemoryOutputStream for the data instead of assuming that
the pointer to the data stayed the same. It could've been realloc'ed.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb), (clear_streams),
(update_streams), (video_combo_cb), (audio_combo_cb),
(text_combo_cb), (volume_spinbutton_changed_cb), (main):
Add some stream switching and volume gui for playbin2.
Original commit message from CVS:
2008-02-08 Julien Moutte <julien@fluendo.com>
* tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
(main): Make sure bus signals are reconnected when pressing STOP
and then PLAY again for a parse launch pipeline. Fix a ref leak
on the bus.
* win32/common/config.h: Updated.
Original commit message from CVS:
* tests/examples/seek/seek.c: (do_seek),
(rate_spinbutton_changed_cb), (update_streams), (main):
Remove obsolete stream_time reset after flushing seek, core does that
automatically now.
Improve accuracy of speed spinbutton.
Only do playbin2 stuff when we actually use it.
Original commit message from CVS:
* tests/check/Makefile.am:
Revert previous change of the test environment's GST_PLUGIN_PATH.
The problem is not with the plugins, but with element factories
and only occurs if elements are split out from existing plugins
or if plugins change name (see #512740).
Original commit message from CVS:
* tests/check/Makefile.am:
Fix the tests environment's GST_PLUGIN_PATH: we want the directory
with the core's plugins first and our local build directories last,
since we might be building against an installed core, and that
core's plugin directory may contain older or other versions of
our own -base plugins, but we really do want to test our local
ones (if there are multiple plugins or element factories with the
same name, those inspected last will trump those read in earlier).
Fixes#512740 for the most part.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(get_group), (get_n_pads), (gst_play_bin_get_property),
(pad_added_cb), (no_more_pads_cb), (perform_eos),
(autoplug_select_cb), (deactivate_group):
Remove stream-info, we going for something easier.
Refactor getting the current group.
Implement getting the number of audio/video/text streams.
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init), (gst_stream_selector_init),
(gst_stream_selector_get_property),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Add property for number of pads.
* tests/examples/seek/seek.c: (set_scale), (update_flag),
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
(text_toggle_cb), (update_streams), (msg_async_done),
(msg_state_changed), (main):
Block slider callback when updating the slider position.
Add gui elements for controlling playbin2.
Add callback for async_done that updates position/duration.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data):
* gst-libs/gst/rtp/gstrtpbuffer.h:
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add gst_rtp_buffer_set_extension_data()
Add a unit test for this addition. Fixes#511478.
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Update for API changes in GIO and require GIO 2.15.2 for this.
Original commit message from CVS:
* tests/check/Makefile.am:
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
* tests/check/elements/playbin.c:
* tests/check/libs/mixer.c: (test_element_interface_supported),
(gst_implements_interface_init):
* tests/check/libs/rtp.c: (GST_START_TEST):
Fix various assignment type mismatches.
Original commit message from CVS:
* ext/theora/theoradec.c: (gst_theora_dec_class_init),
(_theora_granule_frame), (_theora_granule_start_time),
(theora_dec_sink_convert), (theora_dec_decode_buffer):
Adapt for post-alpha meaning of granulepos, when we
have a newer version of libtheora.
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
(theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
(theora_enc_is_discontinuous), (theora_enc_chain):
Likewise.
* tests/check/Makefile.am:
Link libtheora into theoraenc test so we can check which version of
libtheora we're testing against.
* tests/check/pipelines/theoraenc.c: (check_libtheora),
(check_buffer_granulepos),
(check_buffer_granulepos_from_starttime), (GST_START_TEST),
(theoraenc_suite):
Adapt tests to check the values that are now defined for theora; make
the tests backwards-adapt the passed values if we're running against an
old libtheora.
Fixes#497964
Original commit message from CVS:
* configure.ac:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
* tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
Update to GMemoryInputStream API changes in GLib SVN and require
gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
We can also report the duration for every GSeekable, not only
GFileInputStream and GMemoryInputStream.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
(check_buffer_timestamp), (check_buffer_duration):
Turn these functions into macros so we can see right away
where the failure occured.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c:
Reenable theoraenc test, which fails on the buildbot but
not locally.
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
Disable theoraenc test long enough to get the buildbot to
compile a recent -base.
