Commit 34c128b6 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
Browse files

Release 0.10.18

parent b4040721
This diff is collapsed.
This is GStreamer Good Plug-ins 0.10.17, "They used to sparkle"
This is GStreamer Good Plug-ins 0.10.18, "Short Circuit"
Changes since 0.10.17:
* v4l2src: implement GstURIHandler interface
* matroskamux: make index size configurable
* matroskademux: support push based mode
* matroskademux: improve stream synchronization
* flacdec: fix possible hanging in pull mode seeking
* flacdec: use a single decoder field for both push and pull mode
* flacenc: optionally add a seek table
* rtp: add BroadcomVoice payloader and depayloader
* rtp: add G.723 payloader and depayloader
* rtph264pay: add option to insert PPS/SPS in streams
* rtph264depay: optionally merge NALUs into Access Units
* rtspsrc: add user-id and user-pw properties; fix major memory leak
* avimux: many fixes, also better compatibility with Windows Media Player
* avidemux: per-stream index parsing (= much faster startup)
* qtdemux: progressive download support / seeking in push mode
* qtdemux: per sample parsing (= much faster start up)
* wavenc: Post warning if file hasn't been finalised properly
* videomixer: MMX optimisations and other improvements; implement basic QoS
* matroska, qtdemux, id3demux: fix language code writing and extraction
Bugs fixed since 0.10.17:
* 609405 : [qtdemux] Issues when seeking with file with lots of tracks and edit lists
* 503582 : [avidemux] Extract date tag (contained in the IDIT chunk)
* 351595 : [flacenc] write seek tables
* 505823 : [matroskademux] language tags have wrong iso code
* 515073 : [goom] Update to goom2k4
* 539858 : not enough NEWSEGMENT events from matroskademux
* 554839 : [rtpbin] Automaticaly remove pads
* 582575 : [rtph263depay] dropping only part of key frames on lost fragmets
* 583367 : gstrtpL16pay ignores max-ptime property
* 583985 : [matroskamux] make index size configurable
* 587323 : rtpmp4vpay does not payload mp4v stream depayloaded with rtpmp4vdepay
* 593354 : rtpjitterbuffer sometimes outputs packets with timestamps in the past
* 595265 : SDES handling in RTPSource
* 597497 : can't play a redirecting .mov file via playbin
* 597823 : Add rtpg723pay plugin
* 599300 : [qtdemux] Doesn't populate video bitrate field
* 601143 : v4l2src: add GstURIHandler interface
* 601242 : [flvmux] ECMA array with file index lacks final 0x09 byte
* 601728 : [rtspsrc] Add username/password properties
* 602231 : Deadlock between rtpjitterbuffer and gstrtpbin
* 602508 : qtdemux: Parse stbl atom per sample instead of all at once
* 602887 : shout2send element won't change from PLAYING state to NULL
* 602940 : jitterbuffer is racy determining basetime
* 603376 : rtpsession : g_type_create_instance performance issue : avoid buffer ref
* 603471 : [flacdec] not timestamping output buffers
* 603547 : shout2send plugin sends data too fast
* 603779 : [ladspa] Remove ladspa plugin code
* 604352 : [rganalysis] miscomputes timestamps
* 604611 : [qtdemux] Provides invalid ALAC codec data
* 604679 : videomixer MMX code doesn't build on fedora12
* 604814 : videomixer make error
* 604872 : [udpsink] Add missing 'gssize len' parameter to g_convert()
* 604913 : rtph264pay/NALU/rtph264depay
* 605222 : Mobile Youtube RTSP streams time out at EOS
* 605269 : [shout2][patch] Setting public flag
* 605447 : Unable to play Real Audio stream for radioBERLIN.
