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Merge branch 'master' into 0.11

Browse source 
Conflicts:
	configure.ac
This commit is contained in:
Wim Taymans 2011-02-28 11:58:05 +01:00
commit 759a3507d7
352 changed files with 20131 additions and 3417 deletions
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2
.gitignore vendored
View file

@ -43,3 +43,5 @@ gst/deinterlace/tvtime.h
tmp-orc.c
*orc.h
/tests/examples/jack/jack_client

3298
ChangeLog
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File diff suppressed because it is too large Load diff

105
NEWS
View file

@ -1,4 +1,107 @@
This is GStreamer Good Plug-ins 0.10.26, "Escapades"
This is GStreamer Good Plug-ins 0.10.27, "Some Kind of Temporal Blend"
Changes since 0.10.26:
* avidemux: add workaround for buggy list size; extract datetime tags
* cacasink: fix masks and strides
* deinterlace: change the default to linear
* deinterlace: avoid infinite loop draining
* deinterlace: rewrite/fix how neighboring scan lines are calculated
* flvdemux: use aac codec-data to adjust samplerate if needed
* flvmux: Fix for nellymoser codecid setting
* icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag
* id3demux: fix parsing of ID3v2.4 genre frames with multiple genres
* imagefreeze: pass along eos if received before buffer arrives
* jpegdec: add "max-errors" property to ignore decoding errors
* jpegdec: avoid infinite loop when resyncing; discard incomplete image
* matroskademux: add stream-format and alignment properties for h264
* matroskademux: assume matroska if no doctype is specified
* matroskademux: increase allowed max. block size for push mode from 10M to 15M
* matroskademux: normalize empty Cues to no Cues
* matroskamux: add support for DTS and E-AC3 audio
* matroskamux: try to write timestamps in all the outgoing buffers
* multifilesink: send stream headers in key-frame mode
* multiudpsink: add buffer-size property
* navseek: add basic support to change playback rate
* pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS
* pulsesink: flush remaining buffered samples on EOS
* pulsesink: make corking during pause synchronous; don't uncork in _start
* pulsesink: Uncork stream while flushing the ringbuffer
* pulsesrc: add "client" property
* qtdemux: add support for fragmented mp4
* qtdemux: add support for (E)AC-3, WMA and VC-1 audio
* qtdemux: allow pulling atoms with unknown size
* qtdemux: fix flow return aggregation and handling of near end-of-file corner cases
* qtdemux: parse and use creation time tag from mvhd
* rtpbin: copy buffering stats
* rtpbin: correctly calculate RTCP packet size
* rtp: fix rank of payloaders and depayloaders
* rtp: flush state on flush-stop for seek handling for many (de)payloaders
* rtp ac3pay: add AC3 payloader
* rtp h264depay: determine output h264 layout using caps negotiation
* rtp h264pay: implement full bytestream scan mode
* rtp j2kdepay: add support for buffer lists; make depayloader more resilient
* rtp j2kpay: use buffer lists for better performance
* rtp j2kpay: handle EOC correctly; stop scanning when we reached the end
* rtp j2kpay: use SOP markers to split bitstream
* rtp jitterbuffer: provide a clock; get better buffering level
* rtp jpegdepay: fix framerate parsing for locales that use a comma as floating point
* rtp mp4adepay: improve timestamps on outgoing packets
* rtpsession: also emit RTCP activity on SR
* rtpsession: remember last sent RB values
* rtspsrc: add and use auto buffering mode
* rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response
* rtspsrc: include range request for all streams with non-aggregate control
* rtspsrc: increase udp buffer size
* rtspsrc: reset session manager base time when flushing
* rtspsrc: select multicast transports in a smarter way
* souphttpsrc: don't send seeks behind the end of file to the server
* v4l2sink: add navigation support; properties to control crop
* vrawdepay: fix length check
* wavparse: detect DTS advertised as PCM correctly in some more cases
* ximagesrc: change from XGetImage to XGetSubImage dependant on a property
Bugs fixed since 0.10.26:
* 596321 : qtdemux: add support for fragmented MP4 and " mfra " boxes
* 618389 : [pulsemixer] Should implement MIXER_FLAG_AUTO_NOTIFICATIONS interface
* 618652 : [effectv] Use of uninitialised value in unit test
* 620283 : Support for Adobe's F4F missing
* 621929 : [PLUGIN-MOVE] move jack plugin from -bad to -good
* 623178 : [matroskademux] error message for unrecognised FOURCC codes should be improved
* 625825 : cannot link rtpmp4adepay ! aacparse
* 629418 : progressreport: add support for determining stream position from buffer timestamps instead of using queries
* 631516 : [navseek] Add support to change playback rate
* 632654 : [matroskamux] try to write timestamps in most of the outgoing buffers
* 632897 : flvmux does not set the correct nellymoser codec id
* 633280 : [icydemux][PATCH] icydemux: Send 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag
* 634314 : pngdec hangs on faulty pngs
* 634391 : [v4l2src] add interlaced field to caps
* 634393 : v4l2src: Set top field first for interlaced captures
* 634910 : [rtph264pay] Implement bytestream scan mode
* 634928 : [qtdemux] report creation/modification time via metadata tag
* 635734 : jpegdec: infinite loop when playing back motion jpeg stream
* 636049 : ximagesrc: fix remote X and off by ones
* 636172 : imagefreeze: eos is not passed before a buffer arrives
* 636234 : [wavparse] dts 6ch played as stereo 16 bit pcm if DTS frame starts at non-zero offset
* 636621 : flvdemux: doesn't set the right sample rate for aac audio
* 636784 : [qtdemux] GST_QUERY_CONVERT implementation for qtdemux
* 637060 : matroskademux: errors out on 13MB blocks when streaming
* 637686 : [jpegenc] Improve sinkpad getcaps results
* 638019 : [matroskademux] some matroska files are not specifying DocType
* 638072 : build failure: rtpsource.c: error: 'have_rb' may be used uninitialized in this function
* 638535 : id3demux: multiple genres as per ID3v2.4 not supported correctly
* 638569 : cacasink crashes when given 15-bit video.
* 639240 : pulsesink: PLAYING- > PAUSED- > PLAYING transition causes dropout
* 639321 : deinterlace: field{1,3} scanline pointers seem to be off by one field line
* 639339 : v4l2: fails to build with older kernels due to missing V4L_FIELD_INTERLACED_{TB,BT}
* 639516 : muxers: fix setting src pad caps
* 639740 : [pulsesink] doesn't uncork in some cases during reverse playback
* 640028 : [qtdemux] crash on malformed mov stream
* 640063 : rtph264depay: leaks codec data buffer in byte-stream=false mode
* 640064 : rtspsrc memory leak
* 640080 : rtspsrc: fails to error out properly on network failure
* 623063 : [jpegdec] add " max-errors " property
Changes since 0.10.25:

255
RELEASE
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@ -1,5 +1,5 @@
Release notes for GStreamer Good Plug-ins 0.10.26 "Escapades"
Release notes for GStreamer Good Plug-ins 0.10.27 "Some Kind of Temporal Blend"
@ -9,8 +9,6 @@ GStreamer Good Plug-ins.
The 0.10.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.8.x series.
It is, however, parallel installable with the 0.8.x series.
@ -54,125 +52,106 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release
* alphacolor: make passthrough work
* avidemux: reverse playback fixes; prevent overlap of subsequent fragments
* deinterlace: remove assembly code in favor of orc
* dvdemux: parse SMPTE time codes
* flvdemux: parse and use cts (fixes jittery H.264 playback in some cases)
* flvmux: resend onMetada tag when tags changes in streamable mode
* g729pay: extend from right parent
* gconf: Don't install schemas when GConf is disabled
* goom, goom2k1: add latency compensation code, report latency correctly
* gstrtpjpegpay: Added Define Restart Interval (DRI) Marker
* h264depay: always mark the codec_data as keyframe
* icydemux: forward tag events
* id3v2mux: Add mapping for album artist
* imagefreeze: generate a perfectly timestamped stream
* level: avoid division by zero on silence
* matroskademux: more robustness for parse errors and corner-cases
* matroskademux: extract H.264 profile and level and set on caps
* matroskamux: reduce newsegment event spam and set discont flag where needed
* pulse: allow setting of pulse stream properties
* pulse: fix device_description in READY
* pulsesink: Add "client" property to set the PA client name
* pulsesink: share the PA context between all clients with the same name
* qtdemux: export AAC/MPEG-4/H.264 profile and level in caps
* rtp: add G722 payloader and depayloader elements
* rtpamr(de)pay: support AMR-WB SID frame
* rtpamrpay: proper duration for multiple frame payload; properly support perfect-rtptime
* rtpbin: add "ntp-sync" property and "use-pipeline-clock" properties
* rtpg729pay: properly support perfect-rtptime
* rtph264depay: only set delta unit on all-non-key units
* rtpmanager: provide additional statistics
* rtpmp4adepay: grab the sampling rate and put into caps
* rtpmparobustdepay: properly insert dummy buffers; use valid bitrate for dummy frame
* rtpmpvpay: fix timestamping of rtp buffers
* rtpsession: Add the option to auto-discover the RTP bandwidth
* rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth
* rtpsession: Count sent RTCP packets after they have been finished
* rtpsession: relax third-party collision detection
* rtpstats: Rectify description of current_time in RTPArrivalStats
* rtspext: stop configuration on first failure
* rtspsrc: Add property to configure udpsrc buffer size
* rtspsrc: add rtsp-sdp protocol support
* rtspsrc: don't add /UDP in the transport, it's the default
* rtspsrc: fix duration reporting
* rtspsrc: handle stale digest authentication session data
* rtspsrc: use sdp uri parse method
* shapewipe: add optional border parameter and slowdown animation
* shapewipe: Force format to AYUV in the example pipeline for the same reason
* shapewipe: Force the input to AYUV to prevent negotiation failures in videomixer
* spectrum: only aggregate magnitude/phase if user asks for it, performance fixes
* v4l2src: add controllable colorbalance parameters, add decimate property
* v4l2src: fix using mpegts via the mmap interface; use GstBaseSrc::block-size as fallback size
* videomixer2: new videomixer2 element that behaves better than videomixer
* vrawdepay: handle invalid payload better
* avidemux: add workaround for buggy list size; extract datetime tags
* cacasink: fix masks and strides
* deinterlace: change the default to linear
* deinterlace: avoid infinite loop draining
* deinterlace: rewrite/fix how neighboring scan lines are calculated
* flvdemux: use aac codec-data to adjust samplerate if needed
* flvmux: Fix for nellymoser codecid setting
* icydemux: Add 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag
* id3demux: fix parsing of ID3v2.4 genre frames with multiple genres
* imagefreeze: pass along eos if received before buffer arrives
* jpegdec: add "max-errors" property to ignore decoding errors
* jpegdec: avoid infinite loop when resyncing; discard incomplete image
* matroskademux: add stream-format and alignment properties for h264
* matroskademux: assume matroska if no doctype is specified
* matroskademux: increase allowed max. block size for push mode from 10M to 15M
* matroskademux: normalize empty Cues to no Cues
* matroskamux: add support for DTS and E-AC3 audio
* matroskamux: try to write timestamps in all the outgoing buffers
* multifilesink: send stream headers in key-frame mode
* multiudpsink: add buffer-size property
* navseek: add basic support to change playback rate
* pulsemixer: Implement MIXER_FLAG_AUTO_NOTIFICATIONS
* pulsesink: flush remaining buffered samples on EOS
* pulsesink: make corking during pause synchronous; don't uncork in _start
* pulsesink: Uncork stream while flushing the ringbuffer
* pulsesrc: add "client" property
* qtdemux: add support for fragmented mp4
* qtdemux: add support for (E)AC-3, WMA and VC-1 audio
* qtdemux: allow pulling atoms with unknown size
* qtdemux: fix flow return aggregation and handling of near end-of-file corner cases
* qtdemux: parse and use creation time tag from mvhd
* rtpbin: copy buffering stats
* rtpbin: correctly calculate RTCP packet size
* rtp: fix rank of payloaders and depayloaders
* rtp: flush state on flush-stop for seek handling for many (de)payloaders
* rtp ac3pay: add AC3 payloader
* rtp h264depay: determine output h264 layout using caps negotiation
* rtp h264pay: implement full bytestream scan mode
* rtp j2kdepay: add support for buffer lists; make depayloader more resilient
* rtp j2kpay: use buffer lists for better performance
* rtp j2kpay: handle EOC correctly; stop scanning when we reached the end
* rtp j2kpay: use SOP markers to split bitstream
* rtp jitterbuffer: provide a clock; get better buffering level
* rtp jpegdepay: fix framerate parsing for locales that use a comma as floating point
* rtp mp4adepay: improve timestamps on outgoing packets
* rtpsession: also emit RTCP activity on SR
* rtpsession: remember last sent RB values
* rtspsrc: add and use auto buffering mode
* rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response
* rtspsrc: include range request for all streams with non-aggregate control
* rtspsrc: increase udp buffer size
* rtspsrc: reset session manager base time when flushing
* rtspsrc: select multicast transports in a smarter way
* souphttpsrc: don't send seeks behind the end of file to the server
* v4l2sink: add navigation support; properties to control crop
* vrawdepay: fix length check
* wavparse: detect DTS advertised as PCM correctly in some more cases
* ximagesrc: change from XGetImage to XGetSubImage dependant on a property
Bugs fixed in this release
* 596321 : qtdemux: add support for fragmented MP4 and " mfra " boxes
* 618389 : [pulsemixer] Should implement MIXER_FLAG_AUTO_NOTIFICATIONS interface
* 618652 : [effectv] Use of uninitialised value in unit test
* 620283 : Support for Adobe's F4F missing
* 621929 : [PLUGIN-MOVE] move jack plugin from -bad to -good
* 623178 : [matroskademux] error message for unrecognised FOURCC codes should be improved
* 625825 : cannot link rtpmp4adepay ! aacparse
* 629047 : segfault in seek matroskademux
* 537544 : [pulse] allow setting pa context properties
* 628996 : pulsesink broken after shared context patch (bug #624338)
* 529672 : Big latency and bad framerate while mixing multiple live streams
* 581294 : rtspext: extensions configure_stream methods conflict
* 598915 : qtdemux: propagate jpeg2000 header data in image/x-j2c
* 612313 : qtdemux: Post AAC profile/level in caps
* 616521 : qtdemux: Export MPEG-4 video profile and level in stream caps
* 617318 : matroskademux, qtdemux: Use pbutils for H.264 profile/level extraction
* 620790 : [matroskademux] general stream error when trying to play certain .mkv file
* 622390 : [v4l2] add controllable color balance properties / programmable camera
* 624338 : [pulsesink] Handle pulse context separate from the ringbuffers and share them
* 625547 : imagefreeze unit test fails occasionally
* 626048 : [videomixer] needs mode that syncs streams based on timestamps
* 626518 : [imagefreeze] better caps negotiation
* 627162 : [pulse] better fallback return value for gst_pulse_client_name()
* 627174 : [pulsesink] new property to tune the PA client name
* 627289 : souphttpsrc: tweak error messages
* 627341 : wavparse: strange handling of files less than 12 bytes
* 627796 : rtpbin: add ntp clock sync
* 628020 : [pulsesink] assertion failure in change_state NULL- > READY
* 628058 : Need a way to set the SO_RCVBUF property on rtsp-based sockets.
* 628127 : jpeg rtp payloader crashes when there is corruption in the jpeg byte stream.
* 628214 : Add support to RTSP initiation through SDP files
* 628349 : [v4l2src] Doesn't support capturing mpegts using mmap
* 628454 : Matroska demuxer doesn't handle DATE tag if it contains only a year number
* 628608 : [alphacolor] element classification is wrong
* 629018 : rtpjpegpay: unable to build because of uninitialized variable warning
* 629522 : [rtpjpegpay] add support for Define Restart Interval (DRI)
* 629839 : [qtdemux] Update xmp tags parsing
* 629896 : Error compiling raw1394 (without iec61883)
* 630088 : [flvdemux] jerky h.264 video playback
* 630205 : [icydemux] Forward tag events downstrem
* 630256 : rtph264-pay/depay: doesn't respect timestamps from incomming buffers
* 630317 : Getting pulsesink device names doesn't work like for alsasink
* 630378 : speexenc/speexdec crash with MSVC
* 630446 : rtpmanager: provide additional statistics
* 630447 : rtpsession: relax third-party collision detection
* 630449 : rtpbin: Unlock before adding pad in new_payload_found
* 630451 : rtpbin: Handle rysnc of iterator when looking for free pad name
* 630452 : rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe
* 630457 : rtpmanager: packet lost should not be a warning.
* 630458 : level: avoid division by zero on silence
* 630500 : [rtspsrc] does rtsp setup message always need " /UDP " string?
* 630888 : v4l2sink does not cope with v4l2loopback kernel module
* 631082 : rtpjitterbuffer: improve document reference
* 631303 : [goom] qos warnings if source is GstAudioSrc
* 631330 : [flvmux][PATCH] Resend updated onMetada tag when tags changes in streamable mode
* 631996 : [h264depay] regression: rtsp://stream.zoovision.com/KibaEp1n900.3gp
* 632548 : [rtspsrc] regression; fails to report duration
* 632553 : --disable-gconf still tries to install schemas
* 632682 : [matroskademux] Handle missing CodecPrivate for Vorbis/Theora
* 632945 : rtph264depay in access-unit=true mode does not aggregate the delta unit flag correctly
* 633205 : Fix for navigation events in videoflip
* 633212 : [goom] Return not-negotiated when bps is unknown
* 633970 : [icydemux] broken taglist handling
* 635532 : rtspsrc: unexpected eos when using authentication (regression)
* 635843 : [rtph264depay] segfault on empty payload
* 636179 : [deinterlace] Fields in wrong order
* 626463 : [matroskademux] " reading large block of size 14688496 not supported "
* 628894 : [matroskademux] sloppy reverse playback
* 633294 : deinterlace breaks some DVD menu scenarios
* 629418 : progressreport: add support for determining stream position from buffer timestamps instead of using queries
* 631516 : [navseek] Add support to change playback rate
* 632654 : [matroskamux] try to write timestamps in most of the outgoing buffers
* 632897 : flvmux does not set the correct nellymoser codec id
* 633280 : [icydemux][PATCH] icydemux: Send 'StreamUrl' metadata as GST_TAG_HOMEPAGE tag
* 634314 : pngdec hangs on faulty pngs
* 634391 : [v4l2src] add interlaced field to caps
* 634393 : v4l2src: Set top field first for interlaced captures
* 634910 : [rtph264pay] Implement bytestream scan mode
* 634928 : [qtdemux] report creation/modification time via metadata tag
* 635734 : jpegdec: infinite loop when playing back motion jpeg stream
* 636049 : ximagesrc: fix remote X and off by ones
* 636172 : imagefreeze: eos is not passed before a buffer arrives
* 636234 : [wavparse] dts 6ch played as stereo 16 bit pcm if DTS frame starts at non-zero offset
* 636621 : flvdemux: doesn't set the right sample rate for aac audio
* 636784 : [qtdemux] GST_QUERY_CONVERT implementation for qtdemux
* 637060 : matroskademux: errors out on 13MB blocks when streaming
* 637686 : [jpegenc] Improve sinkpad getcaps results
* 638019 : [matroskademux] some matroska files are not specifying DocType
* 638072 : build failure: rtpsource.c: error: 'have_rb' may be used uninitialized in this function
* 638535 : id3demux: multiple genres as per ID3v2.4 not supported correctly
* 638569 : cacasink crashes when given 15-bit video.
* 639240 : pulsesink: PLAYING- > PAUSED- > PLAYING transition causes dropout
* 639321 : deinterlace: field{1,3} scanline pointers seem to be off by one field line
* 639339 : v4l2: fails to build with older kernels due to missing V4L_FIELD_INTERLACED_{TB,BT}
* 639516 : muxers: fix setting src pad caps
* 639740 : [pulsesink] doesn't uncork in some cases during reverse playback
* 640028 : [qtdemux] crash on malformed mov stream
* 640063 : rtph264depay: leaks codec data buffer in byte-stream=false mode
* 640064 : rtspsrc memory leak
* 640080 : rtspsrc: fails to error out properly on network failure
* 623063 : [jpegdec] add " max-errors " property
Download
@ -202,34 +181,38 @@ Applications
Contributors to this release
* Alessandro Decina
* American Dynamics
* Andoni Morales Alastruey
* Andy Wingo
* Arun Raghavan
* Bastien Nocera
* Benjamin Gaignard
* Benjamin Otte
* Christian Schaller
* David Hoyt
* David Schleef
* Edward Hervey
* Havard Graff
* IOhannes m zmölnig
* Erich Schubert
* Guillaume Emont
* Iain Holmes
* Jan Schmidt
* Jonathan Matthew
* Janne Grunau
* Johan Dahlin
* Kishore Arepalli
* Leif Johnson
* Marc-André Lureau
* Mark Nauwelaerts
* Olivier Crête
* Pascal Buhler
* Pavel Kostyuchenko
* Philip Jägenstedt
* Philippe Normand
* René Stadler
* Robert Swain
* Paul Davis
* Rob Clark
* Ronald S. Bultje
* Sebastian Dröge
* Sjoerd Simons
* Stefan Kost
* Steve Baker
* Stéphane Loeuillet
* Tambet Ingo
* Thiago Santos
* Thibault Saunier
* Thijs Vermeir
* Thomas Vander Stichele
* Tim-Philipp Müller
* Trond Andersen
* Vladimir Eremeev
* Tom Janiszewski
* Tristan Matthews
* Vincent Penquerc'h
* Wim Taymans
* Zaheer Abbas Merali
 

