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2008-05-16ext/timidity/gstwildmidi.c: Check some more common locations for a valid ↵Sebastian Dröge2-0/+29
configuration file. Original commit message from CVS: * ext/timidity/gstwildmidi.c: (wildmidi_open_config): Check some more common locations for a valid configuration file. Fixes bug #533435. Packagers should still #define WILDMIDI_CFG to the distributions default location.
2008-05-16gst/interleave/: Add support for all raw audio formats and provide better ↵Sebastian Dröge5-64/+403
negotiation if the caps are changing. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: (deinterleave_24), (gst_deinterleave_finalize), (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_init), (gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps), (gst_deinterleave_set_process_function), (gst_deinterleave_sink_setcaps), (__remove_channels), (__set_channels), (gst_deinterleave_getcaps), (gst_deinterleave_process), (gst_deinterleave_chain), (gst_deinterleave_sink_activate_push): * gst/interleave/deinterleave.h: Add support for all raw audio formats and provide better negotiation if the caps are changing. Don't allow changes of the channel positions and set the position of the corresponding channel on the src pad caps. General cleanup and smaller bugfixes. * tests/check/elements/deinterleave.c: (float_buffer_check_probe): Check the channel positions on the output buffer caps.
2008-05-16docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.Jan Schmidt5-7/+28
Original commit message from CVS: * docs/Makefile.am: Don't attempt to build plugin docs when they're disabled. * gst/bayer/Makefile.am: Add libgstvideo to the link. * gst/rtpmanager/Makefile.am: Fix link order, and move LIBS things to _LIBS
2008-05-16docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull ↵Jan Schmidt2-4/+6
'metadata' plugin's base class into the docs. Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull 'metadata' plugin's base class into the docs.
2008-05-14gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a ↵Wim Taymans2-4/+12
warning instead of just posting an error and ... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain): Simply drop bad RTP packets with a warning instead of just posting an error and stopping. This is a perfectly recoverable event and we don't force people to use an rtpbin to filter out bad packets first.
2008-05-14gst/mpeg4videoparse/mpeg4videoparse.c: Set fixed caps on the srcpad after we ↵Wim Taymans3-1/+6
created the pad... Original commit message from CVS: * gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init): Set fixed caps on the srcpad after we created the pad...
2008-05-14tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again ↵Tim-Philipp Müller3-1/+6
now that there are suppressions in gst.s... Original commit message from CVS: * tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again now that there are suppressions in gst.supp which make this work for me.
2008-05-14tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, ↵Tim-Philipp Müller3-0/+202
since it causes weird invalid free errors in ... Original commit message from CVS: * tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in valgrind/libc after _exit for some reason. * tests/check/elements/deinterleave.c: (pads_created), (set_channel_positions), (src_handoff_float32_8ch), (float_buffer_check_probe), (pad_added_setup_data_check_float32_8ch_cb), (make_fake_src_8chans_float32), (GST_START_TEST), (deinterleave_suite): Add some more deinterleave unit test bits I had locally.
2008-05-14gst/audioresample/gstaudioresample.c: Revert previous change which made ↵Tim-Philipp Müller1-0/+4
basetransform handle buffer_alloc and which b... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Revert previous change which made basetransform handle buffer_alloc and which breaks things badly in the non-passthrough case since it returned buffers with a different (ie. sometimes smaller) size than the size requested.
2008-05-14gst/interleave/: Split definitions into separate header files for better ↵Sebastian Dröge9-120/+594
documentation generation. Original commit message from CVS: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.h: * gst/interleave/interleave.h: * gst/interleave/plugin.h: Split definitions into separate header files for better documentation generation. * gst/interleave/deinterleave.c: (gst_deinterleave_base_init), (gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps), (gst_deinterleave_process): Don't use alloca, allow caps changes as long as the number of channels does not change, don't use g_warning, return NOT_NEGOTIATED as early as possible and some other cleanup. * gst/interleave/interleave.c: (gst_interleave_base_init), (gst_interleave_class_init): Do some random cleanup. * tests/check/Makefile.am: * tests/check/elements/deinterleave.c: (GST_START_TEST), (deinterleave_chain_func), (deinterleave_pad_added), (deinterleave_suite): Add unit tests for the deinterleave element.