* 605882 : rtpg723pay is incorrect
* 606198 : rtph264pay is causing alignment trap on ARM arch
* 606438 : multiudpsink: warningfixes for windows
* 606692 : Incorrect Center Frequency For Band3
* 606807 : audioamplify: allow negative amplifications
* 607353 : rtph264pay & base: Don't crash if the other side specifies the profile-level-id
* 607440 : [wavenc] should post warning if the file isn't finished properly on pipeline shutting down
* 607718 : [qtdemux] Infinite loop doing negative rate playback for single audio stream
* 607949 : [avidemux] regression in stop position for mp3 streams
* 608209 : [videomixer] blend_mmx.h:173: Error: can't encode register '%ah' in an instruction requiring REX prefix
* 608255 : [speex] speexenc crash and leaks in rtpspeexpay and speexdec
* 608268 : [flvmux] index timestamps should be in seconds, not milliseconds
* 608629 : [pngdec] png_set_gray_1_2_4_to_8() removed in libpng > = 1.4.0
* 608671 : [mkv] issues when seeking
* 608990 : [qtdemux] Segment start timestamps can be broken
* 609107 : [qtdemux] Unknown atoms should also be skipped when looking for moov
* 598610 : [matroskademux] Support push mode operation
* 594381 : audiofirfilter: Implement FFT convolution
Changes since 0.10.16:
......
Release notes for GStreamer Good Plug-ins 0.10.17 "They used to sparkle"
Release notes for GStreamer Good Plug-ins 0.10.18 "Short Circuit"
......@@ -54,79 +54,82 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release
* RTP improvements
* Support automatic cropping in videobox
* Add TTL multicast UDP property
* AVI demux push mode fixes and performance improvements
* Support large and unusual chunks sizes in wav
* Quicktime demuxer improvements
* JPEG decode fixes and speedups
* Support interlaced Y4M file output
* DV demuxer improvements
* Pulseaudio fixes and improvements
* Support Pulseaudio PLAY/PAUSE requests
* speexdec improvements
* FLV demuxer improvements
* Fix audio noise in the Equalizer plugin, and other improvements
* Fix compilation on OS/X Snow Leopard
* AVI muxer fixes
* Support MPEG V4L2 devices and improve timestamping
* Better jpeg2k support
* Many other bug fixes and improvements
* v4l2src: implement GstURIHandler interface
* matroskamux: make index size configurable
* matroskademux: support push based mode
* matroskademux: improve stream synchronization
* flacdec: fix possible hanging in pull mode seeking
* flacdec: use a single decoder field for both push and pull mode
* flacenc: optionally add a seek table
* rtp: add BroadcomVoice payloader and depayloader
* rtp: add G.723 payloader and depayloader
* rtph264pay: add option to insert PPS/SPS in streams
* rtph264depay: optionally merge NALUs into Access Units
* rtspsrc: add user-id and user-pw properties; fix major memory leak
* avimux: many fixes, also better compatibility with Windows Media Player
* avidemux: per-stream index parsing (= much faster startup)
* qtdemux: progressive download support / seeking in push mode
* qtdemux: per sample parsing (= much faster start up)
* wavenc: Post warning if file hasn't been finalised properly
* videomixer: MMX optimisations and other improvements; implement basic QoS
* matroska, qtdemux, id3demux: fix language code writing and extraction
Bugs fixed in this release
* 597848 : " Media Player Classic " won't play certain files produced by avimux.
