Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* 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_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Original commit message from CVS:
Patch by: Sebastien Merle <sylane at gmail dot com>
* ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_chain):
Fix the sign of the data we pass to the decoder so that the decoder
compiles with newer reference code as well. Fixes#528618.
Original commit message from CVS:
Patch by: Sebastien Merle <sylane at gmail dot com>
* ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_chain):
Fix the sign of the data we pass to the decoder so that the decoder
compiles with newer reference code as well. Fixes#528618.
Original commit message from CVS:
Based on a patch by: Sonicadvance1 at GMAIL dot COM
* ext/timidity/gstwildmidi.c: (gst_wildmidi_init),
(gst_wildmidi_change_state), (plugin_init):
* ext/timidity/gstwildmidi.h:
Initialize wildmidi only once in the plugin init function instead
of once for every instance. The second and following calls to the
wildmidi initialization function will fail. Fixes bug #525613.
Also don't register the element at all if wildmidi initialization
fails.
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
Original commit message from CVS:
* examples/app/appsrc-stream.c: (found_source), (main):
Use deep-notify until we can depend on a playbin2 with support for the
source property.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
Fix deadlock when shutting down, use a new lock instead to properly
shutdown.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(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_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_src_query):
* gst/interleave/interleave.c: (gst_interleave_src_query_duration),
(gst_interleave_src_query):
Properly implement duration and position queries in bytes format. We
have to take the upstream reply and divide/multiply it by the number
of channels to get the correct result.
Original commit message from CVS:
* sys/dshowvideosink/dshowvideofakesrc.cpp:
* sys/dshowvideosink/dshowvideofakesrc.h:
* sys/dshowvideosink/dshowvideosink.cpp:
* sys/dshowvideosink/dshowvideosink.h:
Fix up copyright notice on new plugin.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_header_buf),
(gst_x264_enc_sink_event), (gst_x264_enc_chain),
(gst_x264_enc_encode_frame):
Try harder not to crash when we get an EOS event but haven't set
up the encoder yet (as may happen when upstream errors out with
not-negotiated, for example). Also, always push the EOS event
downstream.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_pad_get_property), (gst_interleave_pad_class_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad):
* gst/interleave/interleave.h:
Use an always increasing integer for the number in the name of the
requested sink pads to guarantuee a unique name. Add a "channel"
property to GstInterleavePad to make it possible for applications
to retrieve the channel number in the output for every pad.
Use g_type_register_static_simple() instead of
g_type_register_static() to save some relocations.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_change_state):
Stop GstCollectPads before calling the parent's state change function
when going from PAUSED to READY as we otherwise deadlock.
Fixes bug #536258.
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_nal_bs_init),
(gst_h264_parse_sink_setcaps), (gst_h264_parse_chain_forward),
(gst_h264_parse_queue_buffer), (gst_h264_parse_chain_reverse),
(gst_h264_parse_chain):
* gst/h264parse/gsth264parse.h:
Parse codec_data and use the nalu_size_length field to get the NALU
length in packetized h264.
When queueing a packetized buffer in reverse mode, don't unref the
buffer twice.
Avoid accessing the buffer TIMESTAMP field after we pushed it on
the adaptor.
Original commit message from CVS:
* gst/interleave/interleave.c:
(gst_interleave_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init):
Use new gst_audio_check_channel_positions() function and register
the GstInterleavePad type from a threadsafe context.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_finalize), (gst_audio_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_set_property), (gst_interleave_get_property),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_collected):
* gst/interleave/interleave.h:
Allow setting channel positions via a property and allow using the
channel positions on the input as the channel positions of the output.
Fix some broken logic and memory leaks.
* tests/check/Makefile.am:
* tests/check/elements/interleave.c: (src_handoff_float32),
(sink_handoff_float32), (GST_START_TEST), (interleave_suite):
Add unit tests for checking correct handling of channel positions.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_dispose), (clear_queued),
(flush_queued), (gst_faad_drain), (gst_faad_do_raw_seek),
(gst_faad_src_event), (gst_faad_sink_event), (gst_faad_chain),
(gst_faad_change_state):
* ext/faad/gstfaad.h:
Add basic reverse playback support.
