Age | Commit message (Collapse) | Author | Files | Lines |
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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Revert unpopular change for GST_MESSAGE_SRC to GObject.
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Original commit message from CVS:
add missing files
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Original commit message from CVS:
2005-08-25 Andy Wingo <wingo@pobox.com>
* ext/ladspa/gstladspa.h:
* ext/ladspa/gstladspa.c: Halfway-ported. Doesn't compile yet.
* ext/ladspa/gstsignalprocessor.h:
* ext/ladspa/gstsignalprocessor.c: New files, the start of a base
class for DSP elements.
* configure.ac: Sort the external libs checks, add a ladspa check,
output the ladspa makefile.
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Original commit message from CVS:
Fixed EOS and improved robustness for malformed indices.
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Original commit message from CVS:
add lame
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Original commit message from CVS:
fix broken header setup in Makefile.am
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Original commit message from CVS:
dist more
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timestamps.
Original commit message from CVS:
2005-08-24 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid
segment end timestamps.
(Also commit an old changelog entry)
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Original commit message from CVS:
port audioresample to basetransform
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Original commit message from CVS:
enable more; update for basetransform
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Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
GST_MESSAGE_SRC became a GObject
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Original commit message from CVS:
fix speex entry
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Original commit message from CVS:
add speex to spec file and remove gstosslement from POTFILES.in
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Original commit message from CVS:
* ext/speex/gstspeexenc.h:
Fixed include path of adapter
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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
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Original commit message from CVS:
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
Fix property warning.
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Original commit message from CVS:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Small updates, RFC reference to payload encoders.
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otherwise.
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/speex/Makefile.am:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexdec.c: (speex_get_query_types),
(gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event),
(speex_dec_event), (speex_dec_chain):
Port speexdec. Leads to some unfamiliar warnings on console,
but works otherwise.
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opening the mixer.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
property after opening the mixer.
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.c:
* sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossmixerelement.h:
* sys/oss/gstossmixerelement.c: Added mixer element like
alsamixer.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Register the ossmixer element.
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Original commit message from CVS:
changelog
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Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssrc.h:
* sys/oss/gstosssrc.c: Totally ported, dude.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Add osssrc.
* sys/oss/gstosssink.c: We do native byte order.
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Original commit message from CVS:
Fixed mishandling events and incorrect audio skipping after seek.
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use get_range instead of this seeking nasti...
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init),
(gst_id3_tag_sink_event), (gst_id3_tag_do_caps_nego),
(gst_id3_tag_chain), (gst_id3_tag_change_state), (plugin_init):
Works a bit better now, but still needs a rewrite to use
get_range instead of this seeking nastiness.
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Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/flac/Makefile.am:
* ext/flac/gstflac.c: (plugin_init):
* ext/flac/gstflacdec.c: (flacdec_get_type), (gst_flacdec_init),
(gst_flacdec_update_metadata), (gst_flacdec_seek),
(gst_flacdec_tell), (gst_flacdec_length), (gst_flacdec_read),
(gst_flacdec_write), (gst_flacdec_loop),
(gst_flacdec_get_src_query_types), (gst_flacdec_src_query),
(gst_flacdec_src_event), (gst_flacdec_sink_activate),
(gst_flacdec_sink_activate_pull), (gst_flacdec_change_state):
* ext/flac/gstflacdec.h:
Port flacdec (seeking is still slow'ish).
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Original commit message from CVS:
Fixed some seeking issues
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Original commit message from CVS:
add mpegaudioparse to spec file
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Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
Remove get_time code that is both wrong and unneeded.
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the caps or a base class...
Original commit message from CVS:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain),
(gst_rtph263penc_set_property), (gst_rtph263penc_get_property):
* gst/rtp/gstrtph263penc.h:
Added configurable pt and ssrc, to be merged in the caps or
a base class...
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Original commit message from CVS:
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_init),
(gst_rtph263pdec_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Some cleanups in the h263p (de)payloaders.
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Original commit message from CVS:
add mpegaudioparse to configure.ac
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Original commit message from CVS:
latest makefile and spec file fixes
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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/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_get_type),
(gst_rtpamrdec_base_init), (gst_rtpamrdec_class_init),
(gst_rtpamrdec_init), (gst_rtpamrdec_chain),
(gst_rtpamrdec_set_property), (gst_rtpamrdec_get_property),
(gst_rtpamrdec_change_state), (gst_rtpamrdec_plugin_init):
* gst/rtp/gstrtpamrdec.h:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_get_type),
(gst_rtpamrenc_base_init), (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property),
(gst_rtpamrenc_change_state), (gst_rtpamrenc_plugin_init):
* gst/rtp/gstrtpamrenc.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_class_init),
(gst_rtpmpaenc_flush), (gst_rtpmpaenc_chain):
Added very simplistic amr payloader. depayloader does not
work yet.