* 588245 : TTL is never applied with udpsink/udpmultisink
* 368681 : avimux + vbr lame always out of sync
* 458629 : [avidemux] high memory usage for many index entries
* 561825 : Problem with RTCP thread using freed objects
* 581334 : [qtdemux] Add support for embedded subtitles
* 582238 : [videobox] Add support for autocrop to caps
* 590362 : [v4l2src] x264enc ! qtmux fails because of missing frame duration
* 591713 : [y4menc] interlaced support
* 609405 : [qtdemux] Issues when seeking with file with lots of tracks and edit lists
* 503582 : [avidemux] Extract date tag (contained in the IDIT chunk)
* 351595 : [flacenc] write seek tables
* 505823 : [matroskademux] language tags have wrong iso code
* 515073 : [goom] Update to goom2k4
* 539858 : not enough NEWSEGMENT events from matroskademux
* 554839 : [rtpbin] Automaticaly remove pads
* 582575 : [rtph263depay] dropping only part of key frames on lost fragmets
* 583367 : gstrtpL16pay ignores max-ptime property
* 583985 : [matroskamux] make index size configurable
* 587323 : rtpmp4vpay does not payload mp4v stream depayloaded with rtpmp4vdepay
* 593354 : rtpjitterbuffer sometimes outputs packets with timestamps in the past
* 593391 : [rtpsession] : rtp_session_on_timeout : Invalid read of size 4
* 593688 : effectv can no longer be compiled with gcc 3
* 593757 : [qtdemux] Lack of support for QualComm PureVoice
* 593764 : [v4l2src] format ordering: put emulated formats behind native formats
* 593955 : rtpjitterbuffer: clock_rate can change between its check and its use
* 594039 : missing unref in rtpsource / leak
* 594133 : [rtspsrc] leaks authentication info
* 594247 : missing math.h include in rtpjpegdepay
* 594248 : Use locked-state on internal rtp-bin to avoid shutdown-state-race
* 594251 : Avoid throwing out reordered packets with the same timestamp
* 594253 : jitterbuf: Only post a warning of clock-rate changed if it is changed from something initialized
* 594254 : propagate the pt-type-changed signal
* 594283 : rtpbin: make free_session() remove dangling stream references
* 594298 : Check if libsoup has SSL support before running HTTPS test in souphttpsrc testsuite
* 594490 : gstrtpbin always uses pt to demux
* 594520 : multipartmux: mark data buffer as delta-unit
* 594599 : videobox: converts AYUV to I420 incorrectly
* 594663 : Patch for multifilesink
* 594691 : rtph263pay: leak
* 595029 : pulse elements fail to connect to pulse 0.9.9
* 595220 : gstreamer crashes on pulseaudio latency change
* 595231 : [pulsesink] Lowers volume after every new track
* 595888 : qtdemux plugin should not return value from void function
* 595897 : Problem linking videomixer
* 595942 : [qtdemux] issue with corrupted 3gp file
* 596319 : [qtdemux] fails to parse pixel aspect ratio data
* 597091 : [flvdemux] not outputting no-more-pads causes playbin2 to fail badly on streamed single-stream flv
* 597214 : [avidemux] Fix printf formats to avoid warnings in avidemux
* 597348 : [qtdemux] Cast variables passed to printf to avoid warnings about incorrect formats
* 597351 : [jpegdec] segfaults on a specific picture
* 597397 : equalizer is non deterministic
* 597463 : [pulsesrc] has no lower bound for fragment size
* 597601 : [pulsesink] needs to take control of minreq value
* 597730 : osssrc rank should be secondary, just like osssink
* 597847 : Windows Media Player won't play large files produced by avimux
* 597867 : Plugins good do not build on Ubuntu Hardy (kernel 2.6.24)
* 598377 : rtpmanager: only forward the lost event to the last seen payloadnumber
* 598517 : [jpegdec] Regression supporting 4:2:2 jpeg videos
* 598810 : wavenc: Fix buffer offset by moving length incrementation
* 598933 : [pulse] Fix the StreamVolume interface not being advertised
* 601381 : v4l2: Make sure to initialize variables before using them
* 595265 : SDES handling in RTPSource
* 597497 : can't play a redirecting .mov file via playbin
* 597823 : Add rtpg723pay plugin
* 599300 : [qtdemux] Doesn't populate video bitrate field
* 601143 : v4l2src: add GstURIHandler interface
* 601242 : [flvmux] ECMA array with file index lacks final 0x09 byte
* 601728 : [rtspsrc] Add username/password properties
* 602231 : Deadlock between rtpjitterbuffer and gstrtpbin
* 602508 : qtdemux: Parse stbl atom per sample instead of all at once
* 602887 : shout2send element won't change from PLAYING state to NULL
* 602940 : jitterbuffer is racy determining basetime
* 603376 : rtpsession : g_type_create_instance performance issue : avoid buffer ref
* 603471 : [flacdec] not timestamping output buffers
* 603547 : shout2send plugin sends data too fast
* 603779 : [ladspa] Remove ladspa plugin code
* 604352 : [rganalysis] miscomputes timestamps
* 604611 : [qtdemux] Provides invalid ALAC codec data
* 604679 : videomixer MMX code doesn't build on fedora12
* 604814 : videomixer make error
* 604872 : [udpsink] Add missing 'gssize len' parameter to g_convert()
* 604913 : rtph264pay/NALU/rtph264depay
* 605222 : Mobile Youtube RTSP streams time out at EOS
* 605269 : [shout2][patch] Setting public flag
* 605447 : Unable to play Real Audio stream for radioBERLIN.