Clear decoder state after disconts.
Remove some unused code.
Mark output buffers with a discont after a decoding error.
Original commit message from CVS:
Based on patch by: <onkarshinde at gmail dot com>
* sys/vcd/vcdsrc.c: (gst_vcdsrc_uri_get_uri),
(gst_vcdsrc_uri_set_uri):
Allow the track to be set by using the uri. Fixes#535043.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
Original commit message from CVS:
* configure.ac:
Require CVS core and base for new audio clock reset method.
* ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_change_state):
Reset the audio clock. See #521761.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels):
Include the element name in the port name to avoid duplicate port names.
Original commit message from CVS:
* gst/interleave/deinterleave.c:
Add another example launch line.
* gst/interleave/interleave.c: (interleave_24),
(gst_interleave_finalize), (gst_interleave_base_init),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_change_state), (__remove_channels),
(__set_channels), (gst_interleave_sink_getcaps),
(gst_interleave_set_process_function),
(gst_interleave_sink_setcaps), (gst_interleave_sink_event),
(gst_interleave_src_query_duration), (gst_interleave_src_query),
(forward_event_func), (forward_event), (gst_interleave_src_event),
(gst_interleave_collected):
* gst/interleave/interleave.h:
Major rewrite of interleave using GstCollectpads. This new version
also supports almost all raw audio formats and has better caps
negotiation. Fixes bug #506594.
Also update docs and add some more examples.
* tests/check/elements/interleave.c: (interleave_chain_func),
(GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
(interleave_suite):
Add some more extensive unit tests for interleave.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
When checking the seqnum, reset the jitterbuffer if the gap is too big,
we need to do this so that we can better handle a restarted source.
Fix some comments.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
Tweak the skew resync diff.
Use our working seqnum compare function in -base.
Rework the jitterbuffer insert code to make it clearer and more
performant by only retrieving the seqnum of the input buffer once and by
adding some G_LIKELY compiler hints.
Improve debugging for duplicate packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Fix a comment, we don't do skew correction here..
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_set_property):
Propagate the do-lost and latency properties to the jitterbuffers when
they are changed on rtpbin.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps):
* gst/interleave/deinterleave.h:
Don't set a getcaps() function on the src pads as it's not required
and the default getcaps() function returns the correct results for
our src pads.
Complete documentation and add myself to the authors of the element.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-oss4.c: (opt_show_mixer_messages), (WAIT_TIME),
(show_mixer_messages), (probe_mixer_tracks), (probe_pad),
(probe_details), (probe_element), (main):
Small oss4 test that probes for available devices and retrieves
their caps and mixer tracks and all that. Also allows testing of
mixer change messages on the bus.
Original commit message from CVS:
* sys/oss4/oss4-mixer.c: (gst_oss4_mixer_open):
* sys/oss4/oss4-property-probe.c:
(gst_oss4_property_probe_find_device_name),
(gst_oss4_property_probe_find_device_name_nofd):
* sys/oss4/oss4-property-probe.h:
* sys/oss4/oss4-sink.c: (gst_oss4_sink_get_property):
* sys/oss4/oss4-source.c: (gst_oss4_source_get_property):
Make device-name probing in NULL state work better (e.g. for the
gnome-control-center sound capplet).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_push),
(gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain),
(gst_mpeg4vparse_change_state):
Move some code around to integrate the startcode searching with the
other bits of parsing, avoid a whole bunch of peeks.
Get rid of invalid data that should not happen according to the specs.
Fixes#533559.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess dot net>
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_class_init),
(gst_mythtv_src_init), (gst_mythtv_src_clear),
(do_read_request_response), (gst_mythtv_src_create),
(gst_mythtv_src_start):
Correctly set duration to get a more correct seek bar in totem.
Disable query and event functions as they don't work and do some
smaller cleanup.