2008-05-13gst/mpeg4videoparse/mpeg4videoparse.*: Parse the config data (either ↵Sjoerd Simons3-30/+563
outbound or in the stream) to set width/height, ... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_set_new_caps), (gst_mpeg4vparse_align), (get_bits), (next_start_code), (gst_mpeg4vparse_handle_vos), (gst_mpeg4vparse_push), (gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain), (gst_mpeg4vparse_sink_setcaps), (gst_mpeg4vparse_sink_event), (gst_mpeg4vparse_src_query), (gst_mpeg4vparse_set_property), (gst_mpeg4vparse_get_property), (gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init): * gst/mpeg4videoparse/mpeg4videoparse.h: Parse the config data (either outbound or in the stream) to set width/height, apect ration, framerate in the caps if applicable. Mark frames as GST_BUFFER_FLAG_DELTA_UNIT when they are not intra frames Set the timestamps of outgoing buffers to the buffer in which the VOP header was found. Drop incoming data untill configuration is found (by default, configurable using a property). Report a 1 frame latency. Fixes #532723.
2008-05-13gst/real/gstrealvideodec.c: Add some debug for where we are searching for ↵Wim Taymans2-0/+8
libraries. Original commit message from CVS: * gst/real/gstrealvideodec.c: (open_library): Add some debug for where we are searching for libraries.
2008-05-13tests/check/elements/audioresample.c: Add unit test for the latest ↵Sjoerd Simons1-43/+77
basetransform negotiation changes. Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * tests/check/elements/audioresample.c: (live_switch_alloc_only_48000), (live_switch_get_sink_caps), (live_switch_push), (GST_START_TEST): Add unit test for the latest basetransform negotiation changes. See bug #526768.
2008-05-13gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.Wim Taymans2-0/+10
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init): Actually add the do-lost property to the object.
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) ↵Wim Taymans2-2/+15
duration when the last packet has a lower ti... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Avoid waiting for a negative (huge) duration when the last packet has a lower timestamp than the current packet.
2008-05-12gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned ↵Peter Kjellerstedt2-0/+10
by gst_pad_get_parent() to prevent a memor... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src): Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memory leak.
2008-05-12docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs outputJan Schmidt2-2/+11
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs output
2008-05-12gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to ↵Jan Schmidt3-1/+7
avoid compiler warning. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
2008-05-11Random doc of the day: the deinterlace element.Jan Schmidt7-11/+102
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-gstinterlace.xml: * gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.h: Random doc of the day: the deinterlace element.
2008-05-09gst/mpegtsparse/: Make sure all schedule EIT and non-actual transport streamZaheer Abbas Merali3-0/+31
Original commit message from CVS: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtsparse.c: Make sure all schedule EIT and non-actual transport stream EITs are parsed. Also add present-following flag and actual-transport-stream flag to eit bus message.
2008-05-09gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to ↵Peter Kjellerstedt3-0/+7
prevent a memory leak. Original commit message from CVS: * gst/rtpmanager/rtpsource.c: (rtp_source_finalize): Make sure to unref the caps used by RTPSource to prevent a memory leak.
2008-05-08sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value ↵Clive Wright2-2/+17
repeated in the upper bits, so mask those o... Original commit message from CVS: Based on patch by: Clive Wright <clive_wright ntlworld com> * sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume): Apparently mono sliders have the mono value repeated in the upper bits, so mask those out when reading them. Probably makes the mixer applet work properly in some more cases.
2008-05-08gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our ↵Olivier Crete2-0/+17
callbacks. Original commit message from CVS: Patch by: Olivier Crete <tester at tester dot ca> * gst/rtpmanager/rtpsession.c: (source_clock_rate), (rtp_session_process_bye), (rtp_session_send_bye_locked): Unlock the session lock when calling one of our callbacks. Fixes #532011.
2008-05-08gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.Sjoerd Simons2-0/+9
Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_event_send_rtp_sink): Send RTP BYE command on EOS. Fixes bug #531955.
2008-05-08gst/audioresample/gstaudioresample.c: Let audioresample use the buffer ↵Sjoerd Simons2-4/+89
allocation of basetransform instead of it's ow... Original commit message from CVS: Patch by: Sjoerd Simons <sjoerd at luon dot net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Let audioresample use the buffer allocation of basetransform instead of it's own stuff. * tests/check/elements/audioresample.c: (alloc_only_48000), (GST_START_TEST), (audioresample_suite): Add unit test for the recent basetransform bugfix, where upstream changes caps to something that can't be passed through anymore.