2
common

@ -1 +1 @@
Subproject commit 011bcc8a0fc7f798ee874a7ba899123fb2470e22
Subproject commit 1de7f6ab2d4bc1af69f06079cf0f4e2cbbfdc178

15
configure.ac
View file

@ -748,6 +748,14 @@ AG_GST_CHECK_FEATURE(HAL, [HAL libraries], halelements, [
AG_GST_PKG_CHECK_MODULES(HAL, [hal >= 0.5.6, dbus-1 >= 0.32])
])
dnl *** Jack ***
translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true)
AG_GST_CHECK_FEATURE(JACK, Jack, jack, [
PKG_CHECK_MODULES(JACK, jack >= 0.99.10, HAVE_JACK="yes", HAVE_JACK="no")
AC_SUBST(JACK_CFLAGS)
AC_SUBST(JACK_LIBS)
])
dnl *** jpeg ***
dnl FIXME: we could use header checks here as well IMO
translit(dnm, m, l) AM_CONDITIONAL(USE_JPEG, true)
@ -1020,6 +1028,7 @@ AM_CONDITIONAL(USE_GCONFTOOL, false)
AM_CONDITIONAL(USE_GDK_PIXBUF, false)
AM_CONDITIONAL(USE_GST_V4L2, false)
AM_CONDITIONAL(USE_HAL, false)
AM_CONDITIONAL(USE_JACK, false)
AM_CONDITIONAL(USE_JPEG, false)
AM_CONDITIONAL(USE_LIBCACA, false)
AM_CONDITIONAL(USE_LIBDV, false)
@ -1144,7 +1153,6 @@ gst/wavenc/Makefile
gst/wavparse/Makefile
gst/flx/Makefile
gst/y4m/Makefile
ext/jpeg/Makefile
ext/Makefile
ext/aalib/Makefile
ext/annodex/Makefile
@ -1155,6 +1163,8 @@ ext/flac/Makefile
ext/gconf/Makefile
ext/gdk_pixbuf/Makefile
ext/hal/Makefile
ext/jack/Makefile
ext/jpeg/Makefile
ext/libcaca/Makefile
ext/libpng/Makefile
ext/pulse/Makefile
@ -1180,6 +1190,7 @@ tests/check/Makefile
tests/examples/Makefile
tests/examples/audiofx/Makefile
tests/examples/equalizer/Makefile
tests/examples/jack/Makefile
tests/examples/level/Makefile
tests/examples/pulse/Makefile
tests/examples/rtp/Makefile
@ -1236,7 +1247,7 @@ sed \
-e 's/.* PLUGINDIR$/#ifdef _DEBUG\n# define PLUGINDIR PREFIX "\\\\debug\\\\lib\\\\gstreamer-0.11"\n#else\n# define PLUGINDIR PREFIX "\\\\lib\\\\gstreamer-0.11"\n#endif/' \
-e 's/.* USE_BINARY_REGISTRY$/#define USE_BINARY_REGISTRY/' \
-e 's/.* VERSION$/#define VERSION "'$VERSION'"/' \
-e "s/.* DEFAULT_AUDIOSINK$/#define DEFAULT_AUDIOSINK \"directaudiosink\"/" \
-e "s/.* DEFAULT_AUDIOSINK$/#define DEFAULT_AUDIOSINK \"directsoundsink\"/" \
-e "s/.* DEFAULT_AUDIOSRC$/#define DEFAULT_AUDIOSRC \"audiotestsrc\"/" \
-e "s/.* DEFAULT_VIDEOSRC$/#define DEFAULT_VIDEOSRC \"videotestsrc\"/" \
-e "s/.* DEFAULT_VISUALIZER$/#define DEFAULT_VISUALIZER \"goom\"/" \

2
docs/plugins/Makefile.am
View file

@ -94,6 +94,8 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/gdk_pixbuf/gstgdkpixbufsink.h \
$(top_srcdir)/ext/hal/gsthalaudiosink.h \
$(top_srcdir)/ext/hal/gsthalaudiosrc.h \
$(top_srcdir)/ext/jack/gstjackaudiosrc.h \
$(top_srcdir)/ext/jack/gstjackaudiosink.h \
$(top_srcdir)/ext/jpeg/gstjpegdec.h \
$(top_srcdir)/ext/jpeg/gstjpegenc.h \
$(top_srcdir)/ext/jpeg/gstsmokedec.h \

3
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -94,6 +94,8 @@
<xi:include href="xml/element-id3v2mux.xml" />
<xi:include href="xml/element-imagefreeze.xml" />
<xi:include href="xml/element-interleave.xml" />
<xi:include href="xml/element-jackaudiosrc.xml" />
<xi:include href="xml/element-jackaudiosink.xml" />
<xi:include href="xml/element-jpegdec.xml" />
<xi:include href="xml/element-jpegenc.xml" />
<xi:include href="xml/element-level.xml" />
@ -206,6 +208,7 @@
<xi:include href="xml/plugin-id3demux.xml" />
<xi:include href="xml/plugin-imagefreeze.xml" />
<xi:include href="xml/plugin-interleave.xml" />
<xi:include href="xml/plugin-jack.xml" />
<xi:include href="xml/plugin-jpeg.xml" />
<xi:include href="xml/plugin-level.xml" />
<xi:include href="xml/plugin-matroska.xml" />

30
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -1113,6 +1113,36 @@ GstInterleaveFunc
gst_interleave_get_type
</SECTION>
<SECTION>
<FILE>element-jackaudiosrc</FILE>
<TITLE>jackaudiosrc</TITLE>
GstJackAudioSrc
<SUBSECTION Standard>
GstJackAudioSrcClass
GST_JACK_AUDIO_SRC
GST_JACK_AUDIO_SRC_CLASS
GST_JACK_AUDIO_SRC_GET_CLASS
GST_IS_JACK_AUDIO_SRC
GST_IS_JACK_AUDIO_SRC_CLASS
GST_TYPE_JACK_AUDIO_SRC
gst_jack_audio_src_get_type
</SECTION>
<SECTION>
<FILE>element-jackaudiosink</FILE>
<TITLE>jackaudiosink</TITLE>
GstJackAudioSink
<SUBSECTION Standard>
GstJackAudioSinkClass
GST_JACK_AUDIO_SINK
GST_JACK_AUDIO_SINK_CLASS
GST_JACK_AUDIO_SINK_GET_CLASS
GST_IS_JACK_AUDIO_SINK
GST_IS_JACK_AUDIO_SINK_CLASS
GST_TYPE_JACK_AUDIO_SINK
gst_jack_audio_sink_get_type
</SECTION>
<SECTION>
<FILE>element-jpegdec</FILE>
<TITLE>jpegdec</TITLE>