Fixes bug #533736.
Original commit message from CVS:
* tests/check/elements/deinterleave.c: (GST_START_TEST):
Set keep-positions property to TRUE for the 8 channel test to ensure
that the original channel position is set on the output.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
(gst_deinterleave_init), (gst_deinterleave_add_new_pads),
(gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
(gst_deinterleave_get_property):
* gst/interleave/deinterleave.h:
Add a property to select whether channel positions should be kept on
the mono output buffers or should be dropped.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
(gst_deinterleave_init), (gst_deinterleave_sink_event),
(gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Queue events until src pads were added and they can be sent. Otherwise
downstream will never get the first newsegment event.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
(gst_deinterleave_getcaps):
Always set the channel positions when gst_audio_get_channel_positions()
returns something, even if they're not set in the caps. This makes
sure that the output channels can be interleaved again correctly
in the mono/stereo cases too.
Don't ask for the peercaps of the current pad in getcaps() as this
might call getcaps() again and deadlock.
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.
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.
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
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.
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.
Original commit message from CVS:
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init):
Set fixed caps on the srcpad after we created the pad...
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.
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.
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.
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.
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.
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.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
Original commit message from CVS:
* tests/check/Makefile.am:
Don't inlcude dc1394src in the generic/states test as it requires
special hardware. Fixes bug #528011.
Original commit message from CVS:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST):
Only check if the generated fingerprints are valid Base64. The
fingerprints are different when running on different architectures
which is a) no problem because the fingerprints are tolerant enough
and b) is caused by libofa. Fixes bug #528266.
Original commit message from CVS:
* ext/timidity/Makefile.am:
Dist all source files, no matter if only timidity or wildmidi or
nothing is found by configure. Fixes bug #528000.
Original commit message from CVS:
* ext/dirac/gstdiracenc.cc:
Fix compilation by casting string constants.
* sys/Makefile.am:
Fix WININET_DIR variable reference.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script):
Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes
crash caused by a strlen on a NULL string (#527622).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (PROP_DEVICE_NAME),
(gst_dshowvideosrc_class_init), (gst_dshowvideosrc_init),
(gst_dshowvideosrc_dispose), (gst_dshowvideosrc_stop),
(gst_dshowvideosrc_unlock), (gst_dshowvideosrc_unlock_stop),
(gst_dshowvideosrc_create), (gst_dshowvideosrc_push_buffer):
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
Don't increase latency by queuing buffers in an async queue when
the streaming thread can't keep up or isn't scheduled often
enough for some other reason, but just drop the previous buffer
in that case. Also implement GstBaseSrc::unlock for faster
unlocking when shutting down. (#520892).
Original commit message from CVS:
* tests/icles/metadata_editor.c: (ENC_UNKNOWN), (last_pixbuf),
(draw_pixbuf), (change_tag_list), (update_draw_pixbuf),
(ui_drawing_size_allocate_cb), (on_drawingMain_expose_event),
(on_buttonSaveFile_clicked), (ui_create), (me_gst_bus_callback_view),
(me_gst_setup_view_pipeline), (process_file):
* tests/icles/metadata_editor.glade:
Remove GstXOverlay stuff and use gdkpixbufsink plus some rather crude
drawing/scaling logic to make this compile and work on all platforms.
Fixes#518227.
Original commit message from CVS:
* configure.ac:
Bump core/base requirements to released versions to avoid
confusion.
* gst/deinterlace/gstdeinterlace.c: (deinterlace_debug),
(GST_CAT_DEFAULT), (gst_deinterlace_base_init),
(gst_deinterlace_set_caps), (plugin_init):
Add debug category, use _set_element_details_simple and
remove special code path for Y42B to calculate offsets and
strides; libgstvideo knows how to handle this format now.