2008-05-07win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC ↵Ole André Vadla Ravnås2-3/+6
we'd much rather use the real thing than h... Original commit message from CVS: * win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than having "???" unconditionally.
2008-05-07gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.Wim Taymans8-80/+465
Original commit message from CVS: * gst-libs/gst/app/.cvsignore: * gst-libs/gst/app/Makefile.am: * gst-libs/gst/app/gstapp-marshal.list: Add marshal.list, make it compile and add to cvsignore. * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose), (gst_app_sink_stop): Small cleanups. * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init), (gst_app_src_init), (gst_app_src_set_property), (gst_app_src_get_property), (gst_app_src_unlock), (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop), (gst_app_src_create), (gst_app_src_set_caps), (gst_app_src_get_caps), (gst_app_src_set_size), (gst_app_src_get_size), (gst_app_src_set_seekable), (gst_app_src_get_seekable), (gst_app_src_set_max_buffers), (gst_app_src_get_max_buffers), (gst_app_src_push_buffer), (gst_app_src_end_of_stream): * gst-libs/gst/app/gstappsrc.h: Beat appsrc in shape, add signals and actions. Add some docs. Add properties for caps, size, seekability and max-buffers. Fix unlock/stop code.
2008-05-07configure.ac: Error out if we don't have the required versions of core/base.Tim-Philipp Müller3-6/+10
Original commit message from CVS: * configure.ac: Error out if we don't have the required versions of core/base.
2008-05-05gst-libs/gst/app/gstappsink.*: Start some docs.Wim Taymans3-25/+185
Original commit message from CVS: * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init), (gst_app_sink_init), (gst_app_sink_set_property), (gst_app_sink_get_property), (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked), (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll), (gst_app_sink_render), (gst_app_sink_set_caps), (gst_app_sink_set_drop), (gst_app_sink_get_drop): * gst-libs/gst/app/gstappsink.h: Start some docs. Add property to drop buffers when the queue is filled Fix unlocking and flushing when the queues are filled.
2008-05-02add wildmidi pluginChristian Schaller1-1/+4
Original commit message from CVS: add wildmidi plugin
2008-04-29gst/subenc/gstsrtenc.c: Declare variables at the beginning of blocks. Fixes ↵Jens Granseuer2-4/+14
compilation with gcc 2.x and other compil... Original commit message from CVS: Patch by: Jens Granseuer <jensgr at gmx dot net> * gst/subenc/gstsrtenc.c: (gst_srt_enc_timestamp_to_string): Declare variables at the beginning of blocks. Fixes compilation with gcc 2.x and other compilers. Fixes bug #530611.
2008-04-29gst/mpegtsparse/: Detect SI pids (NIT, SDT, EIT etc.) based on table id and ↵Zaheer Abbas Merali4-36/+85
not by pid number. This allows for exampl... Original commit message from CVS: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtspacketizer.h: * gst/mpegtsparse/mpegtsparse.c: Detect SI pids (NIT, SDT, EIT etc.) based on table id and not by pid number. This allows for example the EPG data from UK's freesat to be picked up.
2008-04-25ext/: Cast NULL sentinels to void * as NULL is defined as an integer ↵Sebastian Dröge3-4/+13
constant in most environments when using C++ and... Original commit message from CVS: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/soundtouch/gstbpmdetect.cc: Cast NULL sentinels to void * as NULL is defined as an integer constant in most environments when using C++ and it's size might be different from a pointer.
2008-04-25gst-libs/gst/app/gstappsink.*: Add more docs.Wim Taymans3-24/+247
Original commit message from CVS: * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init), (gst_app_sink_init), (gst_app_sink_set_property), (gst_app_sink_get_property), (gst_app_sink_event), (gst_app_sink_preroll), (gst_app_sink_render), (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers), (gst_app_sink_pull_buffer): * gst-libs/gst/app/gstappsink.h: Add more docs. Add signals for when preroll and render buffers are available. Add property to control signal emission. Add property to control the max queue size.
2008-04-25gst-libs/gst/dshow/Makefile.am: Use CXXFLAGS rather than CFLAGS; these are ↵Michael Smith3-5/+14
C++ files. Original commit message from CVS: * gst-libs/gst/dshow/Makefile.am: Use CXXFLAGS rather than CFLAGS; these are C++ files. Define required constants appropriately. * sys/dshowdecwrapper/Makefile.am: Add required include dir, libraries. Define required constants appropriately.