192
docs/plugins/gst-plugins-good-plugins.args
View file

@ -745,7 +745,7 @@
<FLAGS>rw</FLAGS>
<NICK>Buffer Mode</NICK>
<BLURB>Control the buffering algorithm in use.</BLURB>
<DEFAULT>Slave receiver to sender clock</DEFAULT>
<DEFAULT>Choose mode depending on stream live</DEFAULT>
</ARG>
<ARG>
@ -765,7 +765,7 @@
<FLAGS>rw</FLAGS>
<NICK>UDP Buffer Size</NICK>
<BLURB>Size of the kernel UDP receive buffer in bytes, 0=default.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>524288</DEFAULT>
</ARG>
<ARG>
@ -1998,6 +1998,16 @@
<DEFAULT>"auto"</DEFAULT>
</ARG>
<ARG>
<NAME>GstProgressReport::do-query</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Use a query instead of buffer metadata to determine stream position</NICK>
<BLURB>Use a query instead of buffer metadata to determine stream position.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstNavSeek::seek-offset</NAME>
<TYPE>gdouble</TYPE>
@ -2518,6 +2528,16 @@
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiUDPSink::buffer-size</NAME>
<TYPE>gint</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer Size</NICK>
<BLURB>Size of the kernel send buffer in bytes, 0=default.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstCmmlDec::wait-clip-end-time</NAME>
<TYPE>gboolean</TYPE>
@ -2688,6 +2708,16 @@
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstXImageSrc::remote</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Remote dispay</NICK>
<BLURB>Whether the display is remote.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstVideoBalance::brightness</NAME>
<TYPE>gdouble</TYPE>
@ -2768,6 +2798,16 @@
<DEFAULT>Faster, less accurate integer method</DEFAULT>
</ARG>
<ARG>
<NAME>GstJpegDec::max-errors</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum Consecutive Decoding Errors</NICK>
<BLURB>Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPiLBCDepay::mode</NAME>
<TYPE>iLBCMode</TYPE>
@ -19674,7 +19714,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Byte Stream</NICK>
<BLURB>Generate byte stream format of NALU.</BLURB>
<BLURB>Generate byte stream format of NALU (deprecated; use caps).</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
@ -19684,7 +19724,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Access Unit</NICK>
<BLURB>Merge NALU into AU (picture).</BLURB>
<BLURB>Merge NALU into AU (picture) (deprecated; use caps).</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
@ -19838,6 +19878,16 @@
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstPulseSrc::client</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Client</NICK>
<BLURB>The PulseAudio client_name_to_use.</BLURB>
<DEFAULT>"<unknown>"</DEFAULT>
</ARG>
<ARG>
<NAME>GstPulseMixer::device</NAME>
<TYPE>gchar*</TYPE>
@ -20315,7 +20365,7 @@
<FLAGS>rw</FLAGS>
<NICK>Method</NICK>
<BLURB>Deinterlace Method.</BLURB>
<DEFAULT>Motion Adaptive: Advanced Detection</DEFAULT>
<DEFAULT>Television: Full resolution</DEFAULT>
</ARG>
<ARG>
@ -20805,7 +20855,7 @@
<FLAGS>rw</FLAGS>
<NICK>Queue size</NICK>
<BLURB>Number of buffers to be enqueud in the driver in streaming mode.</BLURB>
<DEFAULT>8</DEFAULT>
<DEFAULT>12</DEFAULT>
</ARG>
<ARG>
@ -20848,6 +20898,56 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::crop-height</NAME>
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Crop height</NICK>
<BLURB>The height of the video crop; default is equal to negotiated image height.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::crop-left</NAME>
<TYPE>gint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Crop left</NICK>
<BLURB>The leftmost (x) coordinate of the video crop; top left corner of image is 0,0.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::crop-top</NAME>
<TYPE>gint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Crop top</NICK>
<BLURB>The topmost (y) coordinate of the video crop; top left corner of image is 0,0.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::crop-width</NAME>
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Crop width</NICK>
<BLURB>The width of the video crop; default is equal to negotiated image width.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::min-queued-bufs</NAME>
<TYPE>guint</TYPE>
<RANGE><= 16</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Minimum queued bufs</NICK>
<BLURB>Minimum number of queued bufs; v4l2sink won't dqbuf if the driver doesn't have more than this number (which normally you shouldn't change).</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstShapeWipe::border</NAME>
<TYPE>gfloat</TYPE>
@ -21018,3 +21118,83 @@
<DEFAULT>Checker pattern</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJ2KPay::buffer-list</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer List</NICK>
<BLURB>Use Buffer Lists.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJ2KDepay::buffer-list</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffer List</NICK>
<BLURB>Use Buffer Lists.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::client</NAME>
<TYPE>JackClient*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>JackClient</NICK>
<BLURB>Handle for jack client.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::connect</NAME>
<TYPE>GstJackConnect</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Connect</NICK>
<BLURB>Specify how the input ports will be connected.</BLURB>
<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::server</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Server</NICK>
<BLURB>The Jack server to connect to (NULL = default).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::client</NAME>
<TYPE>JackClient*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>JackClient</NICK>
<BLURB>Handle for jack client.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::connect</NAME>
<TYPE>GstJackConnect</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Connect</NICK>
<BLURB>Specify how the output ports will be connected.</BLURB>
<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::server</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Server</NICK>
<BLURB>The Jack server to connect to (NULL = default).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>

10
docs/plugins/gst-plugins-good-plugins.hierarchy
View file

@ -30,6 +30,7 @@ GObject
GstRtpG723Depay
GstRtpG726Depay
GstRtpG729Depay
GstRtpGSTDepay
GstRtpH263Depay
GstRtpH263PDepay
GstRtpH264Depay
@ -70,8 +71,10 @@ GObject
GstRTPGSMPay
GstRTPMP2TPay
GstRTPMPVPay
GstRtpAC3Pay
GstRtpAMRPay
GstRtpCELTPay
GstRtpGSTPay
GstRtpH263PPay
GstRtpH263Pay
GstRtpH264Pay
@ -92,6 +95,7 @@ GObject
GstEsdSink
GstOss4Sink
GstOssSink
GstJackAudioSink
GstPulseSink
GstCACASink
GstDynUDPSink
@ -110,6 +114,7 @@ GObject
GstOss4Source
GstOssSrc
GstPulseSrc
GstJackAudioSrc
GstDV1394Src
GstHDV1394Src
GstMultiFileSrc
@ -218,6 +223,7 @@ GObject
GstMatroskaDemux
GstMatroskaMux
GstWebMMux
GstMonoscope
GstMuLawDec
GstMuLawEnc
GstMultipartDemux
@ -269,6 +275,8 @@ GObject
GstRingBuffer
GstAudioSinkRingBuffer
GstAudioSrcRingBuffer
GstJackAudioSinkRingBuffer
GstJackAudioSrcRingBuffer
GstTask
GstTaskPool
GstSignalObject
@ -282,6 +290,7 @@ GInterface
GstColorBalance
GstImplementsInterface
GstMixer
GstNavigation
GstPreset
GstPropertyProbe
GstStreamVolume
@ -289,3 +298,4 @@ GInterface
GstTuner
GstURIHandler
GstVideoOrientation
GstXOverlay

2
docs/plugins/gst-plugins-good-plugins.interfaces
View file

@ -19,7 +19,7 @@ GstRgVolume GstChildProxy
GstAspectRatioCrop GstChildProxy
GstPulseSink GstStreamVolume GstImplementsInterface GstPropertyProbe
GstOss4Sink GstStreamVolume GstPropertyProbe
GstV4l2Sink GstImplementsInterface GstColorBalance GstVideoOrientation GstPropertyProbe
GstV4l2Sink GstImplementsInterface GstXOverlay GstNavigation GstColorBalance GstVideoOrientation GstPropertyProbe
GstShout2send GstTagSetter
GstUDPSink GstURIHandler
GstDV1394Src GstURIHandler GstPropertyProbe

1
docs/plugins/gst-plugins-good-plugins.prerequisites
View file

@ -6,4 +6,5 @@ GstMixer GstImplementsInterface GstElement
GstTuner GstImplementsInterface GstElement
GstColorBalance GstImplementsInterface GstElement
GstVideoOrientation GstImplementsInterface GstElement
GstXOverlay GstImplementsInterface GstElement
GIcon GObject

2
docs/plugins/inspect/plugin-1394.xml
View file

@ -3,7 +3,7 @@
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-aasink.xml
View file

@ -3,7 +3,7 @@
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-alaw.xml
View file

@ -3,7 +3,7 @@
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-alpha.xml
View file

@ -3,7 +3,7 @@
<description>adds an alpha channel to video - constant or via chroma-keying</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-alphacolor.xml
View file

@ -3,7 +3,7 @@
<description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-annodex.xml
View file

@ -3,7 +3,7 @@
<description>annodex stream manipulation (info about annodex: http://www.annodex.net)</description>
<filename>../../ext/annodex/.libs/libgstannodex.so</filename>
<basename>libgstannodex.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-apetag.xml
View file

@ -3,7 +3,7 @@
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-audiofx.xml
View file

@ -3,7 +3,7 @@
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-auparse.xml
View file

@ -3,7 +3,7 @@
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-autodetect.xml
View file

@ -3,7 +3,7 @@
<description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
<filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
<basename>libgstautodetect.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-avi.xml
View file

@ -3,7 +3,7 @@
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-cacasink.xml
View file

@ -3,7 +3,7 @@
<description>Colored ASCII Art video sink</description>
<filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
<basename>libgstcacasink.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-cairo.xml
View file

@ -3,7 +3,7 @@
<description>Cairo-based elements</description>
<filename>../../ext/cairo/.libs/libgstcairo.so</filename>
<basename>libgstcairo.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-cutter.xml
View file

@ -3,7 +3,7 @@
<description>Audio Cutter to split audio into non-silent bits</description>
<filename>../../gst/cutter/.libs/libgstcutter.so</filename>
<basename>libgstcutter.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-debug.xml
View file

@ -3,7 +3,7 @@
<description>elements for testing and debugging</description>
<filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
<basename>libgstdebug.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

4
docs/plugins/inspect/plugin-deinterlace.xml
View file

@ -3,7 +3,7 @@
<description>Deinterlacer</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
@ -12,7 +12,7 @@
<element>
<name>deinterlace</name>
<longname>Deinterlacer</longname>
<class>Filter/Video</class>
<class>Filter/Effect/Video/Deinterlace</class>
<description>Deinterlace Methods ported from DScaler/TvTime</description>
<author>Martin Eikermann &lt;meiker@upb.de&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>

2
docs/plugins/inspect/plugin-dv.xml
View file

@ -3,7 +3,7 @@
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-efence.xml
View file

@ -3,7 +3,7 @@
<description>This element converts a stream of normal GStreamer buffers into a stream of buffers that are allocated in such a way that out-of-bounds access to data in the buffer is more likely to cause segmentation faults. This allocation method is very similar to the debugging tool &quot;Electric Fence&quot;.</description>
<filename>../../gst/debugutils/.libs/libgstefence.so</filename>
<basename>libgstefence.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-effectv.xml
View file

@ -3,7 +3,7 @@
<description>effect plugins from the effectv project</description>
<filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
<basename>libgsteffectv.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-equalizer.xml
View file

@ -3,7 +3,7 @@
<description>GStreamer audio equalizers</description>
<filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
<basename>libgstequalizer.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-esdsink.xml
View file

@ -3,7 +3,7 @@
<description>ESD Element Plugins</description>
<filename>../../ext/esd/.libs/libgstesd.so</filename>
<basename>libgstesd.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-flac.xml
View file

@ -3,7 +3,7 @@
<description>The FLAC Lossless compressor Codec</description>
<filename>../../ext/flac/.libs/libgstflac.so</filename>
<basename>libgstflac.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-flv.xml
View file

@ -3,7 +3,7 @@
<description>FLV muxing and demuxing plugin</description>
<filename>../../gst/flv/.libs/libgstflv.so</filename>
<basename>libgstflv.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-flxdec.xml
View file

@ -3,7 +3,7 @@
<description>FLC/FLI/FLX video decoder</description>
<filename>../../gst/flx/.libs/libgstflxdec.so</filename>
<basename>libgstflxdec.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-gconfelements.xml
View file

@ -3,7 +3,7 @@
<description>elements wrapping the GStreamer/GConf audio/video output settings</description>
<filename>../../ext/gconf/.libs/libgstgconfelements.so</filename>
<basename>libgstgconfelements.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-gdkpixbuf.xml
View file

@ -3,7 +3,7 @@
<description>GdkPixbuf-based image decoder, scaler and sink</description>
<filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
<basename>libgstgdkpixbuf.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-goom.xml
View file

@ -3,7 +3,7 @@
<description>GOOM visualization filter</description>
<filename>../../gst/goom/.libs/libgstgoom.so</filename>
<basename>libgstgoom.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-goom2k1.xml
View file

@ -3,7 +3,7 @@
<description>GOOM 2k1 visualization filter</description>
<filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
<basename>libgstgoom2k1.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-gstrtpmanager.xml
View file

@ -3,7 +3,7 @@
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-halelements.xml
View file

@ -3,7 +3,7 @@
<description>elements wrapping the GStreamer/HAL audio input/output devices</description>
<filename>../../ext/hal/.libs/libgsthalelements.so</filename>
<basename>libgsthalelements.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-icydemux.xml
View file

@ -3,7 +3,7 @@
<description>Demux ICY tags from a stream</description>
<filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
<basename>libgsticydemux.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-id3demux.xml
View file

@ -3,7 +3,7 @@
<description>Demux ID3v1 and ID3v2 tags from a file</description>
<filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
<basename>libgstid3demux.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-imagefreeze.xml
View file

@ -3,7 +3,7 @@
<description>Still frame stream generator</description>
<filename>../../gst/imagefreeze/.libs/libgstimagefreeze.so</filename>
<basename>libgstimagefreeze.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-interleave.xml
View file

@ -3,7 +3,7 @@
<description>Audio interleaver/deinterleaver</description>
<filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
<basename>libgstinterleave.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

43
docs/plugins/inspect/plugin-jack.xml Normal file
View file

@ -0,0 +1,43 @@
<plugin>
<name>jack</name>
<description>JACK audio elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>jackaudiosink</name>
<longname>Audio Sink (Jack)</longname>
<class>Sink/Audio</class>
<description>Output audio to a JACK server</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element>
<name>jackaudiosrc</name>
<longname>Audio Source (Jack)</longname>
<class>Source/Audio</class>
<description>Captures audio from a JACK server</description>
<author>Tristan Matthews &lt;tristan@sat.qc.ca&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-float, endianness=(int)1234, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
</elements>
</plugin>

2
docs/plugins/inspect/plugin-jpeg.xml
View file

@ -3,7 +3,7 @@
<description>JPeg plugin library</description>
<filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
<basename>libgstjpeg.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-level.xml
View file

@ -3,7 +3,7 @@
<description>Audio level plugin</description>
<filename>../../gst/level/.libs/libgstlevel.so</filename>
<basename>libgstlevel.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

6
docs/plugins/inspect/plugin-matroska.xml
View file

@ -3,7 +3,7 @@
<description>Matroska and WebM stream handling</description>
<filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
<basename>libgstmatroska.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
@ -53,13 +53,13 @@
<name>audio_%d</name>
<direction>sink</direction>
<presence>request</presence>
<details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], stream-format=(string){ raw }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]</details>
<details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], stream-format=(string){ raw }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>subtitle_%d</name>
<direction>sink</direction>
<presence>request</presence>
<details>ANY</details>
<details>subtitle/x-kate</details>
</caps>
<caps>
<name>video_%d</name>