* 605882 : rtpg723pay is incorrect
* 606198 : rtph264pay is causing alignment trap on ARM arch
* 606438 : multiudpsink: warningfixes for windows
* 606692 : Incorrect Center Frequency For Band3
* 606807 : audioamplify: allow negative amplifications
* 607353 : rtph264pay & base: Don't crash if the other side specifies the profile-level-id
* 607440 : [wavenc] should post warning if the file isn't finished properly on pipeline shutting down
* 607718 : [qtdemux] Infinite loop doing negative rate playback for single audio stream
* 607949 : [avidemux] regression in stop position for mp3 streams
* 608209 : [videomixer] blend_mmx.h:173: Error: can't encode register '%ah' in an instruction requiring REX prefix
* 608255 : [speex] speexenc crash and leaks in rtpspeexpay and speexdec
* 608268 : [flvmux] index timestamps should be in seconds, not milliseconds
* 608629 : [pngdec] png_set_gray_1_2_4_to_8() removed in libpng > = 1.4.0
* 608671 : [mkv] issues when seeking
* 608990 : [qtdemux] Segment start timestamps can be broken
* 609107 : [qtdemux] Unknown atoms should also be skipped when looking for moov
* 598610 : [matroskademux] Support push mode operation
* 594381 : audiofirfilter: Implement FFT convolution
Download
......@@ -156,40 +159,31 @@ Applications
Contributors to this release
* Alessandro Decina
* Andy Wingo
* Arnout Vandecappelle
* Arun Raghavan
* Aurelien Grimaud
* Bastien Nocera
* Brian Cameron
* Christian F.K. Schaller
* David Henningsson
* David Schleef
* Branko Čibej
* David Hoyt
* Edward Hervey
* Gabriel Millaire
* Havard Graff
* Håvard Graff
* Jan Schmidt
* Jarkko Palviainen
* Josep Torra
* Laurent Glayal
* Lennart Poettering
* Marc Leeman
* Marc-André Lureau
* Jan Urbański
* Jonathan Conder
* Kipp Cannon
* Marco Ballesio
* Mark Nauwelaerts
* Marvin Schmidt
* Michael Smith
* Olivier Crête
* Pau Garcia i Quiles
* Peter Kjellerstedt
* Priit Laes
* René Stadler
* Pascal Buhler
* Peter van Hardenberg
* Robert Swain
* Robert Weidlich
* Roland Krikava
* Sebastian Dröge
* Sjoerd Simons
* Stefan Kost
* Stephen Jungels
* Stig Sandnes
* Thiago Santos
* Tiago Katcipis
* Tim-Philipp Müller
* Wim Taymans
* Zaheer Merali
 
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
AC_INIT(GStreamer Good Plug-ins, 0.10.17.3,
AC_INIT(GStreamer Good Plug-ins, 0.10.18,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-good)
......@@ -47,8 +47,8 @@ AC_LIBTOOL_WIN32_DLL
AM_PROG_LIBTOOL
dnl *** required versions of GStreamer stuff ***
GST_REQ=0.10.25.2
GSTPB_REQ=0.10.25.2
GST_REQ=0.10.26
GSTPB_REQ=0.10.26
dnl *** autotools stuff ****
......
