Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Added property to post a message on timeout.
Updated docs.
When restarting the select, initialize the fdsets again.
Init control sockets so we don't accidentally close a random socket.
API: GstUDPSrc::timeout property

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
Fix flag registration.
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
Reading 0 also means 'no more commands'

Antoine Tremblay authored Sep 29, 2006 and Wim Taymans committed Sep 29, 2006

gst/udp/gstudpsrc.c: Fix possible infinite loop when shutting down, a read can also return 0 to indicate no more mess...

Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Fix possible infinite loop when shutting down, a read can also return
0 to indicate no more messages are available. Fixes #358156.

Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_all_caps),
(gst_v4l2src_get_caps):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
Framerate can be 0/1 too.
Init framerate to 0/1 before querying it so that we can detect
devices that don't know about a framerate.
Add some more debugging info.

Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add support for 'yv12' fourcc.

Original commit message from CVS:
Removed set-undef-fps.

Original commit message from CVS:
* sys/v4l2/gstv4l2object.c:
(gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new),
(gst_v4l2_object_set_property_helper),
(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_create):
* sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_contains_channel),
(gst_v4l2_tuner_list_channels),
(gst_v4l2_tuner_set_channel_and_notify),
(gst_v4l2_tuner_get_channel), (gst_v4l2_tuner_contains_norm),
(gst_v4l2_tuner_list_norms), (gst_v4l2_tuner_set_norm_and_notify),
(gst_v4l2_tuner_get_norm):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
(gst_v4l2_fill_lists), (gst_v4l2_empty_lists):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_get_fps):
Renamed some properties to match the tuner interface naming.

Original commit message from CVS:
* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_set_property_helper),
(gst_v4l2_set_defaults):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
(gst_v4l2src_create):
* sys/v4l2/gstv4l2xoverlay.c: (gst_v4l2_xoverlay_open):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
(gst_v4l2_fill_lists), (gst_v4l2_open), (gst_v4l2_set_norm),
(gst_v4l2_get_frequency), (gst_v4l2_set_frequency),
(gst_v4l2_signal_strength), (gst_v4l2_get_attribute),
(gst_v4l2_set_attribute), (gst_v4l2_get_input),
(gst_v4l2_set_input):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_get_capture),
(gst_v4l2src_set_capture), (gst_v4l2src_capture_init),
(gst_v4l2src_capture_start), (gst_v4l2src_capture_stop),
(gst_v4l2src_buffer_new):
* tests/icles/v4l2src-test.c: (my_bus_callback), (main):
Small cleanups.
Fix error messages.
Use locks when getting timestamps.
Fix leaks in test.
Add licensing header to tests.

Original commit message from CVS:
Some cleanups and comments.

Original commit message from CVS:
add audiofx plugin

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:
Add v4l2 plugin to the docs.
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
(gst_v4l2src_get_mmap), (gst_v4l2src_create):
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/gstv4l2vidorient.c:
Fix docs.
Remove some more externs.

Original commit message from CVS:
* sys/v4l2/Makefile.am:
Fix makefile, list libs in stack order.
* sys/v4l2/gstv4l2colorbalance.c:
* sys/v4l2/gstv4l2colorbalance.h:
* sys/v4l2/gstv4l2object.c: (gst_v4l2_device_get_type),
(gst_v4l2_object_install_properties_helper):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_get_read),
(gst_v4l2src_get_mmap), (gst_v4l2src_create):
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/gstv4l2tuner.h:
* sys/v4l2/gstv4l2vidorient.h:
* sys/v4l2/gstv4l2xoverlay.h:
* sys/v4l2/v4l2_calls.h:
* sys/v4l2/v4l2src_calls.h:
Fix coding style:
- Remove extern from functions.
- Fix header indentation.
Fix Flags, add defaults for properties.
Remove unused enums.
Fix TOO_LAZY in error messages.

sys/v4l2/: Fix pass at code cleanups, move errors cases out of the normal flow for additional code clarity.