4
docs/plugins/inspect/plugin-monoscope.xml
View file

@ -3,10 +3,10 @@
<description>Monoscope visualization</description>
<filename>../../gst/monoscope/.libs/libgstmonoscope.so</filename>
<basename>libgstmonoscope.so</basename>
<version>0.10.24.5</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins prerelease</package>
<package>GStreamer Good Plug-ins git</package>
<origin>Unknown package origin</origin>
<elements>
<element>

2
docs/plugins/inspect/plugin-mulaw.xml
View file

@ -3,7 +3,7 @@
<description>MuLaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstmulaw.so</filename>
<basename>libgstmulaw.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-multifile.xml
View file

@ -3,7 +3,7 @@
<description>Reads/Writes buffers from/to sequentially named files</description>
<filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
<basename>libgstmultifile.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-multipart.xml
View file

@ -3,7 +3,7 @@
<description>multipart stream manipulation</description>
<filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
<basename>libgstmultipart.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-navigationtest.xml
View file

@ -3,7 +3,7 @@
<description>Template for a video filter</description>
<filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
<basename>libgstnavigationtest.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-oss4.xml
View file

@ -3,7 +3,7 @@
<description>Open Sound System (OSS) version 4 support for GStreamer</description>
<filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
<basename>libgstoss4audio.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-ossaudio.xml
View file

@ -3,7 +3,7 @@
<description>OSS (Open Sound System) support for GStreamer</description>
<filename>../../sys/oss/.libs/libgstossaudio.so</filename>
<basename>libgstossaudio.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-png.xml
View file

@ -3,7 +3,7 @@
<description>PNG plugin library</description>
<filename>../../ext/libpng/.libs/libgstpng.so</filename>
<basename>libgstpng.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-pulseaudio.xml
View file

@ -3,7 +3,7 @@
<description>PulseAudio plugin library</description>
<filename>../../ext/pulse/.libs/libgstpulse.so</filename>
<basename>libgstpulse.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-quicktime.xml
View file

@ -3,7 +3,7 @@
<description>Quicktime support</description>
<filename>../../gst/qtdemux/.libs/libgstqtdemux.so</filename>
<basename>libgstqtdemux.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-replaygain.xml
View file

@ -3,7 +3,7 @@
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

199
docs/plugins/inspect/plugin-rtp.xml
View file

@ -3,7 +3,7 @@
<description>Real-time protocol plugins</description>
<filename>../../gst/rtp/.libs/libgstrtp.so</filename>
<basename>libgstrtp.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>
@ -12,7 +12,7 @@
<element>
<name>asteriskh263</name>
<longname>RTP Asterisk H263 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts H263 video from RTP and encodes in Asterisk H263 format</description>
<author>Neil Stratford &lt;neils@vipadia.com&gt;</author>
<pads>
@ -33,7 +33,7 @@
<element>
<name>rtpL16depay</name>
<longname>RTP audio depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts raw audio from RTP packets</description>
<author>Zeeshan Ali &lt;zak147@yahoo.com&gt;,Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -54,7 +54,7 @@
<element>
<name>rtpL16pay</name>
<longname>RTP audio payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode Raw audio into RTP packets (RFC 3551)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -75,7 +75,7 @@
<element>
<name>rtpac3depay</name>
<longname>RTP AC3 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts AC3 audio from RTP packets (RFC 4184)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -93,10 +93,31 @@
</caps>
</pads>
</element>
<element>
<name>rtpac3pay</name>
<longname>RTP AC3 audio payloader</longname>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload AC3 audio as RTP packets (RFC 4184)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/ac3; audio/x-ac3</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int){ 32000, 44100, 48000 }, encoding-name=(string)AC3</details>
</caps>
</pads>
</element>
<element>
<name>rtpamrdepay</name>
<longname>RTP AMR depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -117,7 +138,7 @@
<element>
<name>rtpamrpay</name>
<longname>RTP AMR payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -138,7 +159,7 @@
<element>
<name>rtpbvdepay</name>
<longname>RTP BroadcomVoice depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts BroadcomVoice audio from RTP packets (RFC 4298)</description>
<author>Wim Taymans &lt;wim.taymans@collabora.co.uk&gt;</author>
<pads>
@ -159,7 +180,7 @@
<element>
<name>rtpbvpay</name>
<longname>RTP BV Payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)</description>
<author>Wim Taymans &lt;wim.taymans@collabora.co.uk&gt;</author>
<pads>
@ -180,7 +201,7 @@
<element>
<name>rtpceltdepay</name>
<longname>RTP CELT depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts CELT audio from RTP packets</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -201,7 +222,7 @@
<element>
<name>rtpceltpay</name>
<longname>RTP CELT payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes CELT audio into a RTP packet</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -222,7 +243,7 @@
<element>
<name>rtpdepay</name>
<longname>Dummy RTP session manager</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Accepts raw RTP and RTCP packets and sends them forward</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -255,7 +276,7 @@
<element>
<name>rtpdvdepay</name>
<longname>RTP DV Depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Depayloads DV from RTP packets (RFC 3189)</description>
<author>Marcel Moreaux &lt;marcelm@spacelabs.nl&gt;, Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -276,7 +297,7 @@
<element>
<name>rtpdvpay</name>
<longname>RTP DV Payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payloads DV into RTP packets (RFC 3189)</description>
<author>Marcel Moreaux &lt;marcelm@spacelabs.nl&gt;, Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -297,7 +318,7 @@
<element>
<name>rtpg722depay</name>
<longname>RTP audio depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts G722 audio from RTP packets</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -318,7 +339,7 @@
<element>
<name>rtpg722pay</name>
<longname>RTP audio payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode Raw audio into RTP packets (RFC 3551)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -339,7 +360,7 @@
<element>
<name>rtpg723depay</name>
<longname>RTP G.723 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts G.723 audio from RTP packets (RFC 3551)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -360,7 +381,7 @@
<element>
<name>rtpg723pay</name>
<longname>RTP G.723 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Packetize G.723 audio into RTP packets</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -381,7 +402,7 @@
<element>
<name>rtpg726depay</name>
<longname>RTP G.726 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts G.726 audio from RTP packets</description>
<author>Axis Communications &lt;dev-gstreamer@axis.com&gt;</author>
<pads>
@ -402,7 +423,7 @@
<element>
<name>rtpg726pay</name>
<longname>RTP G.726 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes G.726 audio into a RTP packet</description>
<author>Axis Communications &lt;dev-gstreamer@axis.com&gt;</author>
<pads>
@ -423,7 +444,7 @@
<element>
<name>rtpg729depay</name>
<longname>RTP G.729 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts G.729 audio from RTP packets (RFC 3551)</description>
<author>Laurent Glayal &lt;spglegle@yahoo.fr&gt;</author>
<pads>
@ -444,7 +465,7 @@
<element>
<name>rtpg729pay</name>
<longname>RTP G.729 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Packetize G.729 audio into RTP packets</description>
<author>Olivier Crete &lt;olivier.crete@collabora.co.uk&gt;</author>
<pads>
@ -465,7 +486,7 @@
<element>
<name>rtpgsmdepay</name>
<longname>RTP GSM depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts GSM audio from RTP packets</description>
<author>Zeeshan Ali &lt;zeenix@gmail.com&gt;</author>
<pads>
@ -486,7 +507,7 @@
<element>
<name>rtpgsmpay</name>
<longname>RTP GSM payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes GSM audio into a RTP packet</description>
<author>Zeeshan Ali &lt;zeenix@gmail.com&gt;</author>
<pads>
@ -504,10 +525,52 @@
</caps>
</pads>
</element>
<element>
<name>rtpgstdepay</name>
<longname>GStreamer depayloader</longname>
<class>Codec/Depayloader/Network</class>
<description>Extracts GStreamer buffers from RTP packets</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)application, payload=(int)[ 96, 127 ], clock-rate=(int)90000, encoding-name=(string)X-GST</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>ANY</details>
</caps>
</pads>
</element>
<element>
<name>rtpgstpay</name>
<longname>RTP GStreamer payloader</longname>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload GStreamer buffers as RTP packets</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>ANY</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp, media=(string)application, payload=(int)[ 96, 127 ], clock-rate=(int)90000, encoding-name=(string)X-GST</details>
</caps>
</pads>
</element>
<element>
<name>rtph263depay</name>
<longname>RTP H263 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts H263 video from RTP packets (RFC 2190)</description>
<author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;, Edward Hervey &lt;bilboed@bilboed.com&gt;</author>
<pads>
@ -528,7 +591,7 @@
<element>
<name>rtph263pay</name>
<longname>RTP H263 packet payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes H263 video in RTP packets (RFC 2190)</description>
<author>Neil Stratford &lt;neils@vipadia.com&gt;Dejan Sakelsak &lt;dejan.sakelsak@marand.si&gt;</author>
<pads>
@ -549,7 +612,7 @@
<element>
<name>rtph263pdepay</name>
<longname>RTP H263 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts H263/+/++ video from RTP packets (RFC 4629)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -570,7 +633,7 @@
<element>
<name>rtph263ppay</name>
<longname>RTP H263 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes H263/+/++ video in RTP packets (RFC 4629)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -591,7 +654,7 @@
<element>
<name>rtph264depay</name>
<longname>RTP H264 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts H264 video from RTP packets (RFC 3984)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -612,7 +675,7 @@
<element>
<name>rtph264pay</name>
<longname>RTP H264 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode H264 video into RTP packets (RFC 3984)</description>
<author>Laurent Glayal &lt;spglegle@yahoo.fr&gt;</author>
<pads>
@ -633,7 +696,7 @@
<element>
<name>rtpilbcdepay</name>
<longname>RTP iLBC depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts iLBC audio from RTP packets (RFC 3952)</description>
<author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;</author>
<pads>
@ -654,7 +717,7 @@
<element>
<name>rtpilbcpay</name>
<longname>RTP iLBC Payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Packetize iLBC audio streams into RTP packets</description>
<author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;</author>
<pads>
@ -675,7 +738,7 @@
<element>
<name>rtpj2kdepay</name>
<longname>RTP JPEG 2000 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts JPEG 2000 video from RTP packets (RFC 5371)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -696,7 +759,7 @@
<element>
<name>rtpj2kpay</name>
<longname>RTP JPEG 2000 payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes JPEG 2000 pictures into RTP packets (RFC 5371)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -717,7 +780,7 @@
<element>
<name>rtpjpegdepay</name>
<longname>RTP JPEG depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts JPEG video from RTP packets (RFC 2435)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -738,7 +801,7 @@
<element>
<name>rtpjpegpay</name>
<longname>RTP JPEG payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes JPEG pictures into RTP packets (RFC 2435)</description>
<author>Axis Communications &lt;dev-gstreamer@axis.com&gt;</author>
<pads>
@ -759,7 +822,7 @@
<element>
<name>rtpmp1sdepay</name>
<longname>RTP MPEG1 System Stream depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG1 System Streams from RTP packets (RFC 3555)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -780,7 +843,7 @@
<element>
<name>rtpmp2tdepay</name>
<longname>RTP MPEG Transport Stream depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG2 TS from RTP packets (RFC 2250)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;, Thijs Vermeir &lt;thijs.vermeir@barco.com&gt;</author>
<pads>
@ -801,7 +864,7 @@
<element>
<name>rtpmp2tpay</name>
<longname>RTP MPEG2 Transport Stream payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes MPEG2 TS into RTP packets (RFC 2250)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -822,7 +885,7 @@
<element>
<name>rtpmp4adepay</name>
<longname>RTP MPEG4 audio depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG4 audio from RTP packets (RFC 3016)</description>
<author>Nokia Corporation (contact &lt;stefan.kost@nokia.com&gt;), Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -843,7 +906,7 @@
<element>
<name>rtpmp4apay</name>
<longname>RTP MPEG4 audio payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload MPEG4 audio as RTP packets (RFC 3016)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -864,7 +927,7 @@
<element>
<name>rtpmp4gdepay</name>
<longname>RTP MPEG4 ES depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG4 elementary streams from RTP packets (RFC 3640)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -885,7 +948,7 @@
<element>
<name>rtpmp4gpay</name>
<longname>RTP MPEG4 ES payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload MPEG4 elementary streams as RTP packets (RFC 3640)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -906,7 +969,7 @@
<element>
<name>rtpmp4vdepay</name>
<longname>RTP MPEG4 video depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG4 video from RTP packets (RFC 3016)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -927,7 +990,7 @@
<element>
<name>rtpmp4vpay</name>
<longname>RTP MPEG4 Video payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload MPEG-4 video as RTP packets (RFC 3016)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -935,7 +998,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false</details>
<details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; video/x-xvid</details>
</caps>
<caps>
<name>src</name>
@ -948,7 +1011,7 @@
<element>
<name>rtpmpadepay</name>
<longname>RTP MPEG audio depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG audio from RTP packets (RFC 2038)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -969,7 +1032,7 @@
<element>
<name>rtpmpapay</name>
<longname>RTP MPEG audio payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload MPEG audio as RTP packets (RFC 2038)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -990,7 +1053,7 @@
<element>
<name>rtpmparobustdepay</name>
<longname>RTP MPEG audio depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG audio from RTP packets (RFC 5219)</description>
<author>Mark Nauwelaerts &lt;mark.nauwelaerts@collabora.co.uk&gt;</author>
<pads>
@ -1011,7 +1074,7 @@
<element>
<name>rtpmpvdepay</name>
<longname>RTP MPEG video depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts MPEG video from RTP packets (RFC 2250)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1032,7 +1095,7 @@
<element>
<name>rtpmpvpay</name>
<longname>RTP MPEG2 ES video payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes MPEG2 ES into RTP packets (RFC 2250)</description>
<author>Thijs Vermeir &lt;thijsvermeir@gmail.com&gt;</author>
<pads>
@ -1053,7 +1116,7 @@
<element>
<name>rtppcmadepay</name>
<longname>RTP PCMA depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts PCMA audio from RTP packets</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;, Zeeshan Ali &lt;zeenix@gmail.com&gt;</author>
<pads>
@ -1074,7 +1137,7 @@
<element>
<name>rtppcmapay</name>
<longname>RTP PCMA payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes PCMA audio into a RTP packet</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;</author>
<pads>
@ -1095,7 +1158,7 @@
<element>
<name>rtppcmudepay</name>
<longname>RTP PCMU depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts PCMU audio from RTP packets</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;, Zeeshan Ali &lt;zeenix@gmail.com&gt;</author>
<pads>
@ -1116,7 +1179,7 @@
<element>
<name>rtppcmupay</name>
<longname>RTP PCMU payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes PCMU audio into a RTP packet</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;</author>
<pads>
@ -1137,7 +1200,7 @@
<element>
<name>rtpqcelpdepay</name>
<longname>RTP QCELP depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts QCELP (PureVoice) audio from RTP packets (RFC 2658)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1158,7 +1221,7 @@
<element>
<name>rtpqdm2depay</name>
<longname>RTP QDM2 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts QDM2 audio from RTP packets (no RFC)</description>
<author>Edward Hervey &lt;bilboed@bilboed.com&gt;</author>
<pads>
@ -1179,7 +1242,7 @@
<element>
<name>rtpsirendepay</name>
<longname>RTP Siren packet depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts Siren audio from RTP packets</description>
<author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;</author>
<pads>
@ -1200,7 +1263,7 @@
<element>
<name>rtpsirenpay</name>
<longname>RTP Payloader for Siren Audio</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Packetize Siren audio streams into RTP packets</description>
<author>Youness Alaoui &lt;kakaroto@kakaroto.homelinux.net&gt;</author>
<pads>
@ -1221,7 +1284,7 @@
<element>
<name>rtpspeexdepay</name>
<longname>RTP Speex depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts Speex audio from RTP packets</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;</author>
<pads>
@ -1242,7 +1305,7 @@
<element>
<name>rtpspeexpay</name>
<longname>RTP Speex payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encodes Speex audio into a RTP packet</description>
<author>Edgard Lima &lt;edgard.lima@indt.org.br&gt;</author>
<pads>
@ -1263,7 +1326,7 @@
<element>
<name>rtpsv3vdepay</name>
<longname>RTP SVQ3 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts SVQ3 video from RTP packets (no RFC)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1284,7 +1347,7 @@
<element>
<name>rtptheoradepay</name>
<longname>RTP Theora depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts Theora video from RTP packets (draft-01 of RFC XXXX)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1305,7 +1368,7 @@
<element>
<name>rtptheorapay</name>
<longname>RTP Theora payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode Theora video into RTP packets (draft-01 RFC XXXX)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1326,7 +1389,7 @@
<element>
<name>rtpvorbisdepay</name>
<longname>RTP Vorbis depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts Vorbis Audio from RTP packets (RFC 5215)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1347,7 +1410,7 @@
<element>
<name>rtpvorbispay</name>
<longname>RTP Vorbis depayloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload-encode Vorbis audio into RTP packets (RFC 5215)</description>
<author>Wim Taymans &lt;wimi.taymans@gmail.com&gt;</author>
<pads>
@ -1368,7 +1431,7 @@
<element>
<name>rtpvrawdepay</name>
<longname>RTP Raw Video depayloader</longname>
<class>Codec/Depayloader/Network</class>
<class>Codec/Depayloader/Network/RTP</class>
<description>Extracts raw video from RTP packets (RFC 4175)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>
@ -1389,7 +1452,7 @@
<element>
<name>rtpvrawpay</name>
<longname>RTP Raw Video payloader</longname>
<class>Codec/Payloader/Network</class>
<class>Codec/Payloader/Network/RTP</class>
<description>Payload raw video as RTP packets (RFC 4175)</description>
<author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
<pads>