......@@ -33,8 +33,8 @@
<TYPE>gdouble</TYPE>
<RANGE>[-24,12]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>227 Hz</NICK>
<BLURB>gain for the frequency band 227 Hz, ranging from -24 dB to +12 dB.</BLURB>
<NICK>237 Hz</NICK>
<BLURB>gain for the frequency band 237 Hz, ranging from -24 dB to +12 dB.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
......@@ -451,7 +451,7 @@
<ARG>
<NAME>GstUDPSrc::sockfd</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket to use for UDP reception. (-1 == allocate).</BLURB>
......@@ -501,7 +501,7 @@
<ARG>
<NAME>GstUDPSrc::sock</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket currently in use for UDP reception. (-1 = no socket).</BLURB>
......@@ -698,6 +698,26 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTSPSrc::user-id</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>user-id</NICK>
<BLURB>RTSP location URI user id for authentication.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTSPSrc::user-pw</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>user-pw</NICK>
<BLURB>RTSP location URI user password for authentication.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPDec::skip</NAME>
<TYPE>gint</TYPE>
......@@ -733,7 +753,7 @@
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>mesage</NICK>
<NICK>message</NICK>
<BLURB>Post a level message for each passed interval.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
......@@ -1038,6 +1058,16 @@
<DEFAULT>"source"</DEFAULT>
</ARG>
<ARG>
<NAME>GstShout2send::public</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>public</NICK>
<BLURB>If the stream should be listed on the server's stream directory.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstSpeexDec::enh</NAME>
<TYPE>gboolean</TYPE>
......@@ -1551,7 +1581,7 @@
<ARG>
<NAME>GstDV1394Src::port</NAME>
<TYPE>gint</TYPE>
<RANGE>[-1,16]</RANGE>
<RANGE>[G_MAXULONG,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
......@@ -1728,6 +1758,16 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstFlacEnc::seekpoints</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -2147483647</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Seekpoints</NICK>
<BLURB>Add SEEKTABLE metadata (if > 0, number of entries, if < 0, interval in sec).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstOssSink::device</NAME>
<TYPE>gchar*</TYPE>
......@@ -1768,10 +1808,20 @@
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstMatroskaMux::min-index-interval</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Minimum time between index entries</NICK>
<BLURB>An index entry is created every so many nanoseconds.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstTest::allowed-timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>allowed timestamp deviation</NICK>
<BLURB>allowed average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
......@@ -1781,7 +1831,7 @@
<ARG>
<NAME>GstTest::buffer-count</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>buffer count</NICK>
<BLURB>number of buffers in stream.</BLURB>
......@@ -1791,7 +1841,7 @@
<ARG>
<NAME>GstTest::expected-buffer-count</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected buffer count</NICK>
<BLURB>expected number of buffers in stream.</BLURB>
......@@ -1801,7 +1851,7 @@
<ARG>
<NAME>GstTest::expected-length</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected length</NICK>
<BLURB>expected length of stream.</BLURB>
......@@ -1821,7 +1871,7 @@
<ARG>
<NAME>GstTest::length</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>length</NICK>
<BLURB>length of stream.</BLURB>
......@@ -1841,7 +1891,7 @@
<ARG>
<NAME>GstTest::timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>timestamp deviation</NICK>
<BLURB>average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
......@@ -1911,7 +1961,7 @@
<ARG>
<NAME>GstBreakMyData::set-to</NAME>
<TYPE>gint</TYPE>
<RANGE>[-1,255]</RANGE>
<RANGE>[G_MAXULONG,255]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>set-to</NICK>
<BLURB>set changed bytes to this value (-1 means random value.</BLURB>
