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

Release 0.10.18

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