Original commit message from CVS:
* gst/cdxaparse/Makefile.am:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxastrip.c:
* gst/cdxaparse/gstcdxastrip.h:
* gst/cdxaparse/gstvcdparse.c:
* gst/cdxaparse/gstvcdparse.h:
Port VCD parser (formerly cdxastrip) from 0.8 to 0.10. Doesn't do
anything the 0.8 version didn't do though.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
* configure.ac:
* sys/Makefile.am:
* sys/wininet/Makefile.am:
* sys/wininet/gstwininetsrc.c:
* sys/wininet/gstwininetsrc.h:
Add wininetsrc for basic http/ftp support on windows (#520897).
Original commit message from CVS:
* tests/check/elements/souphttpsrc.c: (got_buffer),
(souphttpsrc_suite):
Increase the timeout for the internet tests to 250 seconds
and check for NULL caps instead of just crashing.
The real fix would be to implement an shoutcast server for the unit test
instead of relying on a working internet connection.
Fixes bug #521749.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_session):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
Avoid leaking pads in the RTP manager.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_response_cb):
Only ignore actual redirects not all responses when in state
GST_SOUP_HTTP_SRC_SESSION_IO_STATUS_RUNNING. Fixes bug #526337.
Original commit message from CVS:
* tests/check/elements/ofa.c: (GST_START_TEST):
Also check that we have processed at least 135 seconds of audio
until we stop and calculated a fingerprint.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST),
(ofa_suite), (main):
Add simple unit tests for the OFA plugin.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Cable delivery subsystem descriptors' frequency's bcd
is measured in 100Hz units so adjust multiplier accordingly.
Original commit message from CVS:
Patch by: kapil <kapil at fluendo dot com>
* ext/gsm/gstgsmdec.c: (gst_gsmdec_sink_setcaps),
(gst_gsmdec_chain):
* ext/gsm/gstgsmdec.h:
Increase the allowed samplerates for the ms-gsm format.
Fixes#481354.
Original commit message from CVS:
* gst/nsf/Makefile.am:
* gst/nsf/fds_snd.c:
* gst/nsf/mmc5_snd.c:
* gst/nsf/nsf.c:
* gst/nsf/types.h:
* gst/nsf/vrc7_snd.c:
* gst/nsf/vrcvisnd.c:
* gst/nsf/memguard.c:
* gst/nsf/memguard.h:
Remove memguard again and apply hopefully all previously dropped
local patches. Should be really better than the old version now.
Original commit message from CVS:
* gst/nsf/memguard.c: (_my_free):
* gst/nsf/types.h:
Unbreak compilation by disabling memguard and doing some dirty hack
fixes to make it compile on 64bits.
Original commit message from CVS:
Patch by: Ed Catmur <ed at catmur dot co dot uk>
* configure.ac:
Add support for neon 0.28, which didn't change API. Fixes bug #524035.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_got_headers_cb),
(gst_soup_http_src_chunk_allocator),
(gst_soup_http_src_got_chunk_cb),
(gst_soup_http_src_uri_get_protocols):
Don't autoplug souphttpsrc for dav/davs. This is better handled by
GIO and GnomeVFS as they provide authentication.
Don't leak the icy caps if we already set them and get a new
icy-metaint header.
Try harder to set the icy caps on the output buffer to have correct
caps for the first buffer already.
* tests/check/elements/souphttpsrc.c: (got_buffer),
(GST_START_TEST):
Check that we get a buffer with application/x-icy caps if iradio-mode
is enabled and we have an icecast URL.
Original commit message from CVS:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_chunk_allocator):
Actually set the icy caps on our src pad if we have icecast data.
Fixes bug #523854.
Original commit message from CVS:
* configure.ac:
Check if the compiler supports do { } while (0) macros. This fixes
a warning when compiling with g++ 4.3, resulting in a build failure
because of -Werror.
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
* ext/mplex/gstmplex.cc:
Include <string.h> for memcpy and friends to fix the build with
gcc 4.3.
