| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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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.
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create UDP sources and connect to the multica...
Original commit message from CVS:
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes #349894.
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element ...
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
There is no taglibmux element ...
* gst/rtsp/gstrtspsrc.c:
Use '%' rather than '&perc;' in gtk-doc blurb, docs build
was complaining about unknown entity here.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
* sys/oss/gstosssink.c: (gst_oss_sink_open),
(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
Small documentation updates.
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Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
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Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_send):
Remove unwanted DEBUG line.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
Don't try doing state changes on a NULL pointer.
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Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.h:
replaced closesocket and close in code with one CLOSE_SOCKET.
Some more cleanups. Fixes #345301.
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fixes.
Original commit message from CVS:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/annodex/gstcmmlparser.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/gdk_pixbuf/pixbufscale.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalphacolor.c:
* gst/cutter/gstcutter.c:
* gst/debug/gstnavigationtest.c:
* gst/icydemux/gsticydemux.c:
* gst/level/gstlevel.c:
* gst/multipart/multipart.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstvideoflip.c:
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
plus two minor macro fixes.
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Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
Added documentation for the rtsp plugin. Fixes #345393.
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Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close), (rtsp_connection_free):
Use better G_OS_* macros. Fixes #345301 some more.
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Original commit message from CVS:
Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
(rtsp_connection_close):
Make RTSP plugin compile on windows. Fixes #345301.
Some changes to original patch to catch errors better.
use ifdef WIN32 instead of ifndef.
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Original commit message from CVS:
* ext/aalib/gstaasink.h:
* ext/annodex/gstcmmldec.h:
* ext/cairo/gsttimeoverlay.h:
* ext/dv/gstdvdec.h:
* ext/dv/gstdvdemux.h:
* ext/esd/esdmon.h:
* ext/esd/esdsink.h:
* ext/flac/gstflacenc.h:
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstgconfaudiosrc.h:
* ext/gconf/gstgconfvideosink.h:
* ext/gconf/gstgconfvideosrc.h:
* ext/gdk_pixbuf/gstgdkanimation.h:
* ext/gdk_pixbuf/pixbufscale.h:
* ext/hal/gsthalaudiosink.h:
* ext/hal/gsthalaudiosrc.h:
* ext/jpeg/gstjpegenc.h:
* ext/jpeg/gstsmokedec.h:
* ext/jpeg/gstsmokeenc.h:
* ext/libcaca/gstcacasink.h:
* ext/libmng/gstmngdec.h:
* ext/libmng/gstmngenc.h:
* ext/libpng/gstpngdec.h:
* ext/libpng/gstpngenc.h:
* ext/raw1394/gstdv1394src.h:
* ext/speex/gstspeexenc.h:
* gst/autodetect/gstautoaudiosink.h:
* gst/autodetect/gstautovideosink.h:
* gst/avi/gstavidemux.h:
* gst/cutter/gstcutter.h:
* gst/debug/efence.h:
* gst/debug/gstnavigationtest.h:
* gst/debug/gstnavseek.h:
* gst/flx/gstflxdec.h:
* gst/goom/gstgoom.h:
* gst/icydemux/gsticydemux.h:
* gst/id3demux/gstid3demux.h:
* gst/law/alaw-decode.h:
* gst/law/alaw-encode.h:
* gst/law/mulaw-decode.h:
* gst/law/mulaw-encode.h:
* gst/matroska/matroska-mux.h:
* gst/median/gstmedian.h:
* gst/oldcore/gstaggregator.h:
* gst/oldcore/gstfdsink.h:
* gst/oldcore/gstmd5sink.h:
* gst/oldcore/gstmultifilesrc.h:
* gst/oldcore/gstpipefilter.h:
* gst/oldcore/gstshaper.h:
* gst/oldcore/gststatistics.h:
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.h:
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.h:
* gst/rtp/gstrtpdepay.h:
* gst/rtp/gstrtpgsmdepay.h:
* gst/rtp/gstrtpgsmpay.h:
* gst/rtp/gstrtph263pay.h:
* gst/rtp/gstrtph263pdepay.h:
* gst/rtp/gstrtph263ppay.h:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtpmp4vdepay.h:
* gst/rtp/gstrtpmp4vpay.h:
* gst/rtp/gstrtpmpadepay.h:
* gst/rtp/gstrtpmpapay.h:
* gst/rtp/gstrtppcmadepay.h:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmudepay.h:
* gst/rtp/gstrtppcmupay.h:
* gst/rtp/gstrtpspeexdepay.h:
* gst/rtp/gstrtpspeexpay.h:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtspsrc.h:
* gst/smpte/gstsmpte.h:
* gst/udp/gstdynudpsink.h:
* gst/udp/gstmultiudpsink.h:
* gst/udp/gstudpsink.h:
* gst/udp/gstudpsrc.h:
* gst/videofilter/gstvideobalance.h:
* gst/videofilter/gstvideoflip.h:
* sys/oss/gstossdmabuffer.h:
* sys/oss/gstossmixerelement.h:
* sys/oss/gstosssink.h:
* sys/oss/gstosssrc.h:
* sys/osxvideo/osxvideosink.h:
* sys/sunaudio/gstsunaudiomixer.h:
* sys/sunaudio/gstsunaudiosink.h:
* sys/ximage/gstximagesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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Original commit message from CVS:
* ext/esd/esdmon.c: (gst_esdmon_depths_get_type),
(gst_esdmon_channels_get_type):
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_profile_get_type):
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type):
* ext/libcaca/gstcacasink.c: (gst_cacasink_dither_get_type):
* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type):
* gst/alpha/gstalpha.c: (gst_alpha_method_get_type):
* gst/rtp/gstrtpilbcdepay.c: (gst_ilbc_mode_get_type):
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
* gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type):
* gst/videofilter/gstvideoflip.c: (gst_video_flip_method_get_type):
* gst/videomixer/videomixer.c:
(gst_video_mixer_background_get_type):
Const-ify GEnumValue arrays.
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Original commit message from CVS:
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Make parsing of urls suck slightly less.
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Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
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Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_class_init):
* ext/esd/esdsink.c: (gst_esdsink_class_init):
* ext/flac/gstflactag.c: (gst_flac_tag_class_init):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
* ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
* gst/alpha/gstalpha.c: (gst_alpha_class_init):
* gst/avi/gstavimux.c: (gst_avimux_class_init):
* gst/debug/efence.c: (gst_efence_class_init):
* gst/debug/negotiation.c: (gst_negotiation_class_init):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
* gst/goom/gstgoom.c: (gst_goom_class_init):
* gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
* gst/interleave/deinterleave.c: (deinterleave_class_init):
* gst/interleave/interleave.c: (interleave_class_init):
* gst/law/alaw-decode.c: (gst_alawdec_class_init):
* gst/law/alaw-encode.c: (gst_alawenc_class_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
* gst/median/gstmedian.c: (gst_median_class_init):
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
* gst/smpte/gstsmpte.c: (gst_smpte_class_init):
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
* gst/videomixer/videomixer.c: (gst_videomixer_class_init):
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
* sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
* sys/osxaudio/gstosxaudioelement.c:
(gst_osxaudioelement_class_init):
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
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on it.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
the OPTIONS request result is optional so don't
fail on it.
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Original commit message from CVS:
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_init):
* ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_add_pads):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
(gst_jpeg_dec_setcaps):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_init):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init):
* ext/libmng/gstmngdec.c: (gst_mngdec_init),
(gst_mngdec_src_getcaps):
* ext/libpng/gstpngdec.c: (gst_pngdec_init),
(gst_pngdec_caps_create_and_set):
* ext/libpng/gstpngenc.c: (gst_pngenc_init):
* ext/mikmod/gstmikmod.c: (gst_mikmod_init):
* ext/speex/gstspeexdec.c: (gst_speex_dec_init):
* gst/alpha/gstalpha.c: (gst_alpha_init):
* gst/auparse/gstauparse.c: (gst_au_parse_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_init),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream):
* gst/cutter/gstcutter.c: (gst_cutter_init):
* gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
(gst_efence_checkgetrange):
* gst/debug/negotiation.c: (gst_negotiation_init):
* gst/flx/gstflxdec.c: (gst_flxdec_init):
* gst/goom/gstgoom.c: (gst_goom_init):
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_init):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_init):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init):
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_init):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_init):
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init):
* gst/smpte/gstsmpte.c: (gst_smpte_init):
* gst/wavparse/gstwavparse.c: (gst_wavparse_init),
(gst_wavparse_create_sourcepad):
Fix memleak with gst_static_pad_template_get().