2008-04-25gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.Wim Taymans3-2/+25
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init), (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): * gst/rtpmanager/gstrtpbin.h: Expose new jitterbuffer property in rtpbin too.
2008-04-25gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost ↵Wim Taymans2-14/+63
events by default and make a property to ena... Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Disable sending out rtp packet lost events by default and make a property to enabe it. We will likely enable it by default when the base depayloaders have a default handler for them so that we don't send these events all through the pipeline for now.
2008-04-25gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function ↵Wim Taymans2-37/+121
that is in -base now. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): Remove private version of a function that is in -base now. Add src event handler. Rework the jitterbuffer pushing loop so that it can quickly react to lost packets and instruct the depayloader of them. This can then be used to implement error concealment data.
2008-04-25gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the ↵Wim Taymans2-0/+43
RTCP and sync pads because the defaults ... Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink), (create_send_rtcp_src): Set up some internal links functions for the RTCP and sync pads because the defaults are really not correct. Implement a query handler for the RTCP src pad, mostly to correctly report about the latency.
2008-04-25gst/rtpmanager/: Also keep track of the first buffer timestamp together with ↵Wim Taymans6-1/+27
the first Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain): * gst/rtpmanager/rtpsession.c: (update_arrival_stats), (rtp_session_process_sr), (rtp_session_on_timeout): * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter): * gst/rtpmanager/rtpsource.h: * gst/rtpmanager/rtpstats.h: Also keep track of the first buffer timestamp together with the first RTP timestamp as they both are needed to construct the timing of outgoing packets in the jitterbuffer and are therefore also needed to manage lip-sync. This fixes lip-sync if the first RTP packets arrive with a wildly different gap.
2008-04-25gst/flv/gstflvdemux.c: Forward unknown queries upstream instead of returning ↵Wim Taymans2-3/+6
FALSE on them. Original commit message from CVS: * gst/flv/gstflvdemux.c: (gst_flv_demux_query): Forward unknown queries upstream instead of returning FALSE on them.
2008-04-24Add support for the new libmpcdec API which magically gets us support for ↵Sebastian Dröge6-28/+148
SV8 files. Also do some random cleanup. Fix... Original commit message from CVS: * configure.ac: * ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init), (gst_musepackdec_init), (gst_musepackdec_dispose), (gst_musepackdec_handle_seek_event), (gst_musepack_stream_init), (gst_musepackdec_loop), (plugin_init): * ext/musepack/gstmusepackdec.h: * ext/musepack/gstmusepackreader.c: * ext/musepack/gstmusepackreader.h: Add support for the new libmpcdec API which magically gets us support for SV8 files. Also do some random cleanup. Fixes bug #526905.
2008-04-24configure.ac: Back to development -> 0.10.7.1Jan Schmidt2-1/+6
Original commit message from CVS: * configure.ac: Back to development -> 0.10.7.1 === release 0.10.7 ===
2008-04-24Release 0.10.7RELEASE-0_10_7Jan Schmidt70-343/+657
Original commit message from CVS: Release 0.10.7
2008-04-24Update .po filesJan Schmidt27-389/+266
Original commit message from CVS: Update .po files
2008-04-22ext/faad/gstfaad.c: Don't leak GstAudioChannelPosition. Fixes #529378.Stefan Kost2-1/+7
Original commit message from CVS: * ext/faad/gstfaad.c: Don't leak GstAudioChannelPosition. Fixes #529378.
2008-04-22gst/sdp/gstsdpdemux.c: Ref caps, see #528245.Wim Taymans2-0/+7
Original commit message from CVS: * gst/sdp/gstsdpdemux.c: (request_pt_map): Ref caps, see #528245.
2008-04-22configure.ac: 0.10.6.4 pre-releaseJan Schmidt2-1/+6
Original commit message from CVS: * configure.ac: 0.10.6.4 pre-release
2008-04-21tests/check/elements/rganalysis.c: Don't leak a tag list. Fixes bug #529285.Sebastian Dröge2-0/+7
Original commit message from CVS: * tests/check/elements/rganalysis.c: (GST_START_TEST): Don't leak a tag list. Fixes bug #529285.
2008-04-21tests/check/elements/ofa.c: Don't leak the tags string and tag list. Fixes ↵Sebastian Dröge2-5/+14
bug #529283. Original commit message from CVS: * tests/check/elements/ofa.c: (bus_handler): Don't leak the tags string and tag list. Fixes bug #529283.
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