* tests/check/Makefile.am:
Remove trailing backslash.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
(gst_input_selector_set_active_pad), (gst_input_selector_switch):
Do g_object_notify() only when not holding the lock to get the property
because otherwise we run into a deadlock with the deep-notify handlers
that are possibly installed.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_event), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_input_selector_set_active_pad):
Release the selector lock when pad alloc happens on a non selected pad.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_init), (gst_selector_pad_set_property),
(gst_selector_pad_get_property), (gst_selector_pad_event),
(gst_selector_pad_bufferalloc), (gst_selector_pad_chain),
(gst_input_selector_set_active_pad):
Add pad property to configure behaviour of the unselected pad, it can
return OK or NOT_LINKED, based on the use case.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
(gst_selector_pad_get_running_time), (gst_selector_pad_reset),
(gst_selector_pad_event), (gst_selector_pad_bufferalloc),
(gst_input_selector_wait), (gst_selector_pad_chain),
(gst_input_selector_class_init), (gst_input_selector_init),
(gst_input_selector_dispose), (gst_segment_set_start),
(gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_get_property),
(gst_input_selector_get_linked_pad),
(gst_input_selector_is_active_sinkpad),
(gst_input_selector_activate_sinkpad),
(gst_input_selector_request_new_pad),
(gst_input_selector_release_pad),
(gst_input_selector_change_state), (gst_input_selector_block),
(gst_input_selector_switch):
* gst/selector/gstinputselector.h:
Figure out the locking a bit more.
Mark buffers with discont after switching.
Fix initial segment forwarding, make sure to only forward one segment
regardless of what the sequence of buffers/segments is. See #522203.
Improve flushing when blocked.
Return NOT_LINKED when a stream is not selected.
Not API change for the switch signal in the docs.
Fix start/time/accum values of the new segment.
Correctly unlock and flush a blocking selector when going to READY.
Original commit message from CVS:
* gst/freeze/FAQ:
* gst/freeze/Makefile.am:
* gst/freeze/gstfreeze.c:
Add example to source code documentation blob and remove the 3 line
FAQ.
* gst/interleave/interleave.c:
Add a source code documentation blob.
Original commit message from CVS:
* ext/ofa/gstofa.c: (create_fingerprint), (gst_ofa_event),
(gst_ofa_transform_ip), (plugin_init):
Improve debugging, clean up a bit and really generate the fingerprint
after 135 seconds.
Original commit message from CVS:
Based on a patch by: Eric Buehl <eric dot buehl at gmail dot com>
* configure.ac:
* ext/ofa/Makefile.am:
* ext/ofa/gstofa.c: (gst_ofa_base_init), (gst_ofa_finalize),
(gst_ofa_class_init), (create_fingerprint), (gst_ofa_event),
(gst_ofa_init), (gst_ofa_transform_ip), (gst_ofa_get_property),
(plugin_init):
* ext/ofa/gstofa.h:
Add an OFA element, the successor of MusicBrainz TRM fingerprinting.
Fixes bug #351309.
Original commit message from CVS:
2008-03-18 Andy Wingo <wingo@pobox.com>
* ext/faad/gstfaad.c (gst_faad_chain): Fix a bad format argument,
and a potential int overflow.
* ext/faad/gstfaad.h: Include <neaacdec.h> if faad is neaac.
Avoids a #warning about an ignored #pragma.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_class_init),
(gst_neonhttp_src_send_request_and_redirect):
Handle HTTP status code 303 (See Other) the same way
as 302 (Found). Not sure what to do about all the other 3xx
redirect status codes. Fixes bug #522884.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_event),
(gst_selector_pad_bufferalloc), (gst_selector_pad_chain),
(gst_input_selector_class_init),
(gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_push_pending_stop):
Add lots of debugging.
Fix time member in the newsegment event.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_input_selector_class_init),
(gst_input_selector_init), (gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_get_property),
(gst_input_selector_request_new_pad),
(gst_input_selector_release_pad),
(gst_input_selector_push_pending_stop),
(gst_input_selector_switch):
* gst/selector/gstinputselector.h:
Various cleanups.
Added tags to the pads.
Select active pad based on the pad object instead of its name.
Fix refcount in set_active_pad.