This uses gst_pad_new_from_static_template() instead.
Fixes #333512
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Original commit message from CVS:
* gst/rtsp/README:
Updated README.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
* gst/rtsp/gstrtspsrc.h:
Make sure the RTP port is an even port an try to allocate
another if not.
Added retry property to control max retries for port allocation.
Make sure RTCP port is RTP port+1.
Cleanup when port allocation fails.
Fixes #319183.
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copy-and-paste paradigm leading to duplication of...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
Add comment in a fultile attempt to stop the copy-and-paste
paradigm leading to duplication of bad code.
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Mime parameters have to be checked case insensitive
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Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type),
(gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp),
(gst_rtpdec_set_property), (gst_rtpdec_get_property),
(gst_rtpdec_change_state):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
* gst/rtsp/rtspconnection.c: (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_dump):
Resurected rtpdec to make rtspsrc happy again.
Skip attributes from the session id.
Don't crash when dumping a message with an empty body.
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Original commit message from CVS:
* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
* gst/rtsp/sdpmessage.h:
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_render):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_get_stats):
C89 fixes: declare variables at the beginning of a block and
make gcc-2.9x happy (#328264; patch by: Jens Granseuer
<jensgr at gmx dot net>).
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include to make things work on OpenBSD a...
Original commit message from CVS:
* gst/rtsp/rtspconnection.c:
Add <netinet/in.h> include and move <arpa/inet.h> include
to make things work on OpenBSD as well (fixes #323717;
patch by: Benjamin Pineau)
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Original commit message from CVS:
expand tabs
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Original commit message from CVS:
* configure.ac:
fix up GST_PLUGIN_LDFLAGS
* gst/rtsp/rtspconnection.c:
fix includes (see #317043)
* gst/videofilter/Makefile.am:
stop installing this library
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Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (read_body):
Apply patch from Sebastien Cote to fix #319184.
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Original commit message from CVS:
rework configure.ac; make asterisk rtp stuff compile on mingw
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Original commit message from CVS:
fix build and use of GST_LIBS
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_send):
Strip spaces for key/value pairs.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_change_state):
More SDP parsing and caps setting.
Do NO_PREROLL differently.
add pads only after negotiated.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_getcaps):
Implement the getcaps function.
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Original commit message from CVS:
* gst/rtp/README:
Update README
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
Make extra params as strings.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
Make state change return NO_PREROLL as this is a live
source.
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Don't unref old caps when NULL.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send),
(gst_rtspsrc_open), (gst_rtspsrc_uri_get_type),
(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri),
(gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init):
* gst/rtsp/sdpmessage.c: (sdp_media_get_format):
* gst/rtsp/sdpmessage.h:
Add URI handler.
Parse SDP and create caps.
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Original commit message from CVS:
2005-09-02 Andy Wingo <wingo@pobox.com>
* All plugins updated for element state changes.
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Original commit message from CVS:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps):
* ext/amrnb/amrnbparse.c:
Update caps with audio/AMR.
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
(gst_rtpamrdec_change_state):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
Dont set FT headers twice, it was already in the encoded
bitstream.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
* gst/rtsp/rtspconnection.c: (parse_line):
Cleanups
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Added caps property, we need this soon to type the buffers.
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Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_chain):
Fix up amr depayloader a bit.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Look for options result in Public and Allow header fields..
spec says Allow but some servers return Public...
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Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property):
* gst/rtp/gstrtpamrenc.h:
Added payload_type and ssrc properties to the payloader.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Options need to be stripped and are in the Public header field.
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix url / parsing...
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open), (gst_rtspsrc_play):
Support absolute control urls too.