Original commit message from CVS:
* tests/examples/seek/seek.c: (stop_cb):
Make sure we reset the slider value to 0.0 without racing against a
possible g_idle that sets it to something else.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
(gst_video_scale_transform):
Don't claim to be able to handle/transform caps that can't really
be handled by the currently selected scaling method (here: RGB or
packed YUV with 4-tap method). Also add locking to method property.
* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
(test_basetransform_based):
Some test pipelines for the above (not entirely valgrind clean yet
apparently).
Original commit message from CVS:
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
(test_suburi_error_wrongproto), (test_missing_primary_decoder):
* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
(cddabasesrc_suite):
Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
deprecated in the future (see #498924).
Original commit message from CVS:
* tests/examples/seek/seek.c: (msg_segment_done), (main):
Don't go to READY on EOS as this avoids testing of seeking and
restarting after EOS, use the stop button when you want to READY.
Don't try to do a flushing seek in segment-done, it does not make
sense to use this for gapless playback and is not needed.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c: (formats),
(format_info_get_desc):
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
(GST_START_TEST):
Add one or two missing formats. Generate ADPCM description
dynamically depending on layout/format.
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes#502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
Original commit message from CVS:
2007-11-24 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Increase the range of the
rate selector as I would like to test QOS behavior at higher
forward and reverse playback speed like say 64x.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): There's a nice macro to
check
GTK version, use it.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Try to support stable version
of GTK.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes#492098.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/snapshot/.cvsignore:
* tests/examples/snapshot/Makefile.am:
* tests/examples/snapshot/snapshot.c: (main):
Add simple snapshot example program using appsink.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
Original commit message from CVS:
* configure.ac:
* tests/examples/Makefile.am:
* tests/examples/dynamic/.cvsignore:
* tests/examples/dynamic/Makefile.am:
* tests/examples/dynamic/addstream.c: (create_stream),
(pause_play_stream), (message_received), (eos_message_received),
(perform_step), (main):
Add simple exmple app to demonstrate starting and pausing live and
non-live bins in a PLAYING pipeline.
Original commit message from CVS:
* tests/check/elements/libvisual.c:
Test all libvisual plugins, not just the first one; this reproduces
bug #450336 quite easily. Looks like a problem with the 'jess'
visualisation.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/libvisual.c:
Add basic libvisual test case in an attempt to reproduce bug #450336.
Doesn't reproduce that bug, but some other crasher instead (invalid
free), at least with make elements/libvisual.forever and the bumscope
plugin on x86-64/gutsy. Leaving test disabled for now.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func):
* tests/check/elements/volume.c: (GST_START_TEST):
Revert the latest change: floating point samples are allowed to
have any value, not only values in the range [-1,1]. Thanks to Andy
Wingo for noticing.
Also fix processing of int32 samples with volumes > 4 by making the
unity value smaller which prevents overflows.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* tests/check/libs/rtp.c:
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
Original commit message from CVS:
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst-libs/gst/rtp/gstrtpbuffer.c:
Fix up GstRTPHeader helper struct so that compilers will not under
any circumstances add padding in between our fields, as currently
happens with MSVC on win32, because that would lead to us sending
out RTP payloads with broken RTP headers (#471194).
Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
* tests/check/Makefile.am:
* tests/check/libs/.cvsignore:
* tests/check/libs/rtp.c:
Add some simple unit tests for GstRTPBuffer. Some are disabled
because the code tested still needs fixing (set_csrc() does not work).
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_choose_func),
(volume_process_double), (volume_process_double_clamp),
(volume_process_float_clamp):
Correctly clamp float/double samples in the [-1.0,1.0] range to
prevent weird effects.
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Add unit tests for all samples types that had none before.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/install-plugins.c:
* gst-libs/gst/pbutils/install-plugins.h:
* tests/check/libs/pbutils.c:
API: also add gst_install_plugins_supported() while we're at it
(see #470456).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/missing-plugins.c:
* gst-libs/gst/pbutils/missing-plugins.h:
* tests/check/libs/pbutils.c:
API: add gst_missing_*_installer_detail_new() convenience API so
that applications that know exactly what they're missing can request
installer detail strings for those items directly instead of having
to first create a dummy missing-plugin message and then get the
installer detail string from that. Fixes#470456.