2
docs/plugins/inspect/plugin-rtsp.xml
View file

@ -3,7 +3,7 @@
<description>transfer data via RTSP</description>
<filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
<basename>libgstrtsp.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-shapewipe.xml
View file

@ -3,7 +3,7 @@
<description>Shape Wipe transition filter</description>
<filename>../../gst/shapewipe/.libs/libgstshapewipe.so</filename>
<basename>libgstshapewipe.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-shout2send.xml
View file

@ -3,7 +3,7 @@
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>

2
docs/plugins/inspect/plugin-smpte.xml
View file

@ -3,7 +3,7 @@
<description>Apply the standard SMPTE transitions on video images</description>
<filename>../../gst/smpte/.libs/libgstsmpte.so</filename>
<basename>libgstsmpte.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-soup.xml
View file

@ -3,7 +3,7 @@
<description>libsoup HTTP client src</description>
<filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
<basename>libgstsouphttpsrc.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-spectrum.xml
View file

@ -3,7 +3,7 @@
<description>Run an FFT on the audio signal, output spectrum data</description>
<filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
<basename>libgstspectrum.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-speex.xml
View file

@ -3,7 +3,7 @@
<description>Speex plugin library</description>
<filename>../../ext/speex/.libs/libgstspeex.so</filename>
<basename>libgstspeex.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-taglib.xml
View file

@ -3,7 +3,7 @@
<description>Tag writing plug-in based on taglib</description>
<filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
<basename>libgsttaglib.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-udp.xml
View file

@ -3,7 +3,7 @@
<description>transfer data via UDP</description>
<filename>../../gst/udp/.libs/libgstudp.so</filename>
<basename>libgstudp.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-video4linux2.xml
View file

@ -3,7 +3,7 @@
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
<basename>libgstvideo4linux2.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-videobox.xml
View file

@ -3,7 +3,7 @@
<description>resizes a video by adding borders or cropping</description>
<filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
<basename>libgstvideobox.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-videocrop.xml
View file

@ -3,7 +3,7 @@
<description>Crops video into a user-defined region</description>
<filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
<basename>libgstvideocrop.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-videofilter.xml
View file

@ -3,7 +3,7 @@
<description>Video filters plugin</description>
<filename>../../gst/videofilter/.libs/libgstvideofilter.so</filename>
<basename>libgstvideofilter.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-videomixer.xml
View file

@ -3,7 +3,7 @@
<description>Video mixer</description>
<filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
<basename>libgstvideomixer.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-wavenc.xml
View file

@ -3,7 +3,7 @@
<description>Encode raw audio into WAV</description>
<filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
<basename>libgstwavenc.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-wavpack.xml
View file

@ -3,7 +3,7 @@
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-wavparse.xml
View file

@ -3,7 +3,7 @@
<description>Parse a .wav file into raw audio</description>
<filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
<basename>libgstwavparse.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-ximagesrc.xml
View file

@ -3,7 +3,7 @@
<description>X11 video input plugin using standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
<basename>libgstximagesrc.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

2
docs/plugins/inspect/plugin-y4menc.xml
View file

@ -3,7 +3,7 @@
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
<version>0.10.26.1</version>
<version>0.10.27.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins git</package>

8
ext/Makefile.am
View file

@ -46,6 +46,12 @@ else
HAL_DIR =
endif
if USE_JACK
JACK_DIR=jack
else
JACK_DIR=
endif
if USE_JPEG
JPEG_DIR = jpeg
else
@ -135,6 +141,7 @@ SUBDIRS = \
$(GCONF_DIR) \
$(GDK_PIXBUF_DIR) \
$(HAL_DIR) \
$(JACK_DIR) \
$(JPEG_DIR) \
$(LIBCACA_DIR) \
$(LIBDV_DIR) \
@ -158,6 +165,7 @@ DIST_SUBDIRS = \
gconf \
gdk_pixbuf \
hal \
jack \
jpeg \
libcaca \
libpng \

61
ext/cairo/gstcairorender.c
View file

@ -264,52 +264,63 @@ gst_cairo_render_setcaps_sink (GstPad * pad, GstCaps * caps)
return TRUE;
}
static GstStaticPadTemplate t_src = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (
#define SIZE_CAPS "width = (int) [ 1, MAX], height = (int) [ 1, MAX] "
#if CAIRO_HAS_PDF_SURFACE
"application/pdf, "
"width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] "
#define PDF_CAPS "application/pdf, " SIZE_CAPS
#else
#define PDF_CAPS
#endif
#if CAIRO_HAS_PDF_SURFACE && (CAIRO_HAS_PS_SURFACE || CAIRO_HAS_SVG_SURFACE || CAIRO_HAS_PNG_FUNCTIONS)
";"
#define JOIN1 ";"
#else
#define JOIN1
#endif
#if CAIRO_HAS_PS_SURFACE
"application/postscript, "
"width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] "
#define PS_CAPS "application/postscript, " SIZE_CAPS
#else
#define PS_CAPS
#endif
#if (CAIRO_HAS_PDF_SURFACE || CAIRO_HAS_PS_SURFACE) && (CAIRO_HAS_SVG_SURFACE || CAIRO_HAS_PNG_FUNCTIONS)
";"
#define JOIN2 ";"
#else
#define JOIN2
#endif
#if CAIRO_HAS_SVG_SURFACE
"image/svg+xml, "
"width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] "
#define SVG_CAPS "image/svg+xml, " SIZE_CAPS
#else
#define SVG_CAPS
#endif
#if (CAIRO_HAS_PDF_SURFACE || CAIRO_HAS_PS_SURFACE || CAIRO_HAS_SVG_SURFACE) && CAIRO_HAS_PNG_FUNCTIONS
";"
#define JOIN3 ";"
#else
#define JOIN3
#endif
#if CAIRO_HAS_PNG_FUNCTIONS
"image/png, " "width = (int) [ 1, MAX], " "height = (int) [ 1, MAX] "
#endif
));
static GstStaticPadTemplate t_snk = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
GST_VIDEO_CAPS_BGRx " ; " GST_VIDEO_CAPS_BGRA " ; "
#define PNG_CAPS "image/png, " SIZE_CAPS
#define PNG_CAPS2 "; image/png, " SIZE_CAPS
#else
GST_VIDEO_CAPS_xRGB " ; " GST_VIDEO_CAPS_ARGB " ; "
#define PNG_CAPS
#define PNG_CAPS2
#endif
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define ARGB_CAPS GST_VIDEO_CAPS_BGRx " ; " GST_VIDEO_CAPS_BGRA " ; "
#else
#define ARGB_CAPS GST_VIDEO_CAPS_xRGB " ; " GST_VIDEO_CAPS_ARGB " ; "
#endif
static GstStaticPadTemplate t_src = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS (PDF_CAPS JOIN1 PS_CAPS JOIN2 SVG_CAPS JOIN3 PNG_CAPS));
static GstStaticPadTemplate t_snk = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (ARGB_CAPS
GST_VIDEO_CAPS_YUV ("Y800") " ; "
"video/x-raw-gray, "
"bpp = 8, "
"depth = 8, "
"width = " GST_VIDEO_SIZE_RANGE ", "
"height = " GST_VIDEO_SIZE_RANGE ", " "framerate = " GST_VIDEO_FPS_RANGE
" ; "
#if CAIRO_HAS_PNG_FUNCTIONS
"image/png, "
"width = " GST_VIDEO_SIZE_RANGE ", " "height = " GST_VIDEO_SIZE_RANGE
#endif
));
PNG_CAPS2));
GST_BOILERPLATE (GstCairoRender, gst_cairo_render, GstElement,
GST_TYPE_ELEMENT);

7
ext/cairo/gsttimeoverlay.c
View file

@ -36,19 +36,16 @@
#include "config.h"
#endif
#include <gst/math-compat.h>
#include <gsttimeoverlay.h>
#include <string.h>
#include <math.h>
#include <cairo.h>
#include <gst/video/video.h>
#ifndef HAVE_RINT
#define rint(x) ((double) floor((x)+(((x) < 0)? -0.5 : 0.5)))
#endif
static GstStaticPadTemplate gst_cairo_time_overlay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,

31
ext/dv/gstdvdemux.c
View file

@ -895,17 +895,13 @@ gst_dvdemux_convert_segment (GstDVDemux * dvdemux, GstSegment * src,
*
* Convert the time seek to a bytes seek and send it
* upstream
*
* FIXME, upstream might be able to perform time based
* seek too.
*
* Does not take ownership of the event.
*/
static gboolean
gst_dvdemux_handle_push_seek (GstDVDemux * dvdemux, GstPad * pad,
GstEvent * event)
{
gboolean res;
gboolean res = FALSE;
gdouble rate;
GstSeekFlags flags;
GstFormat format;
@ -917,19 +913,24 @@ gst_dvdemux_handle_push_seek (GstDVDemux * dvdemux, GstPad * pad,
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
/* we convert the start/stop on the srcpad to the byte format
* on the sinkpad and forward the event */
res = gst_dvdemux_convert_src_to_sink (dvdemux, pad,
format, cur, stop, GST_FORMAT_BYTES, &start_position, &end_position);
if (!res)
goto done;
/* First try if upstream can handle time based seeks */
if (format == GST_FORMAT_TIME)
res = gst_pad_push_event (dvdemux->sinkpad, event);
/* now this is the updated seek event on bytes */
newevent = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags,
cur_type, start_position, stop_type, end_position);
if (!res) {
/* we convert the start/stop on the srcpad to the byte format
* on the sinkpad and forward the event */
res = gst_dvdemux_convert_src_to_sink (dvdemux, pad,
format, cur, stop, GST_FORMAT_BYTES, &start_position, &end_position);
if (!res)
goto done;
res = gst_pad_push_event (dvdemux->sinkpad, newevent);
/* now this is the updated seek event on bytes */
newevent = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags,
cur_type, start_position, stop_type, end_position);
res = gst_pad_push_event (dvdemux->sinkpad, newevent);
}
done:
return res;
}

1
ext/jack/.gitignore vendored Normal file
View file

@ -0,0 +1 @@
*.loT

10
ext/jack/Makefile.am Normal file
View file

@ -0,0 +1,10 @@
plugin_LTLIBRARIES = libgstjack.la
libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS)
libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstjack_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h

97
ext/jack/gstjack.c Normal file
View file

@ -0,0 +1,97 @@
/* GStreamer Jack plugins
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstjackaudiosrc.h"
#include "gstjackaudiosink.h"
GType
gst_jack_connect_get_type (void)
{
static volatile gsize jack_connect_type = 0;
if (g_once_init_enter (&jack_connect_type)) {
static const GEnumValue jack_connect_enums[] = {
{GST_JACK_CONNECT_NONE,
"Don't automatically connect ports to physical ports", "none"},
{GST_JACK_CONNECT_AUTO,
"Automatically connect ports to physical ports", "auto"},
{GST_JACK_CONNECT_AUTO_FORCED,
"Automatically connect ports to as many physical ports as possible",
"auto-forced"},
{0, NULL, NULL},
};
GType tmp = g_enum_register_static ("GstJackConnect", jack_connect_enums);
g_once_init_leave (&jack_connect_type, tmp);
}
return (GType) jack_connect_type;
}
static gpointer
gst_jack_client_copy (gpointer jclient)
{
return jclient;
}
static void
gst_jack_client_free (gpointer jclient)
{
return;
}
GType
gst_jack_client_get_type (void)
{
static volatile gsize jack_client_type = 0;
if (g_once_init_enter (&jack_client_type)) {
/* hackish, but makes it show up nicely in gst-inspect */
GType tmp = g_boxed_type_register_static ("JackClient",
(GBoxedCopyFunc) gst_jack_client_copy,
(GBoxedFreeFunc) gst_jack_client_free);
g_once_init_leave (&jack_client_type, tmp);
}
return (GType) jack_client_type;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY,
GST_TYPE_JACK_AUDIO_SRC))
return FALSE;
if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
GST_TYPE_JACK_AUDIO_SINK))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"jack",
"JACK audio elements",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

