mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-12 19:36:38 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/tests/examples/Makefile.am

68 lines
1.2 KiB
Makefile
Raw Normal View History

UVC H264 plugin
2012-09-10 20:09:26 +00:00
if USE_UVCH264
UVCH264_DIR=uvch264
else
UVCH264_DIR=
endif
examples/Makefile.am: Only built scaletempo example if we have gtk. Original commit message from CVS: * examples/Makefile.am: Only built scaletempo example if we have gtk.
2008-09-15 09:34:14 +00:00
if HAVE_GTK
scaletempo: remove scaletempo plugin, moved to -good https://bugzilla.gnome.org/show_bug.cgi?id=687262
2012-12-14 14:51:19 +00:00
GTK_EXAMPLES=mxf camerabin2 $(UVCH264_DIR)
examples/Makefile.am: Only built scaletempo example if we have gtk. Original commit message from CVS: * examples/Makefile.am: Only built scaletempo example if we have gtk.
2008-09-15 09:34:14 +00:00
else
GTK_EXAMPLES=
endif
hrm Original commit message from CVS: hrm
2002-01-07 02:42:44 +00:00
Enable documentation. Original commit message from CVS: 2005-11-06 Julien MOUTTE <julien@moutte.net> * Makefile.am: * autogen.sh: * configure.ac: * docs/Makefile.am: * docs/plugins/.cvsignore: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-decl-list.txt: * docs/plugins/gst-plugins-bad-plugins-decl.txt: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins-undocumented.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/gst-plugins-bad-plugins.types: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/version.entities.in: Enable documentation. * examples/Makefile.am: * examples/directfb/Makefile.am: * examples/directfb/decker.ttf: * examples/directfb/dfblogo.png: * examples/directfb/gstdfb.c: (myclock), (dynamic_link), (size_changed), (setup_dynamic_link), (main): Add an example application for DirectFB.
2005-11-06 21:55:01 +00:00
if USE_DIRECTFB
DIRECTFB_DIR=directfb
configure.ac: Add sunaudio Original commit message from CVS: * configure.ac: Add sunaudio * examples/Makefile.am: make gstplay depend on gconf * gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette), (convert_table_lookup), (img_convert): remove c99-isms * gst/ffmpegcolorspace/imgconvert_template.h: make a constant unsigned, to fix a warning on Solaris * gst/mpeg1sys/systems.c: bcopy->memcpy * gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy * sys/Makefile.am: Add sunaudio
2004-05-19 06:10:49 +00:00
else
Enable documentation. Original commit message from CVS: 2005-11-06 Julien MOUTTE <julien@moutte.net> * Makefile.am: * autogen.sh: * configure.ac: * docs/Makefile.am: * docs/plugins/.cvsignore: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-decl-list.txt: * docs/plugins/gst-plugins-bad-plugins-decl.txt: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins-undocumented.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/gst-plugins-bad-plugins.types: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/version.entities.in: Enable documentation. * examples/Makefile.am: * examples/directfb/Makefile.am: * examples/directfb/decker.ttf: * examples/directfb/dfblogo.png: * examples/directfb/gstdfb.c: (myclock), (dynamic_link), (size_changed), (setup_dynamic_link), (main): Add an example application for DirectFB.
2005-11-06 21:55:01 +00:00
DIRECTFB_DIR=
configure.ac: Add sunaudio Original commit message from CVS: * configure.ac: Add sunaudio * examples/Makefile.am: make gstplay depend on gconf * gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette), (convert_table_lookup), (img_convert): remove c99-isms * gst/ffmpegcolorspace/imgconvert_template.h: make a constant unsigned, to fix a warning on Solaris * gst/mpeg1sys/systems.c: bcopy->memcpy * gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy * sys/Makefile.am: Add sunaudio
2004-05-19 06:10:49 +00:00
endif
gstmotioncells_dynamic_test: test tool what can to do dynamic change properties
2011-07-27 20:34:23 +00:00
OPENCV_EXAMPLES=opencv
audiomixmatrix: New element that mixes audio channels This element transforms a given number of input channels into a given number of output channels according to a given transformation matrix. The matrix coefficients must be between -1 and 1. In the auto mode, input/output channels are automatically negotiated and the transformation matrix is a truncated or zero-padded identity matrix. https://bugzilla.gnome.org/show_bug.cgi?id=777376
2017-01-16 17:36:07 +00:00
MATRIXMIX_DIR=audiomixmatrix
applemedia: new AVSampleBufferLayerSink Renders buffers using the CALayer subclass AVSampleBufferDisplayLayer which can be placed inside a Core Animation render tree.