Original commit message from CVS:
* sys/v4l2/gstv4l2object.c: (gst_v4l2_class_probe_devices),
(gst_v4l2_probe_needs_probe),
(gst_v4l2_object_install_properties_helper), (gst_v4l2_object_new),
(gst_v4l2_object_destroy), (gst_v4l2_object_set_property_helper),
(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults),
(gst_v4l2_object_start), (gst_v4l2_object_stop):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_init), (gst_v4l2src_dispose),
(gst_v4l2src_set_property), (gst_v4l2src_get_property),
(gst_v4l2src_fixate), (gst_v4l2src_get_caps),
(gst_v4l2src_set_caps), (gst_v4l2src_get_read),
(gst_v4l2src_get_mmap), (gst_v4l2src_create):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
(gst_v4l2_open), (gst_v4l2_close), (gst_v4l2_get_norm),
(gst_v4l2_set_norm), (gst_v4l2_get_frequency),
(gst_v4l2_set_frequency), (gst_v4l2_signal_strength),
(gst_v4l2_get_attribute), (gst_v4l2_set_attribute),
(gst_v4l2_get_input), (gst_v4l2_set_input):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
(gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
(gst_v4l2src_capture_stop), (gst_v4l2src_capture_deinit),
(gst_v4l2src_get_size_limits), (gst_v4l2src_set_fps),
(gst_v4l2src_get_fps), (gst_v4l2src_buffer_finalize),
(gst_v4l2src_buffer_new):
Fix pass at code cleanups, move errors cases out of the normal
flow for additional code clarity.

Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
(gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
Small cleanups.
don't try to set "sync" property when it is not available.

Peter Kjellerstedt authored Sep 25, 2006 and Tim-Philipp Müller committed Sep 25, 2006

Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/alpha/gstalpha.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstudpsrc.c:
* gst/videomixer/videomixer.c:
Include stdlib.h in some more places, makes things compile
with uClibc and -Werror (#357592).

ext/jpeg/gstjpegdec.c: our code should handle that fine. Some of the buttons on the apple trailer site are apparently...

Original commit message from CVS:
* ext/jpeg/gstjpegdec.c:
Set minimum height to 8 (from 16), our code should handle
that fine. Some of the buttons on the apple trailer site
are apparently only 15 pixels high (see #357470).

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_open):
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Improve error reporting.

Original commit message from CVS:
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_plugin_init):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
Fix klass typos.
Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.

Original commit message from CVS:
* configure.ac:
Need  -base CVS for gst_base_rtp_depayload_push_ts().

gst/avi/gstavidemux.c: Don't check for a tag that is never there and check if we read the correct tag. Fixes seeking ...

Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
Don't check for a tag that is never there and check if we read the
correct tag. Fixes seeking again.
We must post an error when all pads are unlinked.

Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
(gst_rtp_vorbis_pay_reset_packet),
(gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
(gst_rtp_vorbis_pay_handle_buffer):
More fixage, set endoder-params correctly in the payloader.

Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
Make static pad templates static to appease valgrind's leak
detector.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/autodetect.c: (GST_START_TEST),
(autodetect_suite):
Add simple test for the ghostpad lockup on shutdown fixed in core
CVS (audio bit disabled because it would need dozens of alsa
suppressions and I'm too lazy to add those now).

Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.

Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
Fix profile-level-id parsing and setup.

Original commit message from CVS:
* gst/udp/README:
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Update README, simple cleanup.

Original commit message from CVS:
* gst/rtp/README:
Update README with some examples.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtpmp4gpay.h:
Make optional RTP parameters of type STRING, as required by the
application/x-rtp caps specification.

Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
Correctly calculate size of each H263+ RTP buffer taking into account MTU and
RTP header.

Original commit message from CVS:
* gst/rtp/Makefile.am:
And makefile too.

Original commit message from CVS:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
(gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
(decode_base64), (gst_rtp_asf_depay_setcaps),
(gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
(gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
(gst_rtp_asf_depay_plugin_init):
* gst/rtp/gstrtpasfdepay.h:
Added preliminary ASF depayloader.
* gst/rtp/gstrtph264depay.c: (decode_base64):
Fix base64 decoding.

Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.

Original commit message from CVS:
Just a small fix to the app options.

Original commit message from CVS:
Add Video Orientation interface support to v4l2src.

Original commit message from CVS:
* gst/rtsp/test.c: (main):
Fix build.

Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Add ms-gsm to the src template.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.

Original commit message from CVS:
Fix GST_BUFFER_DURATION.

gst/rtsp/gstrtspsrc.*: Export sometimes source pad with correct caps on the template, create the ghostpad from the te...

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Export sometimes source pad with correct caps on the template, create
the ghostpad from the template.
Remove RTCP template as we never expose RTCP.
Protect against invalid body size.
Avoid memcpy when creating the output buffer.
Properly post an error and send EOS when the loop function is shut down.

Lutz Mueller authored Sep 18, 2006 and Wim Taymans committed Sep 18, 2006

Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Make sure we can never set an invalid location.
* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
* gst/rtsp/rtspmessage.h:
Added _steal_body method for future use.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
Make freeing of NULL url return immediatly.

Lutz Mueller authored Sep 18, 2006 and Wim Taymans committed Sep 18, 2006

Original commit message from CVS:
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Use boilerplate.
Make rtspsrc subclass GstBin to make state changes easier.
Add Range header field on the PLAY request.