55
ext/jack/gstjack.h Normal file
View file

@ -0,0 +1,55 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjack.h:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_JACK_H_
#define _GST_JACK_H_
/**
* GstJackConnect:
* @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
* In this mode, the element will accept any number of input channels and will
* create (but not connect) an output port for each channel.
* @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
* output port to a random physical jack input pin. The sink will
* expose the number of physical channels on its pad caps.
* @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each
* output port to a random physical jack input pin. The element will accept any number
* of input channels.
*
* Specify how the output ports will be connected.
*/
typedef enum {
GST_JACK_CONNECT_NONE,
GST_JACK_CONNECT_AUTO,
GST_JACK_CONNECT_AUTO_FORCED
} GstJackConnect;
typedef jack_default_audio_sample_t sample_t;
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
GType gst_jack_client_get_type(void);
GType gst_jack_connect_get_type(void);
#endif // _GST_JACK_H_

525
ext/jack/gstjackaudioclient.c Normal file
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/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudioclient.c: jack audio client implementation
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include "gstjackaudioclient.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
#define GST_CAT_DEFAULT gst_jack_audio_client_debug
void
gst_jack_audio_client_init (void)
{
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0,
"jackclient helpers");
}
/* a list of global connections indexed by id and server. */
G_LOCK_DEFINE_STATIC (connections_lock);
static GList *connections;
/* the connection to a server */
typedef struct
{
gint refcount;
GMutex *lock;
GCond *flush_cond;
/* id/server pair and the connection */
gchar *id;
gchar *server;
jack_client_t *client;
/* lists of GstJackAudioClients */
gint n_clients;
GList *src_clients;
GList *sink_clients;
} GstJackAudioConnection;
/* an object sharing a jack_client_t connection. */
struct _GstJackAudioClient
{
GstJackAudioConnection *conn;
GstJackClientType type;
gboolean active;
gboolean deactivate;
void (*shutdown) (void *arg);
JackProcessCallback process;
JackBufferSizeCallback buffer_size;
JackSampleRateCallback sample_rate;
gpointer user_data;
};
typedef jack_default_audio_sample_t sample_t;
typedef struct
{
jack_nframes_t nframes;
gpointer user_data;
} JackCB;
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
GList *walk;
int res = 0;
g_mutex_lock (conn->lock);
/* call sources first, then sinks. Sources will either push data into the
* ringbuffer of the sinks, which will then pull the data out of it, or
* sinks will pull the data from the sources. */
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
/* only call active clients */
if ((client->active || client->deactivate) && client->process) {
res = client->process (nframes, client->user_data);
if (client->deactivate) {
client->deactivate = FALSE;
g_cond_signal (conn->flush_cond);
}
}
}
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
/* only call active clients */
if ((client->active || client->deactivate) && client->process) {
res = client->process (nframes, client->user_data);
if (client->deactivate) {
client->deactivate = FALSE;
g_cond_signal (conn->flush_cond);
}
}
}
g_mutex_unlock (conn->lock);
return res;
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
return 0;
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
return 0;
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
GList *walk;
GST_DEBUG ("disconnect client %s from server %s", conn->id,
GST_STR_NULL (conn->server));
g_mutex_lock (conn->lock);
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
if (client->shutdown)
client->shutdown (client->user_data);
}
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
if (client->shutdown)
client->shutdown (client->user_data);
}
g_mutex_unlock (conn->lock);
}
typedef struct
{
const gchar *id;
const gchar *server;
} FindData;
static gint
connection_find (GstJackAudioConnection * conn, FindData * data)
{
/* id's must match */
if (strcmp (conn->id, data->id))
return 1;
/* both the same or NULL */
if (conn->server == data->server)
return 0;
/* we cannot compare NULL */
if (conn->server == NULL || data->server == NULL)
return 1;
if (strcmp (conn->server, data->server))
return 1;
return 0;
}
/* make a connection with @id and @server. Returns NULL on failure with the
* status set. */
static GstJackAudioConnection *
gst_jack_audio_make_connection (const gchar * id, const gchar * server,
jack_client_t * jclient, jack_status_t * status)
{
GstJackAudioConnection *conn;
jack_options_t options;
gint res;
*status = 0;
GST_DEBUG ("new client %s, connecting to server %s", id,
GST_STR_NULL (server));
/* never start a server */
options = JackNoStartServer;
/* if we have a servername, use it */
if (server != NULL)
options |= JackServerName;
/* open the client */
if (jclient == NULL)
jclient = jack_client_open (id, options, status, server);
if (jclient == NULL)
goto could_not_open;
/* now create object */
conn = g_new (GstJackAudioConnection, 1);
conn->refcount = 1;
conn->lock = g_mutex_new ();
conn->flush_cond = g_cond_new ();
conn->id = g_strdup (id);
conn->server = g_strdup (server);
conn->client = jclient;
conn->n_clients = 0;
conn->src_clients = NULL;
conn->sink_clients = NULL;
/* set our callbacks */
jack_set_process_callback (jclient, jack_process_cb, conn);
/* these callbacks cause us to error */
jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
jack_on_shutdown (jclient, jack_shutdown_cb, conn);
/* all callbacks are set, activate the client */
if ((res = jack_activate (jclient)))
goto could_not_activate;
GST_DEBUG ("opened connection %p", conn);
return conn;
/* ERRORS */
could_not_open:
{
GST_DEBUG ("failed to open jack client, %d", *status);
return NULL;
}
could_not_activate:
{
GST_ERROR ("Could not activate client (%d)", res);
*status = JackFailure;
g_mutex_free (conn->lock);
g_free (conn->id);
g_free (conn->server);
g_free (conn);
return NULL;
}
}
static GstJackAudioConnection *
gst_jack_audio_get_connection (const gchar * id, const gchar * server,
jack_client_t * jclient, jack_status_t * status)
{
GstJackAudioConnection *conn;
GList *found;
FindData data;
GST_DEBUG ("getting connection for id %s, server %s", id,
GST_STR_NULL (server));
data.id = id;
data.server = server;
G_LOCK (connections_lock);
found =
g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
if (found != NULL && jclient != NULL) {
/* we found it, increase refcount and return it */
conn = (GstJackAudioConnection *) found->data;
conn->refcount++;
GST_DEBUG ("found connection %p", conn);
} else {
/* make new connection */
conn = gst_jack_audio_make_connection (id, server, jclient, status);
if (conn != NULL) {
GST_DEBUG ("created connection %p", conn);
/* add to list on success */
connections = g_list_prepend (connections, conn);
} else {
GST_WARNING ("could not create connection");
}
}
G_UNLOCK (connections_lock);
return conn;
}
static void
gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
{
gint res;
gboolean zero;
GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount);
G_LOCK (connections_lock);
conn->refcount--;
if ((zero = (conn->refcount == 0))) {
GST_DEBUG ("closing connection %p", conn);
/* remove from list, we can release the mutex after removing the connection
* from the list because after that, nobody can access the connection anymore. */
connections = g_list_remove (connections, conn);
}
G_UNLOCK (connections_lock);
/* if we are zero, close and cleanup the connection */
if (zero) {
/* don't use conn->lock here. two reasons:
*
* 1) its not necessary: jack_deactivate() will not return until the JACK thread
* associated with this connection is cleaned up by a thread join, hence
* no more callbacks can occur or be in progress.
*
* 2) it would deadlock anyway, because jack_deactivate() will sleep
* waiting for the JACK thread, and can thus cause deadlock in
* jack_process_cb()
*/
if ((res = jack_deactivate (conn->client))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_WARNING ("Could not deactivate Jack client (%d)", res);
}
/* close connection */
if ((res = jack_client_close (conn->client))) {
/* we assume the client is gone. */
GST_WARNING ("close failed (%d)", res);
}
/* free resources */
g_mutex_free (conn->lock);
g_cond_free (conn->flush_cond);
g_free (conn->id);
g_free (conn->server);
g_free (conn);
}
}
static void
gst_jack_audio_connection_add_client (GstJackAudioConnection * conn,
GstJackAudioClient * client)
{
g_mutex_lock (conn->lock);
switch (client->type) {
case GST_JACK_CLIENT_SOURCE:
conn->src_clients = g_list_append (conn->src_clients, client);
conn->n_clients++;
break;
case GST_JACK_CLIENT_SINK:
conn->sink_clients = g_list_append (conn->sink_clients, client);
conn->n_clients++;
break;
default:
g_warning ("trying to add unknown client type");
break;
}
g_mutex_unlock (conn->lock);
}
static void
gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
GstJackAudioClient * client)
{
g_mutex_lock (conn->lock);
switch (client->type) {
case GST_JACK_CLIENT_SOURCE:
conn->src_clients = g_list_remove (conn->src_clients, client);
conn->n_clients--;
break;
case GST_JACK_CLIENT_SINK:
conn->sink_clients = g_list_remove (conn->sink_clients, client);
conn->n_clients--;
break;
default:
g_warning ("trying to remove unknown client type");
break;
}
g_mutex_unlock (conn->lock);
}
/**
* gst_jack_audio_client_get:
* @id: the client id
* @server: the server to connect to or NULL for the default server
* @type: the client type
* @shutdown: a callback when the jack server shuts down
* @process: a callback when samples are available
* @buffer_size: a callback when the buffer_size changes
* @sample_rate: a callback when the sample_rate changes
* @user_data: user data passed to the callbacks
* @status: pointer to hold the jack status code in case of errors
*
* Get the jack client connection for @id and @server. Connections to the same
* @id and @server will receive the same physical Jack client connection and
* will therefore be scheduled in the same process callback.
*
* Returns: a #GstJackAudioClient.
*/
GstJackAudioClient *
gst_jack_audio_client_new (const gchar * id, const gchar * server,
jack_client_t * jclient, GstJackClientType type,
void (*shutdown) (void *arg), JackProcessCallback process,
JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate,
gpointer user_data, jack_status_t * status)
{
GstJackAudioClient *client;
GstJackAudioConnection *conn;
g_return_val_if_fail (id != NULL, NULL);
g_return_val_if_fail (status != NULL, NULL);
/* first get a connection for the id/server pair */
conn = gst_jack_audio_get_connection (id, server, jclient, status);
if (conn == NULL)
goto no_connection;
GST_INFO ("new client %s", id);
/* make new client using the connection */
client = g_new (GstJackAudioClient, 1);
client->active = client->deactivate = FALSE;
client->conn = conn;
client->type = type;
client->shutdown = shutdown;
client->process = process;
client->buffer_size = buffer_size;
client->sample_rate = sample_rate;
client->user_data = user_data;
/* add the client to the connection */
gst_jack_audio_connection_add_client (conn, client);
return client;
/* ERRORS */
no_connection:
{
GST_DEBUG ("Could not get server connection (%d)", *status);
return NULL;
}
}
/**
* gst_jack_audio_client_free:
* @client: a #GstJackAudioClient
*
* Free the resources used by @client.
*/
void
gst_jack_audio_client_free (GstJackAudioClient * client)
{
GstJackAudioConnection *conn;
g_return_if_fail (client != NULL);
GST_INFO ("free client");
conn = client->conn;
/* remove from connection first so that it's not scheduled anymore after this
* call */
gst_jack_audio_connection_remove_client (conn, client);
gst_jack_audio_unref_connection (conn);
g_free (client);
}
/**
* gst_jack_audio_client_get_client:
* @client: a #GstJackAudioClient
*
* Get the jack audio client for @client. This function is used to perform
* operations on the jack server from this client.
*
* Returns: The jack audio client.
*/
jack_client_t *
gst_jack_audio_client_get_client (GstJackAudioClient * client)
{
g_return_val_if_fail (client != NULL, NULL);
/* no lock needed, the connection and the client does not change
* once the client is created. */
return client->conn->client;
}
/**
* gst_jack_audio_client_set_active:
* @client: a #GstJackAudioClient
* @active: new mode for the client
*
* Activate or deactive @client. When a client is activated it will receive
* callbacks when data should be processed.
*
* Returns: 0 if all ok.
*/
gint
gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
{
g_return_val_if_fail (client != NULL, -1);
/* make sure that we are not dispatching the client */
g_mutex_lock (client->conn->lock);
if (client->active && !active) {
/* we need to process once more to flush the port */
client->deactivate = TRUE;
/* need to wait for process_cb run once more */
while (client->deactivate)
g_cond_wait (client->conn->flush_cond, client->conn->lock);
}
client->active = active;
g_mutex_unlock (client->conn->lock);
return 0;
}

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ext/jack/gstjackaudioclient.h Normal file
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/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudioclient.h:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_CLIENT_H__
#define __GST_JACK_AUDIO_CLIENT_H__
#include <jack/jack.h>
#include <gst/gst.h>
G_BEGIN_DECLS
typedef enum
{
GST_JACK_CLIENT_SOURCE,
GST_JACK_CLIENT_SINK
} GstJackClientType;
typedef struct _GstJackAudioClient GstJackAudioClient;
void gst_jack_audio_client_init (void);
GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server,
jack_client_t *jclient,
GstJackClientType type,
void (*shutdown) (void *arg),
JackProcessCallback process,
JackBufferSizeCallback buffer_size,
JackSampleRateCallback sample_rate,
gpointer user_data,
jack_status_t *status);
void gst_jack_audio_client_free (GstJackAudioClient *client);
jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *client);
gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_CLIENT_H__ */

