Release 0.10.18

NEWS

-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:
     

configure.ac

@@ -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 ****
 

docs/plugins/gst-plugins-good-plugins.args

@@ -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>
+</ARG>
+
 <ARG>
 <NAME>GstRtpJitterBuffer::do-lost</NAME>
 <TYPE>gboolean</TYPE>
@@ -20391,7 +20471,7 @@
 <ARG>
 <NAME>GstV4l2Sink::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>

docs/plugins/gst-plugins-good-plugins.hierarchy

@@ -11,68 +11,69 @@ GObject
     GstElement
       GstBin
         GstPipeline
-        GstHalAudioSink
-        GstHalAudioSrc
         GstGConfVideoSink
         GstGConfVideoSrc
         GstSwitchSink
           GstGConfAudioSink
         GstGConfAudioSrc
+        GstHalAudioSink
+        GstHalAudioSrc
+        GstRtpBin
         GstAutoVideoSink
         GstAutoVideoSrc
         GstAutoAudioSink
         GstAutoAudioSrc
-        GstRgVolume
-        GstRtpBin
-        GstRTSPSrc
         GstPushFileSrc
+        GstRTSPSrc
+        GstRgVolume
         GstAspectRatioCrop
+      GstCmmlEnc
+      GstCmmlDec
       GstBaseSink
+        GstAASink
         GstBaseAudioSink
+          GstPulseSink
           GstAudioSink
             GstEsdSink
             GstOssSink
-          GstPulseSink
-        GstShout2send
         GstCACASink
         GstVideoSink
           GstGdkPixbufSink
           GstV4l2Sink
-        GstAASink
+        GstShout2send
+        GstTest
+        GstMultiFileSink
         GstMultiUDPSink
           GstUDPSink
         GstDynUDPSink
-        GstMultiFileSink
-        GstTest
-      GstJpegEnc
-      GstJpegDec
-      GstSmokeEnc
-      GstSmokeDec
       GstBaseSrc
         GstPushSrc
+          GstDV1394Src
+          GstHDV1394Src
+          GstSoupHTTPSrc
           GstBaseAudioSrc
             GstAudioSrc
               GstPulseSrc
               GstOssSrc
-          GstSoupHTTPSrc
-          GstDV1394Src
-          GstHDV1394Src
           GstV4l2Src
           GstXImageSrc
-          GstUDPSrc
           GstMultiFileSrc
-      GstPulseMixer
-      GstFlacEnc
-      GstFlacDec
-      GstFlacTag
+          GstUDPSrc
+      GstWavpackParse
+      GstWavpackDec
+      GstWavpackEnc
+      GstDVDemux
+      GstDVDec
       GstTagLibMux
         GstId3v2Mux
         GstApev2Mux
+      GstFlacEnc
+      GstFlacDec
+      GstFlacTag
       GstCairoTextOverlay
       GstBaseTransform
         GstCairoTimeOverlay
         GstPixbufScale
-        GstLevel
         GstVideoFilter
           GstEdgeTV
           GstAgingTV
@@ -86,19 +87,25 @@ GObject
           GstRadioacTV
           GstStreakTV
           GstRippleTV
-          GstVideoBalance
+          GstNavigationtest
           GstVideoFlip
+          GstVideoBalance
           GstGamma
           GstSMPTEAlpha
-          GstNavigationtest
-        GstRgAnalysis
-        GstRgLimiter
+        GstVideoBox
+        GstBreakMyData
+        GstNavSeek
+        GstProgressReport
+        GstTagInject
+        GstLevel
+        GstAlpha
+        GstAlphaColor
         GstAudioFilter
-          GstSpectrum
           GstIirEqualizer
             GstIirEqualizerNBands
             GstIirEqualizer3Bands
             GstIirEqualizer10Bands
+          GstSpectrum
           GstAudioInvert
           GstAudioKaraoke
           GstAudioAmplify
@@ -112,36 +119,33 @@ GObject
             GstAudioWSincBand
             GstAudioFIRFilter
           GstAudioEcho
-        GstAlpha
-        GstAlphaColor
-        GstAudioPanorama
-        GstVideoBox
-        GstBreakMyData
-        GstNavSeek
-        GstProgressReport
-        GstTagInject
+        GstRgAnalysis
+        GstRgLimiter
         GstVideoCrop
+        GstAudioPanorama
       GstCairoRender
+      GstPulseMixer
+      GstSpeexEnc
+      GstSpeexDec
+      GstJpegEnc
+      GstJpegDec
+      GstSmokeEnc
+      GstSmokeDec
       GstPngDec
       GstPngEnc
-      GstWavpackParse
-      GstWavpackDec
-      GstWavpackEnc
       GstGdkPixbuf
-      GstCmmlEnc
-      GstCmmlDec
-      GstSpeexEnc
-      GstSpeexDec
-      GstDVDemux
-      GstDVDec
       GstOssMixerElement
-      GstWavEnc
+      GstAviDemux
+      GstAviMux
+      GstAviSubtitle
       GstRTPDepay
       GstBaseRTPDepayload
         GstRtpAC3Depay
+        GstRTPBVDepay
         GstRtpCELTDepay
         GstRTPDVDepay
         GstRTPiLBCDepay
+        GstRtpG723Depay
         GstRtpG726Depay
         GstRtpG729Depay
         GstRTPGSMDepay
@@ -170,15 +174,18 @@ GObject
         GstRtpVRawDepay
         GstRtpXQTDepay
       GstBaseRTPPayload
-        GstRtpCELTPay
-        GstRTPDVPay
         GstBaseRTPAudioPayload
+          GstRTPBVPay
           GstRTPILBCPay
           GstRtpG726Pay
           GstRTPG729Pay