Add property to get the number of pads.
* gst/selector/gstoutputselector.c:
(gst_output_selector_class_init),
(gst_output_selector_set_property),
(gst_output_selector_get_property):
Various cleanups.
Select the active pad based on the pad object instead of its name.
Fix locking when setting the active pad.
* gst/selector/gstselector-marshal.list:
* tests/check/elements/selector.c: (cleanup_pad),
(selector_set_active_pad), (run_input_selector_buffer_count):
Fixes for pad instead of padname for pad selection.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_init),
(gst_soup_http_src_finished_cb), (gst_soup_http_src_response_cb),
(gst_soup_http_src_build_message), (gst_soup_http_src_create):
* ext/soup/gstsouphttpsrc.h:
Try to resume on server disconnect. Fixes bug #522134.
Original commit message from CVS:
Based on patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
(join_rtcp_thread), (gst_rtp_session_change_state):
Avoid a deadlock when joining the RTCP thread in PAUSED because it might
be blocked downstream. Also avoid spawning multiple rtcp threads.
Fixes#520894.
Original commit message from CVS:
Patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Don't try to reset the clock skew when we have no timestamps.
Fixes#519005.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* ext/faad/gstfaad.c: (looks_like_valid_header):
Improve the header checking to look for what faad2 looks
for too. Fixes playback of same apple trailers.
Fixes bug #469979.
Original commit message from CVS:
* configure.ac:
Really check for libdc1394 >= 2.0.0, pkg-config thinks that
2.0.0-rcX is newer than 2.0.0 so we check for this too.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet dot be>
* configure.ac:
Clean up detection of different mjpegtoolsAPI versions.
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mpeg2enc/gstmpeg2enc.hh:
* ext/mpeg2enc/gstmpeg2encoder.cc:
* ext/mpeg2enc/gstmpeg2encoptions.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.cc:
* ext/mpeg2enc/gstmpeg2encpicturereader.hh:
* ext/mpeg2enc/gstmpeg2encstreamwriter.cc:
* ext/mpeg2enc/gstmpeg2encstreamwriter.hh:
Streamline conditional code for evolving mjpegtools API,
optimize and fix/prevent crash in log handling, use
names/nicks for enums in the usual way andm inor updates
in code and properties/settings. Partially fixes bug #520329.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Add parsing of cable delivery system descriptor.
Original commit message from CVS:
* configure.ac:
Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
plug-ins are included/excluded. (#498222)
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/mve/gstmvedemux.c: (gst_mve_audio_data),
(gst_mve_demux_get_type):
Fix audio discontinuity that happens when silent chunks are
followed by real data again. Fixes bug #519905.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
* sys/dvb/parsechannels.c:
Add DVB-C support. Special thanks to Christian Schaller
for a testing ground.
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Only send PMTs to program pads that the PMT is for even if
on same pid.
As a by-product, we now no longer hardcode any psi pid numbers.
Also remove pcr stream from old pmt when we apply a new pmt.
Original commit message from CVS:
Map Date-Time and GPS tags and Convert from EXIF to XMP Datatime as local time (those changes has been done in previous comit but had to be revert in 2008-02-10 due to frozen)
Original commit message from CVS:
* sys/dvb/camutils.c:
Don't free the program descriptors, this structure
containing them is stills tored after.
Fixes data corruption.
Original commit message from CVS:
Patch by: Daniel Fischer <dan at f3c dot com>
* configure.ac:
* ext/dc1394/gstdc1394.c: (gst_dc1394_change_state),
(gst_dc1394_get_cam_caps), (gst_dc1394_open_cam_with_best_caps):
* ext/dc1394/gstdc1394.h:
Add support for libdc1394 2.0.0 and above and require this version
now. Fixes bug #514964.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_class_init),
(gst_soup_http_src_init), (gst_soup_http_src_dispose),
(gst_soup_http_src_set_property), (gst_soup_http_src_get_property),
(gst_soup_http_src_create):
* ext/soup/gstsouphttpsrc.h:
* tests/check/elements/souphttpsrc.c: (run_test), (GST_START_TEST),
(souphttpsrc_suite):
Add support for specifying a list of cookies to be passed in
the HTTP request. Fixes bug #518722.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
* gst/selector/gstinputselector.h:
Added "select-all" property to make it work like aggregator in 0.8.