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ext/jack/gstjackaudiosink.c Normal file
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/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudiosink.c: jack audio sink implementation
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-jackaudiosink
* @see_also: #GstBaseAudioSink, #GstRingBuffer
*
* A Sink that outputs data to Jack ports.
*
* It will create N Jack ports named out_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
* When the #GstJackAudioSink:connect property is set to auto, this element
* will try to connect each output port to a random physical jack input pin. In
* this mode, the sink will expose the number of physical channels on its pad
* caps.
*
* When the #GstJackAudioSink:connect property is set to none, the element will
* accept any number of input channels and will create (but not connect) an
* output port for each channel.
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack.
* </refsect2>
*
* Last reviewed on 2006-11-30 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <stdlib.h>
#include <string.h>
#include "gstjackaudiosink.h"
#include "gstjackringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
static gboolean
gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
{
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* remove ports we don't need */
while (sink->port_count > channels) {
jack_port_unregister (client, sink->ports[--sink->port_count]);
}
/* alloc enough output ports */
sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
/* create an output port for each channel */
while (sink->port_count < channels) {
gchar *name;
/* port names start from 1 and are local to the element */
name =
g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
sink->port_count + 1);
sink->ports[sink->port_count] =
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput, 0);
if (sink->ports[sink->port_count] == NULL)
return FALSE;
sink->port_count++;
g_free (name);
}
return TRUE;
}
static void
gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
{
gint res, i = 0;
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* get rid of all ports */
while (sink->port_count) {
GST_LOG_OBJECT (sink, "unregister port %d", i);
if ((res = jack_port_unregister (client, sink->ports[i++]))) {
GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
}
sink->port_count--;
}
g_free (sink->ports);
sink->ports = NULL;
}
/* ringbuffer abstract base class */
static GType
gst_jack_ring_buffer_get_type (void)
{
static volatile gsize ringbuffer_type = 0;
if (g_once_init_enter (&ringbuffer_type)) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstJackRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_jack_ring_buffer_class_init,
NULL,
NULL,
sizeof (GstJackRingBuffer),
0,
(GInstanceInitFunc) gst_jack_ring_buffer_init,
NULL
};
GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
"GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
g_once_init_leave (&ringbuffer_type, tmp);
}
return (GType) ringbuffer_type;
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
}
/* this is the callback of jack. This should RT-safe.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstRingBuffer *buf;
GstJackRingBuffer *abuf;
gint readseg, len;
guint8 *readptr;
gint i, j, flen, channels;
sample_t **buffers, *data;
buf = GST_RING_BUFFER_CAST (arg);
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
channels = buf->spec.channels;
/* alloc pointers to samples */
buffers = g_alloca (sizeof (sample_t *) * channels);
/* get target buffers */
for (i = 0; i < channels; i++) {
buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
}
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
nframes, readptr, flen, channels);
data = (sample_t *) readptr;
/* the samples in the ringbuffer have the channels interleaved, we need to
* deinterleave into the jack target buffers */
for (i = 0; i < nframes; i++) {
for (j = 0; j < channels; j++) {
buffers[j][i] = *data++;
}
}
/* clear written samples in the ringbuffer */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
/* We are not allowed to read from the ringbuffer, write silence to all
* jack output buffers */
for (i = 0; i < channels; i++) {
memset (buffers[i], 0, nframes * sizeof (sample_t));
}
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
(gint) (nframes * sizeof (sample_t)), flen);
return 1;
}
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
(NULL), ("Jack changed the sample rate, which is not supported"));
return 1;
}
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
(NULL), ("Jack changed the buffer size, which is not supported"));
return 1;
}
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
GST_DEBUG_OBJECT (sink, "shutdown");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(NULL), ("Jack server shutdown"));
}
static void
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
GstJackRingBufferClass * g_class)
{
buf->channels = -1;
buf->buffer_size = -1;
buf->sample_rate = -1;
}
/* the _open_device method should make a connection with the server
*/
static gboolean
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
jack_status_t status = 0;
const gchar *name;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "open");
name = g_get_application_name ();
if (!name)
name = "GStreamer";
sink->client = gst_jack_audio_client_new (name, sink->server,
sink->jclient,
GST_JACK_CLIENT_SINK,
jack_shutdown_cb,
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
if (sink->client == NULL)
goto could_not_open;
GST_DEBUG_OBJECT (sink, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(_("Jack server not found")),
("Cannot connect to the Jack server (status %d)", status));
} else {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
(NULL), ("Jack client open error (status %d)", status));
}
return FALSE;
}
}
/* close the connection with the server
*/
static gboolean
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "close");
gst_jack_audio_sink_free_channels (sink);
gst_jack_audio_client_free (sink->client);
sink->client = NULL;
return TRUE;
}
/* allocate a buffer and setup resources to process the audio samples of
* the format as specified in @spec.
*
* We allocate N jack ports, one for each channel. If we are asked to
* automatically make a connection with physical ports, we connect as many
* ports as there are physical ports, leaving leftover ports unconnected.
*
* It is assumed that samplerate and number of channels are acceptable since our
* getcaps method will always provide correct values. If unacceptable caps are
* received for some reason, we fail here.
*/
static gboolean
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, channels, res;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (sink, "acquire");
client = gst_jack_audio_client_get_client (sink->client);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != spec->rate)
goto wrong_samplerate;
channels = spec->channels;
if (!gst_jack_audio_sink_allocate_channels (sink, channels))
goto out_of_ports;
buffer_size = jack_get_buffer_size (client);
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
spec->segtotal = 2;
spec->buffer_time = spec->latency_time * spec->segtotal;
}
GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
goto could_not_activate;
/* if we need to automatically connect the ports, do so now. We must do this
* after activating the client. */
if (sink->connect == GST_JACK_CONNECT_AUTO
|| sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical input ports. A physical input port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to hear something. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No physical input ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all input ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (sink, "try connecting to %s",
jack_port_name (sink->ports[i]));
/* connect the port to a physical port */
res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
if (res != 0 && res != EEXIST)
goto cannot_connect;
}
free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = spec->channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
sample_rate, spec->rate));
return FALSE;
}
out_of_ports:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not connect output ports to physical ports (%d:%s)",
res, g_strerror (res)));
free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
gint res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "release");
if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
("Could not deactivate Jack client (%d)", res));
}
abuf->channels = -1;
abuf->buffer_size = -1;
abuf->sample_rate = -1;
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "start");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "pause");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "stop");
return TRUE;
}
static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
guint i, res = 0, latency;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (sink->client);
for (i = 0; i < sink->port_count; i++) {
latency = jack_port_get_total_latency (client, sink->ports[i]);
if (latency > res)
res = latency;
}
GST_LOG_OBJECT (sink, "delay %u", res);
return res;
}
static GstStaticPadTemplate jackaudiosink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
/* AudioSink signals and args */
enum
{
/* FILL ME */
SIGNAL_LAST
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_LAST
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static void gst_jack_audio_sink_dispose (GObject * object);
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
sink);
static void
gst_jack_audio_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
"Sink/Audio", "Output audio to a JACK server",
"Wim Taymans <wim.taymans@gmail.com>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&jackaudiosink_sink_factory));
}
static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gobject_class->dispose = gst_jack_audio_sink_dispose;
gobject_class->get_property = gst_jack_audio_sink_get_property;
gobject_class->set_property = gst_jack_audio_sink_set_property;
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"Specify how the output ports will be connected",
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SERVER,
g_param_spec_string ("server", "Server",
"The Jack server to connect to (NULL = default)",
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CLIENT,
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
GST_TYPE_JACK_CLIENT,
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
gst_jack_audio_client_init ();
}
static void
gst_jack_audio_sink_init (GstJackAudioSink * sink,
GstJackAudioSinkClass * g_class)
{
sink->connect = DEFAULT_PROP_CONNECT;
sink->server = g_strdup (DEFAULT_PROP_SERVER);
sink->jclient = NULL;
sink->ports = NULL;
sink->port_count = 0;
}
static void
gst_jack_audio_sink_dispose (GObject * object)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
gst_caps_replace (&sink->caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CONNECT:
sink->connect = g_value_get_enum (value);
break;
case PROP_SERVER:
g_free (sink->server);
sink->server = g_value_dup_string (value);
break;
case PROP_CLIENT:
if (GST_STATE (sink) == GST_STATE_NULL ||
GST_STATE (sink) == GST_STATE_READY) {
sink->jclient = g_value_get_boxed (value);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CONNECT:
g_value_set_enum (value, sink->connect);
break;
case PROP_SERVER:
g_value_set_string (value, sink->server);
break;
case PROP_CLIENT:
g_value_set_boxed (value, sink->jclient);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (sink->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (sink->client);
if (sink->connect == GST_JACK_CONNECT_AUTO) {
/* get a port count, this is the number of channels we can automatically
* connect. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
max = 0;
if (ports != NULL) {
for (; ports[max]; max++);
free (ports);
} else
max = 0;
} else {
/* we allow any number of pads, something else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (client);
GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!sink->caps) {
sink->caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
return gst_caps_ref (sink->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (sink, "device not open, using template caps");
/* base class will get template caps for us when we return NULL */
return NULL;
}
}
static GstRingBuffer *
gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}

78
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/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjacksink.h:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_SINK_H__
#define __GST_JACK_AUDIO_SINK_H__
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosink.h>
#include "gstjack.h"
#include "gstjackaudioclient.h"
G_BEGIN_DECLS
#define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type())
#define GST_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink))
#define GST_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
#define GST_JACK_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
#define GST_IS_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK))
#define GST_IS_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK))
typedef struct _GstJackAudioSink GstJackAudioSink;
typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
/**
* GstJackAudioSink:
*
* Opaque #GstJackAudioSink.
*/
struct _GstJackAudioSink {
GstBaseAudioSink element;
/*< private >*/
/* cached caps */
GstCaps *caps;
/* properties */
GstJackConnect connect;
gchar *server;
jack_client_t *jclient;
/* our client */
GstJackAudioClient *client;
/* our ports */
jack_port_t **ports;
int port_count;
};
struct _GstJackAudioSinkClass {
GstBaseAudioSinkClass parent_class;
};
GType gst_jack_audio_sink_get_type (void);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_SINK_H__ */

874
ext/jack/gstjackaudiosrc.c Normal file
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/* GStreamer
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-jackaudiosrc
* @see_also: #GstBaseAudioSrc, #GstRingBuffer
*
* A Src that inputs data from Jack ports.
*
* It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
* When the #GstJackAudioSrc:connect property is set to auto, this element
* will try to connect each input port to a random physical jack output pin.
*
* When the #GstJackAudioSrc:connect property is set to none, the element will
* accept any number of output channels and will create (but not connect) an
* input port for each channel.
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack.
* </refsect2>
*
* Last reviewed on 2008-07-22 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <stdlib.h>
#include <string.h>
#include "gstjackaudiosrc.h"
#include "gstjackringbuffer.h"
#include "gstjackutil.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
#define GST_CAT_DEFAULT gst_jack_audio_src_debug
static gboolean
gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
{
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
/* remove ports we don't need */
while (src->port_count > channels)
jack_port_unregister (client, src->ports[--src->port_count]);
/* alloc enough input ports */
src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
/* create an input port for each channel */
while (src->port_count < channels) {
gchar *name;
/* port names start from 1 and are local to the element */
name =
g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
src->port_count + 1);
src->ports[src->port_count] =
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput, 0);
if (src->ports[src->port_count] == NULL)
return FALSE;
src->port_count++;
g_free (name);
}
return TRUE;
}
static void
gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
{
gint res, i = 0;
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
/* get rid of all ports */
while (src->port_count) {
GST_LOG_OBJECT (src, "unregister port %d", i);
if ((res = jack_port_unregister (client, src->ports[i++])))
GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
src->port_count--;
}
g_free (src->ports);
src->ports = NULL;
g_free (src->buffers);
src->buffers = NULL;
}
/* ringbuffer abstract base class */
static GType
gst_jack_ring_buffer_get_type (void)
{
static volatile gsize ringbuffer_type = 0;
if (g_once_init_enter (&ringbuffer_type)) {
static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_jack_ring_buffer_class_init,
NULL,
NULL,
sizeof (GstJackRingBuffer),
0,
(GInstanceInitFunc) gst_jack_ring_buffer_init,
NULL
};
GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
"GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
g_once_init_leave (&ringbuffer_type, tmp);
}
return (GType) ringbuffer_type;
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
}
/* this is the callback of jack. This should be RT-safe.
* Writes samples from the jack input port's buffer to the gst ring buffer.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstRingBuffer *buf;
gint len;
guint8 *writeptr;
gint writeseg;
gint channels, i, j, flen;
sample_t *data;
buf = GST_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
channels = buf->spec.channels;
/* get input buffers */
for (i = 0; i < channels; i++)
src->buffers[i] =
(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
/* the samples in the jack input buffers have to be interleaved into the
* ringbuffer */
data = (sample_t *) writeptr;
for (i = 0; i < nframes; ++i)
for (j = 0; j < channels; ++j)
*data++ = src->buffers[j][i];
GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
len / channels, channels);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
(gint) (nframes * sizeof (sample_t)), flen);
return 1;
}
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
(NULL), ("Jack changed the sample rate, which is not supported"));
return 1;
}
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
(NULL), ("Jack changed the buffer size, which is not supported"));
return 1;
}
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
GST_DEBUG_OBJECT (src, "shutdown");
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
(NULL), ("Jack server shutdown"));
}
static void
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
GstJackRingBufferClass * g_class)
{
buf->channels = -1;
buf->buffer_size = -1;
buf->sample_rate = -1;
}
/* the _open_device method should make a connection with the server
*/
static gboolean
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
jack_status_t status = 0;
const gchar *name;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "open");
name = g_get_application_name ();
if (!name)
name = "GStreamer";
src->client = gst_jack_audio_client_new (name, src->server,
src->jclient,
GST_JACK_CLIENT_SOURCE,
jack_shutdown_cb,
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
if (src->client == NULL)
goto could_not_open;
GST_DEBUG_OBJECT (src, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
(_("Jack server not found")),
("Cannot connect to the Jack server (status %d)", status));
} else {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE,
(NULL), ("Jack client open error (status %d)", status));
}
return FALSE;
}
}
/* close the connection with the server
*/
static gboolean
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "close");
gst_jack_audio_src_free_channels (src);
gst_jack_audio_client_free (src->client);
src->client = NULL;
return TRUE;
}
/* allocate a buffer and setup resources to process the audio samples of
* the format as specified in @spec.
*
* We allocate N jack ports, one for each channel. If we are asked to
* automatically make a connection with physical ports, we connect as many
* ports as there are physical ports, leaving leftover ports unconnected.
*
* It is assumed that samplerate and number of channels are acceptable since our
* getcaps method will always provide correct values. If unacceptable caps are
* received for some reason, we fail here.
*/
static gboolean
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, channels, res;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (src, "acquire");
client = gst_jack_audio_client_get_client (src->client);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != spec->rate)
goto wrong_samplerate;
channels = spec->channels;
if (!gst_jack_audio_src_allocate_channels (src, channels))
goto out_of_ports;
gst_jack_set_layout_on_caps (&spec->caps, channels);
buffer_size = jack_get_buffer_size (client);
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
spec->segtotal = 2;
spec->buffer_time = spec->latency_time * spec->segtotal;
}
GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
goto could_not_activate;
/* if we need to automatically connect the ports, do so now. We must do this
* after activating the client. */
if (src->connect == GST_JACK_CONNECT_AUTO
|| src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical output ports. A physical output port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to capture something. */
ports =
jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
("No physical output ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all output ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (src, "try connecting to %s",
jack_port_name (src->ports[i]));
/* connect the physical port to a port */
res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
if (res != 0 && res != EEXIST)
goto cannot_connect;
}
free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = spec->channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
sample_rate, spec->rate));
return FALSE;
}
out_of_ports:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not connect input ports to physical ports (%d:%s)",
res, g_strerror (res)));
free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
gint res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "release");
if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
("Could not deactivate Jack client (%d)", res));
}
abuf->channels = -1;
abuf->buffer_size = -1;
abuf->sample_rate = -1;
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "start");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "pause");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "stop");
return TRUE;
}
static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0, latency;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (src->client);
for (i = 0; i < src->port_count; i++) {
latency = jack_port_get_total_latency (client, src->ports[i]);
if (latency > res)
res = latency;
}
GST_DEBUG_OBJECT (src, "delay %u", res);
return res;
}
/* Audiosrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_LAST
};
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
static void gst_jack_audio_src_dispose (GObject * object);
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
src);
/* GObject vmethod implementations */
static void
gst_jack_audio_src_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
"Source/Audio", "Captures audio from a JACK server",
"Tristan Matthews <tristan@sat.qc.ca>");
}
/* initialize the jack_audio_src's class */
static void
gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gobject_class->dispose = gst_jack_audio_src_dispose;
gobject_class->set_property = gst_jack_audio_src_set_property;
gobject_class->get_property = gst_jack_audio_src_get_property;
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"Specify how the input ports will be connected",
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SERVER,
g_param_spec_string ("server", "Server",
"The Jack server to connect to (NULL = default)",
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CLIENT,
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
GST_TYPE_JACK_CLIENT,
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
gst_jack_audio_client_init ();
}
/* initialize the new element
* instantiate pads and add them to element
* set pad calback functions
* initialize instance structure
*/
static void
gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
{
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
src->connect = DEFAULT_PROP_CONNECT;
src->server = g_strdup (DEFAULT_PROP_SERVER);
src->jclient = NULL;
src->ports = NULL;
src->port_count = 0;
src->buffers = NULL;
}
static void
gst_jack_audio_src_dispose (GObject * object)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
gst_caps_replace (&src->caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
switch (prop_id) {
case PROP_CONNECT:
src->connect = g_value_get_enum (value);
break;
case PROP_SERVER:
g_free (src->server);
src->server = g_value_dup_string (value);
break;
case PROP_CLIENT:
if (GST_STATE (src) == GST_STATE_NULL ||
GST_STATE (src) == GST_STATE_READY) {
src->jclient = g_value_get_boxed (value);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
switch (prop_id) {
case PROP_CONNECT:
g_value_set_enum (value, src->connect);
break;
case PROP_SERVER:
g_value_set_string (value, src->server);
break;
case PROP_CLIENT:
g_value_set_boxed (value, src->jclient);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (src->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (src->client);
if (src->connect == GST_JACK_CONNECT_AUTO) {
/* get a port count, this is the number of channels we can automatically
* connect. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
max = 0;
if (ports != NULL) {
for (; ports[max]; max++);
free (ports);
} else
max = 0;
} else {
/* we allow any number of pads, something else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (client);
GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!src->caps) {
src->caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
return gst_caps_ref (src->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (src, "device not open, using template caps");
/* base class will get template caps for us when we return NULL */
return NULL;
}
}
static GstRingBuffer *
gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
return buffer;
}