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Original commit message from CVS:
* gst/rtsp/README:
Added rtsp server implementation docs.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_add_element),
(gst_rtspsrc_play):
Fix for core changes.
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Original commit message from CVS:
2005-06-29 Andy Wingo <wingo@pobox.com>
* gst/udp/gstudpsink.c (gst_udpsink_get_type): Actually add the
URI handler.
* gst/udp/gstudpsrc.c (gst_udpsrc_start):
(gst_udpsrc_create): Signedness.
* gst/rtsp/sdpmessage.c (sdp_message_parse_buffer): Thanks
compiler!
(sdp_parse_line): Signedness fix.
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Original commit message from CVS:
2005-06-29 Andy Wingo <wingo@pobox.com>
* configure.ac (GST_CFLAGS): GCC strikes back!!! Let the build
breakage ensue!!!
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_loop, gst_rtspsrc_open):
Signedness, unused var fixes.
(gst_rtspsrc_close): Unused?
* gst/realmedia/rmdemux.c (re_hexdump_bytes): Unused.
* gst/law/mulaw-encode.c (gst_mulawenc_chain): Signeness fix.
* gst/law/alaw-encode.c (alawenc_getcaps): Remove unneeded
declarations. Typo (probably crasher) fix.
* gst/law/mulaw-encode.c (mulawdec_getcaps):
* gst/law/mulaw-encode.c (mulawenc_getcaps):
* gst/law/alaw-decode.c (alawdec_getcaps): Same crasher fix.
* gst/goom/gstgoom.c (gst_goom_init): Hook up the event function.
* gst/effectv/gstwarp.c (gst_warptv_setup): Signedness fix.
* gst/effectv/gstdice.c (gst_dicetv_draw): Um, deferencing
uninitialized pointer not good.
* gst/videofilter/gstvideoexample.c (plugin_init):
* gst/videofilter/Makefile.am (libgstvideoexample_la_LIBADD): Link
to libgstvideofilter instead of gst_library_load.
* gst/alpha/gstalpha.c (gst_alpha_chroma_key_i420)
(gst_alpha_chroma_key_ayuv): Signedness fixen.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
Fix case where outpad could not be decided.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Make rtspsrc a live source.
Don't try to parse NULL urls.
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Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.
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Original commit message from CVS:
add rtsp and esd files to spec and fix some disting issues
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Original commit message from CVS:
* gst/rtsp/README:
* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_get_type),
(gst_tcpclientsrc_base_init), (gst_tcpclientsrc_class_init),
(gst_tcpclientsrc_init), (gst_tcpclientsrc_getcaps),
(gst_tcpclientsrc_stop), (gst_tcpclientsrc_eos),
(gst_tcpclientsrc_create), (gst_tcpclientsrc_start):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_get_type),
(gst_tcpserversrc_base_init), (gst_tcpserversrc_class_init),
(gst_tcpserversrc_init), (gst_tcpserversrc_create),
(gst_tcpserversrc_start), (gst_tcpserversrc_stop):
* gst/tcp/gsttcpserversrc.h:
* gst/tcp/gsttcpsrc.c: (gst_tcpsrc_get_type),
(gst_tcpsrc_base_init), (gst_tcpsrc_class_init), (gst_tcpsrc_init),
(gst_tcpsrc_create), (gst_tcpsrc_start), (gst_tcpsrc_stop):
* gst/tcp/gsttcpsrc.h:
* gst/udp/gstudpsink.c: (gst_udpsink_base_init),
(gst_udpsink_init), (gst_udpsink_get_times), (gst_udpsink_render),
(gst_udpsink_set_property), (gst_udpsink_get_property),
(gst_udpsink_change_state):
* gst/udp/gstudpsink.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_get_type),
(gst_udpsrc_base_init), (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_uri), (gst_udpsrc_start),
(gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Make UDP and TCP elements use PushSrc.
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Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play):
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_send), (read_line), (parse_request_line),
(parse_line), (read_body), (rtsp_connection_receive),
(rtsp_connection_free):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c: (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (rtsp_message_set_body),
(rtsp_message_take_body):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtspstream.h:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
Added README
Some cleanups.
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