Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain),
(close_library), (open_library),
(gst_real_audio_dec_probe_modules), (gst_real_audio_dec_getcaps),
(gst_real_audio_dec_setcaps), (gst_real_audio_dec_init),
(gst_real_audio_dec_change_state), (gst_real_audio_dec_finalize):
Add raversions we can support on the caps.
Refactor the loading of the real codecs like realvideo so that we can
implement probing.
Probe all supported formats by trying to load the .so files, only report
the versions on the caps that we can actually load.
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_getcaps), (gst_real_video_dec_setcaps),
(open_library), (close_library),
(gst_real_video_dec_probe_modules),
(gst_real_video_dec_change_state), (gst_real_video_dec_init),
(gst_real_video_dec_finalize), (gst_real_video_dec_class_init):
* gst/real/gstrealvideodec.h:
Change the loading of the library like the audio decoder.
Probe the supported formats by trying to load the .so files and only
report the versions on the caps that we can actually load.
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegtsparse/mpegtspacketizer.c:
Handle character sets in strings coming from DVB SI according
to the DVB SI spec.
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..