97
ext/jack/gstjackaudiosrc.h Normal file
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@ -0,0 +1,97 @@
/* GStreamer
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_SRC_H__
#define __GST_JACK_AUDIO_SRC_H__
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiosrc.h>
#include "gstjackaudioclient.h"
#include "gstjack.h"
G_BEGIN_DECLS
#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type())
#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
typedef struct _GstJackAudioSrc GstJackAudioSrc;
typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
struct _GstJackAudioSrc
{
GstBaseAudioSrc src;
/*< private >*/
/* cached caps */
GstCaps *caps;
/* properties */
GstJackConnect connect;
gchar *server;
jack_client_t *jclient;
/* our client */
GstJackAudioClient *client;
/* our ports */
jack_port_t **ports;
int port_count;
sample_t **buffers;
};
struct _GstJackAudioSrcClass
{
GstBaseAudioSrcClass parent_class;
};
GType gst_jack_audio_src_get_type (void);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_SRC_H__ */

88
ext/jack/gstjackringbuffer.h Normal file
View file

@ -0,0 +1,88 @@
/*
* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_RING_BUFFER_H__
#define __GST_JACK_RING_BUFFER_H__
#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type())
#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj)
#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
typedef struct _GstJackRingBuffer GstJackRingBuffer;
typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
struct _GstJackRingBuffer
{
GstRingBuffer object;
gint sample_rate;
gint buffer_size;
gint channels;
};
struct _GstJackRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass);
static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer,
GstJackRingBufferClass * klass);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec);
static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf);
static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf);
#endif

114
ext/jack/gstjackutil.c Normal file
View file

@ -0,0 +1,114 @@
/* GStreamer Jack utility functions
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "gstjackutil.h"
#include <gst/audio/multichannel.h>
static const GstAudioChannelPosition default_positions[8][8] = {
/* 1 channel */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
},
/* 2 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
/* 3 channels (2.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */
},
/* 4 channels (4.0 or 3.1?) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
/* 5 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
},
/* 6 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
},
/* 7 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
},
/* 8 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
}
};
/* if channels are less than or equal to 8, we set a default layout,
* otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */
void
gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels)
{
int c;
GValue pos = { 0 };
GValue chanpos = { 0 };
gst_caps_unref (*caps);
if (channels <= 8) {
g_assert (channels >= 1);
gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0),
default_positions[channels - 1]);
} else {
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < channels; c++) {
g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (gst_caps_get_structure (*caps, 0),
"channel-positions", &chanpos);
g_value_unset (&chanpos);
}
gst_caps_ref (*caps);
}

30
ext/jack/gstjackutil.h Normal file
View file

@ -0,0 +1,30 @@
/* GStreamer
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
*
* gstjackutil.h:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_JACK_UTIL_H_
#define _GST_JACK_UTIL_H_
#include <gst/gst.h>
void
gst_jack_set_layout_on_caps (GstCaps **caps, gint channels);
#endif // _GST_JACK_UTIL_H_

147
ext/jpeg/gstjpegdec.c
View file

@ -52,11 +52,13 @@
(((struct GstJpegDecSourceMgr*)((cinfo_ptr)->src))->dec)
#define JPEG_DEFAULT_IDCT_METHOD JDCT_FASTEST
#define JPEG_DEFAULT_MAX_ERRORS 0
enum
{
PROP_0,
PROP_IDCT_METHOD
PROP_IDCT_METHOD,
PROP_MAX_ERRORS
};
/* *INDENT-OFF* */
@ -192,6 +194,21 @@ gst_jpeg_dec_class_init (GstJpegDecClass * klass)
JPEG_DEFAULT_IDCT_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstJpegDec:max-errors
*
* Error out after receiving N consecutive decoding errors
* (-1 = never error out, 0 = automatic, 1 = fail on first error, etc.)
*
* Since: 0.10.27
**/
g_object_class_install_property (gobject_class, PROP_MAX_ERRORS,
g_param_spec_int ("max-errors", "Maximum Consecutive Decoding Errors",
"Error out after receiving N consecutive decoding errors "
"(-1 = never fail, 0 = automatic, 1 = fail on first error)",
-1, G_MAXINT, JPEG_DEFAULT_MAX_ERRORS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_jpeg_dec_change_state);
@ -199,6 +216,81 @@ gst_jpeg_dec_class_init (GstJpegDecClass * klass)
GST_DEBUG_CATEGORY_GET (GST_CAT_PERFORMANCE, "GST_PERFORMANCE");
}
static void
gst_jpeg_dec_clear_error (GstJpegDec * dec)
{
g_free (dec->error_msg);
dec->error_msg = NULL;
dec->error_line = 0;
dec->error_func = NULL;
}
static void
gst_jpeg_dec_set_error_va (GstJpegDec * dec, const gchar * func, gint line,
const gchar * debug_msg_format, va_list args)
{
#ifndef GST_DISABLE_GST_DEBUG
gst_debug_log_valist (GST_CAT_DEFAULT, GST_LEVEL_WARNING, __FILE__, func,
line, (GObject *) dec, debug_msg_format, args);
#endif
g_free (dec->error_msg);
if (debug_msg_format)
dec->error_msg = g_strdup_vprintf (debug_msg_format, args);
else
dec->error_msg = NULL;
dec->error_line = line;
dec->error_func = func;
}
static void
gst_jpeg_dec_set_error (GstJpegDec * dec, const gchar * func, gint line,
const gchar * debug_msg_format, ...)
{
va_list va;
va_start (va, debug_msg_format);
gst_jpeg_dec_set_error_va (dec, func, line, debug_msg_format, va);
va_end (va);
}
static GstFlowReturn
gst_jpeg_dec_post_error_or_warning (GstJpegDec * dec)
{
GstFlowReturn ret;
int max_errors;
++dec->error_count;
max_errors = g_atomic_int_get (&dec->max_errors);
if (max_errors < 0) {
ret = GST_FLOW_OK;
} else if (max_errors == 0) {
/* FIXME: do something more clever in "automatic mode" */
if (dec->packetized) {
ret = (dec->error_count < 3) ? GST_FLOW_OK : GST_FLOW_ERROR;
} else {
ret = GST_FLOW_ERROR;
}
} else {
ret = (dec->error_count < max_errors) ? GST_FLOW_OK : GST_FLOW_ERROR;
}
GST_INFO_OBJECT (dec, "decoding error %d/%d (%s)", dec->error_count,
max_errors, (ret == GST_FLOW_OK) ? "ignoring error" : "erroring out");
gst_element_message_full (GST_ELEMENT (dec),
(ret == GST_FLOW_OK) ? GST_MESSAGE_WARNING : GST_MESSAGE_ERROR,
GST_STREAM_ERROR, GST_STREAM_ERROR_DECODE,
g_strdup (_("Failed to decode JPEG image")), dec->error_msg,
__FILE__, dec->error_func, dec->error_line);
dec->error_msg = NULL;
gst_jpeg_dec_clear_error (dec);
return ret;
}
static boolean
gst_jpeg_dec_fill_input_buffer (j_decompress_ptr cinfo)
{
@ -346,6 +438,7 @@ gst_jpeg_dec_init (GstJpegDec * dec)
/* init properties */
dec->idct_method = JPEG_DEFAULT_IDCT_METHOD;
dec->max_errors = JPEG_DEFAULT_MAX_ERRORS;
dec->adapter = gst_adapter_new ();
}
@ -954,10 +1047,9 @@ gst_jpeg_dec_decode_direct (GstJpegDec * dec, guchar * base[3],
format_not_supported:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
(_("Failed to decode JPEG image")),
("Unsupported subsampling schema: v_samp factors: %u %u %u",
v_samp[0], v_samp[1], v_samp[2]));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Unsupported subsampling schema: v_samp factors: %u %u %u",
v_samp[0], v_samp[1], v_samp[2]);
return GST_FLOW_ERROR;
}
}
@ -1440,6 +1532,11 @@ again:
goto drop_buffer;
}
/* reset error count on successful decode */
dec->error_count = 0;
++dec->good_count;
GST_LOG_OBJECT (dec, "pushing buffer (ts=%" GST_TIME_FORMAT ", dur=%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
@ -1452,6 +1549,11 @@ done:
exit:
if (G_UNLIKELY (ret == GST_FLOW_ERROR)) {
jpeg_abort_decompress (&dec->cinfo);
ret = gst_jpeg_dec_post_error_or_warning (dec);
}
return ret;
/* special cases */
@ -1468,9 +1570,8 @@ need_more_data:
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
("Picture is too small or too big (%ux%u)", width, height),
("Picture is too small or too big (%ux%u)", width, height));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Picture is too small or too big (%ux%u)", width, height);
ret = GST_FLOW_ERROR;
goto done;
}
@ -1480,8 +1581,9 @@ decode_error:
dec->jerr.pub.format_message ((j_common_ptr) (&dec->cinfo), err_msg);
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
(_("Failed to decode JPEG image")), ("Error #%u: %s", code, err_msg));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Decode error #%u: %s", code, err_msg);
if (outbuf) {
gst_buffer_unref (outbuf);
outbuf = NULL;
@ -1507,9 +1609,8 @@ alloc_failed:
jpeg_abort_decompress (&dec->cinfo);
if (ret != GST_FLOW_UNEXPECTED && ret != GST_FLOW_WRONG_STATE &&
ret != GST_FLOW_NOT_LINKED) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE,
("Buffer allocation failed, reason: %s", reason),
("Buffer allocation failed, reason: %s", reason));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Buffer allocation failed, reason: %s", reason);
}
goto exit;
}
@ -1522,22 +1623,22 @@ drop_buffer:
}
components_not_supported:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("more components than supported: %d > 3", dec->cinfo.num_components));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"more components than supported: %d > 3", dec->cinfo.num_components);
ret = GST_FLOW_ERROR;
goto done;
}
unsupported_colorspace:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Picture has unknown or unsupported colourspace"));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Picture has unknown or unsupported colourspace");
ret = GST_FLOW_ERROR;
goto done;
}
invalid_yuvrgbgrayscale:
{
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("Picture is corrupt or unhandled YUV/RGB/grayscale layout"));
gst_jpeg_dec_set_error (dec, GST_FUNCTION, __LINE__,
"Picture is corrupt or unhandled YUV/RGB/grayscale layout");
ret = GST_FLOW_ERROR;
goto done;
}
@ -1632,6 +1733,9 @@ gst_jpeg_dec_set_property (GObject * object, guint prop_id,
case PROP_IDCT_METHOD:
dec->idct_method = g_value_get_enum (value);
break;
case PROP_MAX_ERRORS:
g_atomic_int_set (&dec->max_errors, g_value_get_int (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -1651,6 +1755,9 @@ gst_jpeg_dec_get_property (GObject * object, guint prop_id, GValue * value,
case PROP_IDCT_METHOD:
g_value_set_enum (value, dec->idct_method);
break;
case PROP_MAX_ERRORS:
g_value_set_int (value, g_atomic_int_get (&dec->max_errors));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@ -1668,6 +1775,8 @@ gst_jpeg_dec_change_state (GstElement * element, GstStateChange transition)
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
dec->error_count = 0;
dec->good_count = 0;
dec->framerate_numerator = 0;
dec->framerate_denominator = 1;
dec->caps_framerate_numerator = dec->caps_framerate_denominator = 0;

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