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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/rtp/Makefile.am:
* gst/rtp/gstrtp-common.h:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16enc.h:
* gst/rtp/gstrtpdec.c: (gst_rtpdec_get_type),
(gst_rtpdec_class_init), (gst_rtpdec_chain_rtp),
(gst_rtpdec_chain_rtcp), (gst_rtpdec_change_state),
(gst_rtpdec_plugin_init):
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_get_type),
(gst_rtph263pdec_base_init), (gst_rtph263pdec_class_init),
(gst_rtph263pdec_init), (gst_rtph263pdec_chain),
(gst_rtph263pdec_set_property), (gst_rtph263pdec_get_property),
(gst_rtph263pdec_change_state), (gst_rtph263pdec_plugin_init):
* gst/rtp/gstrtph263pdec.h:
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_get_type),
(gst_rtph263penc_base_init), (gst_rtph263penc_class_init),
(gst_rtph263penc_init), (gst_rtph263penc_flush),
(gst_rtph263penc_chain), (gst_rtph263penc_set_property),
(gst_rtph263penc_get_property), (gst_rtph263penc_change_state),
(gst_rtph263penc_plugin_init):
* gst/rtp/gstrtph263penc.h:
* gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_get_type),
(gst_rtpmpadec_base_init), (gst_rtpmpadec_class_init),
(gst_rtpmpadec_init), (gst_rtpmpadec_chain),
(gst_rtpmpadec_set_property), (gst_rtpmpadec_get_property),
(gst_rtpmpadec_change_state), (gst_rtpmpadec_plugin_init):
* gst/rtp/gstrtpmpadec.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_get_type),
(gst_rtpmpaenc_base_init), (gst_rtpmpaenc_class_init),
(gst_rtpmpaenc_init), (gst_rtpmpaenc_flush), (gst_rtpmpaenc_chain),
(gst_rtpmpaenc_set_property), (gst_rtpmpaenc_get_property),
(gst_rtpmpaenc_change_state), (gst_rtpmpaenc_plugin_init):
* gst/rtp/gstrtpmpaenc.h:
* gst/rtp/rtp-packet.c:
* gst/rtp/rtp-packet.h:
Remove old code that is now in gst-libs/gst/rtp/.
Added some payload/depayloaders.
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
Fix port number range.
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Original commit message from CVS:
* configure.ac:
Added mpegaudioparse
* ext/lame/gstlame.c: (gst_lame_src_getcaps),
(gst_lame_src_setcaps), (gst_lame_sink_setcaps),
(gst_lame_sink_event), (gst_lame_chain):
Some cleanups.
Fix memleak.
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_class_init), (gst_mp3parse_init),
(gst_mp3parse_chain), (gst_mp3parse_change_state):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Ported mpegaudioparse
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Original commit message from CVS:
removing README from Makefile.am as its gone from CVS
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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/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
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Original commit message from CVS:
* configure.ac:
* ext/amrnb/amrnbparse.c: (gst_amrnbparse_read_header):
Fix compile warning.
* ext/lame/gstlame.c: (gst_lame_class_init),
(gst_lame_src_getcaps), (gst_lame_src_setcaps),
(gst_lame_sink_setcaps), (gst_lame_init), (gst_lame_sink_event),
(gst_lame_chain), (gst_lame_change_state):
* ext/lame/gstlame.h:
Port lame plugin
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the source code -- was only in the commi...
Original commit message from CVS:
2005-08-16 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c (gst_dv1394src_iso_receive): Note
license info in the source code -- was only in the commit log
before.
* ext/dv/gstdvdec.h:
* ext/dv/gstdvdec.c: Only decodes systemstream=FALSE dv video --
old pipelines using dvdec should probably have a dvdemux first.
* ext/dv/gstdvdemux.h:
* ext/dv/gstdvdemux.c: Split out from dvdec, chunks the incoming
systemstream=TRUE data into frames, sets caps data, and spits out
PCM audio in addition to systemstream=FALSE video frames. Operates
in chain mode only for now; should make a getrange version as
well.
* ext/dv/gstdv.c: New file, registers the libgstdv plugin.
* ext/dv/Makefile.am: Library name changed to libgstdv. Split
dvdec into dvdemux and dvdec.
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Original commit message from CVS:
remove seeking example, they're in gst-plugins-base
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Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event), (gst_faad_chain):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Handle _push() return values.
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Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_event):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Fix debug.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps):
Forwardport from 0.8 to implement RLE.
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Original commit message from CVS:
* gst/rtsp/README:
Added rtsp server implementation docs.
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