* gst/selector/gstoutputselector.c:
Fix resend-latest behavoiur.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/selector.c:
Add unit tests for selector.
Original commit message from CVS:
* configure.ac:
* ext/timidity/gsttimidity.c: (plugin_init):
* ext/timidity/gstwildmidi.c: (plugin_init):
Remove midi typefinders and require base CVS as they moved there.
Original commit message from CVS:
Patch by: Emilio Pozuelo Monfort <pochu at ubuntu dot com>
* ext/Makefile.am:
Build the wildmidi plugin if it's enabled and not only when
both, the timidity and wildmidi plugin, are enabled.
Original commit message from CVS:
* configure.ac:
* ext/mpeg2enc/Makefile.am:
* ext/soundtouch/Makefile.am:
* gst/modplug/Makefile.am:
Check for and define ERROR_CXXFLAGS and GST_CXXFLAGS and use them
when building C++ code.
Original commit message from CVS:
* sys/dvb/gstdvbsrc.c:
Add initial support for multiproto driver (not yet merged into
v4l-dvb mainline yet).
Only works for DVB-S not DVB-S2, DVB-T, DVB-C or other.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/inspect/plugin-gdp.xml:
* docs/plugins/inspect/plugin-quicktime.xml:
* docs/plugins/inspect/plugin-switch.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
Remove docs for elements that have moved to other modules
or been renamed.
Original commit message from CVS:
Patch by: Wouter Cloetens <wouter at mind dot be>
* configure.ac:
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_cancel_message),
(gst_soup_http_src_finished_cb), (gst_soup_http_src_chunk_free),
(gst_soup_http_src_chunk_allocator),
(gst_soup_http_src_got_chunk_cb), (gst_soup_http_src_create),
(gst_soup_http_src_start), (gst_soup_http_src_set_proxy):
* ext/soup/gstsouphttpsrc.h:
Implement zero-copy and make the buffer size configurable.
Prefix proxy URIs with "http://" if they don't start with it
already and catch errors earlier, fixes hanging in some situations.
Fixes bug #514948.
Original commit message from CVS:
* tests/check/gst-plugins-bad.supp:
Add suppressions for SoundTouch valgrind warnings and
a valgrind warning caused by the LADSPA sine plugin and
happening on every exit().
Remove GIO suppressions as it's now in -base.
Original commit message from CVS:
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_create):
Don't allocate and copy the data to a new place but instead
put the data from gmyth (which we own) into the buffers that
are passed downstream.
Original commit message from CVS:
Based on a patch by:
Hans de Goede <j dot w dot r dot degoede at hhs dot nl>
* configure.ac:
* ext/mpeg2enc/gstmpeg2enc.cc:
Remove hack to work with mjpegtools 1.9.0rc3 and above and instead
use mjpeg_loglev_t() for getting the log levels. Check for this
function in configure.ac as the pkg-config file doesn't tell us
which release candidate we have. Fixes bug #517896.
Original commit message from CVS:
* tests/check/Makefile.am:
Ignore some more elements for the states unit test, like
dfbvideosink which produces a segfault. Fixes bug #517854.
Original commit message from CVS:
2008-02-20 Bastien Nocera <hadess@hadess.net>
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_do_seek),
(gst_mythtv_src_start): Using the wrong GstFormat for the filesize,
and fail seek properly on anything but _BYTES format
Fixes bug #517684
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
Ignore streams that did not receive an SR packet when doing
synchronisation. Fixes#516160.