2015-01-19 01:43:23 +00:00
if HAVE_AVFOUNDATION
if HAVE_IOS
AVSAMPLE_DIR=
else
AVSAMPLE_DIR=avsamplesink
endif
else
AVSAMPLE_DIR=
endif
tests/examples: add a waylandsink example https://bugzilla.gnome.org/show_bug.cgi?id=748322
2015-10-26 15:24:40 +00:00
if USE_WAYLAND
waylandsink: Only build if gtk-3.0 was built with wayland target Check in configure if the gtk-3.0 has wayland support, and don't build the waylandsink example if it doesn't.
2018-05-05 14:32:59 +00:00
if HAVE_GTK3_WAYLAND
tests/examples: add a waylandsink example https://bugzilla.gnome.org/show_bug.cgi?id=748322
2015-10-26 15:24:40 +00:00
WAYLAND_DIR=waylandsink
else
WAYLAND_DIR=
endif
else
WAYLAND_DIR=
endif
examples: Makefile.am: add ipcpipeline in SUBDIRS if enabled
2017-09-04 13:13:41 +00:00
if USE_IPCPIPELINE
IPCPIPELINE_DIR=ipcpipeline
else
IPCPIPELINE_DIR=
endif
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
if USE_WEBRTC
WEBRTC_DIR=webrtc
else
WEBRTC_DIR=
endif
playout: New example for seamless audio/video playback An example app that takes video URIs as command line arguments and switches between them seamlessly one after the other using compositor and audiomixer. Both audio-video and video-only media files are valid inputs, but mixing files of both types in a single invocation is cumbersome to support, and hence does not work. The example attempts to keep the audio stream moving along perfectly, and duplicates video frames where necessary to cover gaps in the video timestamps using the 'ignore-eos' videoaggregator pad property. Ensuring seamless (and mostly-glitch-free) switching is harder than it sounds, and hence the example contains plenty of pad probes and running time calculations to make things work. The GPtrArray play_queue contains items that are being played back, have been prepared for playback, and will be played back in the future. The queue itself is mutable besides the first two items (playing and prepared). The item that has been prepared should not be edited or removed since it has been prepared in advance to be activated immediately on the current item's EOS. The example also has support for switching to the next item in the queue prematurely; see the --switch-after/-s flag to the application. Note: the output video is hard-coded at 1280x720, and input video is scaled as needed to fit this size. Set OUTPUT_VIDEO_WIDTH/HEIGHT to change this. https://bugzilla.gnome.org/show_bug.cgi?id=748947
2015-01-29 00:56:26 +00:00
noinst_PROGRAMS = playout
playout_SOURCES = playout.c
playout_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
playout_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(GST_LIBS)
Remove compositor plugin which was moved to -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/138
2018-12-27 12:41:05 +00:00
SUBDIRS= codecparsers mpegts $(DIRECTFB_DIR) $(GTK_EXAMPLES) $(OPENCV_EXAMPLES) \
gtk: remove, plugin has moved to -good
2018-02-08 19:11:21 +00:00
$(AVSAMPLE_DIR) $(WAYLAND_DIR) $(MATRIXMIX_DIR) \
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
$(IPCPIPELINE_DIR) $(WEBRTC_DIR)
Remove compositor plugin which was moved to -base https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/138
2018-12-27 12:41:05 +00:00
DIST_SUBDIRS= codecparsers mpegts camerabin2 directfb mxf opencv uvch264 \
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
2017-01-31 09:56:59 +00:00
avsamplesink waylandsink audiomixmatrix ipcpipeline webrtc
selector: move input-selector and output-selector to core https://bugzilla.gnome.org/show_bug.cgi?id=614306
2010-12-31 01:38:02 +00:00
include $(top_srcdir)/common/parallel-subdirs.mak
Reference in a new issue Copy permalink