......@@ -2241,7 +2291,7 @@
<ARG>
<NAME>GstDynUDPSink::sockfd</NAME>
<TYPE>gint</TYPE>
<RANGE>[-1,32767]</RANGE>
<RANGE>[G_MAXULONG,32767]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>socket handle</NICK>
<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
......@@ -2311,7 +2361,7 @@
<ARG>
<NAME>GstMultiUDPSink::sock</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket currently in use for UDP sending. (-1 == no socket).</BLURB>
......@@ -2321,7 +2371,7 @@
<ARG>
<NAME>GstMultiUDPSink::sockfd</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
......@@ -2351,7 +2401,7 @@
<ARG>
<NAME>GstMultiUDPSink::qos-dscp</NAME>
<TYPE>gint</TYPE>
<RANGE>[-1,63]</RANGE>
<RANGE>[G_MAXULONG,63]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>QoS diff srv code point</NICK>
<BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB>
......@@ -2671,7 +2721,7 @@
<ARG>
<NAME>GstAudioAmplify::amplification</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Amplification</NICK>
<BLURB>Factor of amplification.</BLURB>
......@@ -2931,7 +2981,7 @@
<ARG>
<NAME>GstV4l2Src::device-fd</NAME>
<TYPE>gint</TYPE>
<RANGE>>= -1</RANGE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>File descriptor</NICK>
<BLURB>File descriptor of the device.</BLURB>
......@@ -3091,7 +3141,7 @@
<ARG>
<NAME>GstAudioWSincBand::length</NAME>
<TYPE>gint</TYPE>
<RANGE>[3,50000]</RANGE>
<RANGE>[3,256000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
......@@ -3111,7 +3161,7 @@
<ARG>
<NAME>GstAudioWSincLimit::length</NAME>
<TYPE>gint</TYPE>
<RANGE>[3,50000]</RANGE>
<RANGE>[3,256000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
......@@ -3151,7 +3201,7 @@
<ARG>
<NAME>GstRndBufferSize::max</NAME>
<TYPE>glong</TYPE>
<RANGE>>= 1</RANGE>
<RANGE>[1,G_MAXINT]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>maximum</NICK>
<BLURB>maximum buffer size.</BLURB>
......@@ -3161,7 +3211,7 @@
<ARG>
<NAME>GstRndBufferSize::min</NAME>
<TYPE>glong</TYPE>
<RANGE>>= 0</RANGE>
<RANGE>[0,G_MAXINT]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>mininum</NICK>
<BLURB>mininum buffer size.</BLURB>
......@@ -3171,7 +3221,7 @@
<ARG>
<NAME>GstRndBufferSize::seed</NAME>
<TYPE>gulong</TYPE>
<RANGE></RANGE>
<RANGE><= G_MAXUINT</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>random number seed</NICK>
<BLURB>seed for randomness (initialized when going from READY to PAUSED).</BLURB>
......@@ -3454,7 +3504,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>profile-level-id</NICK>
<BLURB>The base64 profile-level-id to set in out caps (set to NULL to extract from stream).</BLURB>
<BLURB>The base64 profile-level-id to set in the sink caps (deprecated).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
......@@ -3488,6 +3538,16 @@
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpH264Pay::config-interval</NAME>
<TYPE>guint</TYPE>
<RANGE><= 3600</RANGE>
<FLAGS>rw</FLAGS>
<NICK>SPS PPS Send Interval</NICK>
<BLURB>Send SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>ladspa-hardLimiter::Residue-level</NAME>
<TYPE>gfloat</TYPE>
......@@ -19468,6 +19528,16 @@
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpH264Depay::access-unit</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Access Unit</NICK>
<BLURB>Merge NALU into AU (picture).</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioKaraoke::filter-band</NAME>
<TYPE>gfloat</TYPE>
......@@ -19801,7 +19871,7 @@
<ARG>
<NAME>GstHDV1394Src::port</NAME>
<TYPE>gint</TYPE>
<RANGE>[-1,16]</RANGE>
<RANGE>[G_MAXULONG,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
......@@ -20075,7 +20145,7 @@
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Deinterlace Mode.</BLURB>
<DEFAULT>Enfore deinterlacing</DEFAULT>
<DEFAULT>Force deinterlacing</DEFAULT>
</ARG>
<ARG>
......@@ -20268,6 +20338,16 @@
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpBin::autoremove</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Auto Remove</NICK>
<BLURB>Automatically removed timed out sources.</BLURB>
<DEFAULT>FALSE</DEFAULT>