Original commit message from CVS:
* gst/dvdspu/gstdvdspu.c: (gst_dvd_spu_handle_new_spu_buf):
Set n_line_ctrl_i to 0 whenever we free line_ctrl_i. Patch based
on an idea by Jan Schmidt, fixes bug #516436.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.c:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtsparse.c:
Make sure the gstmpegdesc debug lines do not critical
when GST_DEBUG is enabled and also actually output.
Thanks to Alessandro Decina for spotting.
Fixes#516448
Original commit message from CVS:
* configure.ac:
Generate the directshow Makefiles so that the directories
get disted. Still needs some configure time detection to enable
building them under MingW.
Original commit message from CVS:
* ext/metadata/Makefile.am:
Don't install a header file. We will have to merge these
tags into libgsttag after the release and use them from there.
Fixes: #515860
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
Add Makefiles to win32 plugins and lib.
They will need to be tested and probably fixed by developers
working with mingw. This is a first step to include source files
with releases.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-gio.xml:
Remove documentation for the GIO plugin as it was moved to
gst-plugins-base. Fixes bug #515964.
Original commit message from CVS:
* ext/xvid/gstxvidenc.c:
* gst/vmnc/vmncdec.c:
* sys/glsink/glimagesink.c:
* sys/glsink/gstgldisplay.c:
Fix some finalize leaks by chaining up to the parent method.
Original commit message from CVS:
* sys/fbdev/gstfbdevsink.c: (gst_fbdevsink_class_init),
(gst_fbdevsink_finalize):
Free the device string in finalize. Fixes bug #515722.
Original commit message from CVS:
* gst/selector/Makefile.am:
Listing the marshal.h in the nodist_HEADERS breaks distcheck, so
let's not do that
* tests/check/Makefile.am:
Disable the crashing cdaudio plugin from the states test so I can make
pre-releases.
Original commit message from CVS:
* sys/dvb/Makefile.am:
* sys/dvb/dvbbasebin.c:
Add URI Handler for dvb.
Re-order pad templates to workaround a bug in playbasebin.
* sys/dvb/parsechannels.c:
* sys/dvb/parsechannels.h:
Add code to parse channels from zap-style channels.conf files.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
Use g_file_[sg]et_contents() instead of using stdio functions.
Should be less error prone.
* tests/check/elements/multifile.c:
Create a temporary directory using standard functions instead of
creating a directory in the current dir.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Remove equalizer plugin docs
* tests/check/Makefile.am:
Add GST_OPTION_CFLAGS, to get -Werror -Wall into the tests as for
other modules.
* tests/check/elements/multifile.c:
* tests/check/elements/rganalysis.c:
* tests/check/elements/rglimiter.c:
Fix compiler warnings from -Wall -Werror
Original commit message from CVS:
* configure.ac:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-xingheader.xml:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c:
* tests/check/elements/xingmux_testdata.h:
Remove the xingmux plugin, as the element has moved into
mpegaudioparse in -ugly.
Original commit message from CVS:
* ext\neon\gstneonhttpsrc.c:
Include unistd.h only if _HAVE_UNISTD_H is defined
* gst\mpegvideoparse\mpegvideoparse.c:
Use G_GUINT64_CONSTANT GLIB macro for constant
* sys\dshowsrcwrapper\gstdshowaudiosrc.c:
* sys\dshowsrcwrapper\gstdshowvideosrc.c:
* sys\dshowdecwrapper\gstdshowaudiodec.c:
* sys\dshowdecwrapper\gstdshowaudiodec.h:
* sys\dshowdecwrapper\gstdshowdecwrapper.c:
* sys\dshowdecwrapper\gstdshowdecwrapper.h:
* sys\dshowdecwrapper\gstdshowvideodec.c
* sys\dshowdecwrapper\gstdshowvideodec.h:
Add a DirectShow decoder wrapper.
* win32\MANIFEST:
Add new win32 files to MANIFEST
* win32\vs6\gst_plugins_bad.dsw:
* win32\vs6\libgstdshow.dsp:
* win32\vs6\libgstdshowdecwrapper.dsp:
* win32\vs6\libgstflv.dsp:
Add new projects to bad workspace
