Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_udp_sink):
Fix default clock-rate for realmedia.
Fix parsing of transport.
Don't try to link NULL pads.
Original commit message from CVS:
* po/POTFILES.skip:
Add POTFILES.skip with list of source files that aren't disted at the
moment but contain translatable strings. Should hopefully pacify
broken tools and make it clearer that these files are left out
intentionally (#461600).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
If the buffer was entirely clipped ... don't try sending it :)
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports):
If we don't hav a session manager, set the caps on outgoing buffers
ourselves.
Force PAUSE/PLAY methods for now until the extensions can overwrite.
Append final bit of the transport string even when it does not contain a
placeholder.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
(gst_rtsp_ext_list_connect):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
Clean up the interface list.
Allow connecting to interface signals for the extensions.
Remove old extension code.
Free list on cleanup.
Allow extensions to send additional RTSP messages.
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
Handle a NULL gconf key gracefully by rendering the default element.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Don't save format information ourselves, this is already saved in
GstAudioFilter.
Original commit message from CVS:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Use rank to filter out extensions.
Add url to stream_select interface call.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't unref the outgoing buffer twice when dropping it because it's
outside of the segment.
Original commit message from CVS:
* configure.ac:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
Use the new buffer clipping function from gstaudio here and
require gst-plugins-base CVS.
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
For framed Wavpack buffers we require a valid timestamp.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
Clip raw audio and video when we can, keep track of current output
segment.
Don't leak buffers and events when there is no output pad.
Improve debugging here and there.
Original commit message from CVS:
Patch by: Alexander Eichner <alexeichi@yahoo.de>
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
Use define here.
* sys/v4l2/gstv4l2tuner.c:
(gst_v4l2_tuner_set_frequency_and_notify):
Don't touch the property - its still disabled.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
(gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
* sys/v4l2/v4l2src_calls.h:
Improve fallback format negotionation. Fixes#451388
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
Fix parsing of esds atoms inside mp4a atoms so that we can set correct
codec_info for AAC audio. Fixes#457097 along with a whole other bunch
of qt/aac files.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c:
(gst_wavpack_dec_clip_outgoing_buffer):
Fix buffer clipping to correctly clip to the segment stop.
Original commit message from CVS:
* configure.ac:
* tests/Makefile.am:
Remove bogus check for libcheck, since we check for
gstreamer-check and it pulls in the required info from there,
and we weren't actually _using_ the information for libcheck
ourselves anyway.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
don't have enough granularity to convert that boolean into a
GstFlowReturn.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
Remove endianness-flipping hack that seems to have been required
only because of a bug in ffmpegcolorspace.
Partially Fixes: #451908
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
Set the encoding-name in the rtp caps to all uppercase, as required by
the caps spec.
Some small cleanups in the error paths. Fixes#453037.
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c:
(gst_wavpack_parse_index_get_last_entry),
(gst_wavpack_parse_index_get_entry_from_sample),
(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
(gst_wavpack_parse_scan_to_find_sample):
* ext/wavpack/gstwavpackparse.h:
Use a GSList for the GArray that is used like a list anyway.
Original commit message from CVS:
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
Add state change function where we set 0/1 as default framerate in
case our setcaps function isn't called, like it might not in a
filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
gdkpixbufdec trying to create caps with a 0/0 framerate.
Also post an error message on the bus if gst_pad_push() fails when
called from our sink event handler (+1 for flow returns for event
functions in 0.11) instead of failing silently.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (gst_rtspsrc_setup_streams):
* gst/rtsp/gstrtspsrc.h:
For container formats we only need to activate one of the streams so
that we correctly signal no-more-pads. Fixes#451015.
