Commit graph

1792 commits

Author SHA1 Message Date
Stefan Kost
048a15698d gst/spectrum/gstspectrum.c: Fix leaking buffers.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_event),
(gst_spectrum_transform_ip):
Fix leaking buffers.
* tests/check/Makefile.am:
* tests/check/elements/spectrum.c: (setup_spectrum),
(cleanup_spectrum), (GST_START_TEST), (spectrum_suite), (main):
Add simple test for spectrum element.
2007-06-20 08:56:17 +00:00
Stefan Kost
42ca9c44e6 gst/qtdemux/: Add MJPG to the variants of motion jpeg.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
(qtdemux_video_caps):
* gst/qtdemux/qtdemux_fourcc.h:
Add MJPG to the variants of motion jpeg.
2007-06-20 08:26:21 +00:00
Wim Taymans
ebce97adf5 gst/rtsp/rtspconnection.c: Use threadsafe inet_ntop to convert an ip number to a string.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close), (rtsp_connection_free):
Use threadsafe inet_ntop to convert an ip number to a string.
Fixes #447961.
Don't leak fd (and ip) when freeing a connection without first closing
it.
2007-06-19 14:48:03 +00:00
Christian Schaller
6fb347e76f add 'LEGAL' file describing why this is in -good and under what circumstances it might need to move.
Original commit message from CVS:
add 'LEGAL' file describing why this is in -good and under what
circumstances it might need to move.
2007-06-19 14:11:49 +00:00
Tim-Philipp Müller
c093c9aa1c gst/rtsp/rtspconnection.c: Revert previous commit again, since we are frozen (sorry).
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
Revert previous commit again, since we are frozen (sorry).
2007-06-17 12:35:03 +00:00
Peter Kjellerstedt
9d2c01b551 gst/rtsp/rtspconnection.c: inet_ntoa() uses a static buffer internally, so we need to copy the returned string if we ...
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_free):
inet_ntoa() uses a static buffer internally, so we need to copy the
returned string if we want to store it for later (#447961).
2007-06-17 12:24:58 +00:00
Vincent Torri
d5e801139c gst/rtsp/rtspconnection.c: Fix the MingW build.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect):
Fix the MingW build.
Patch By: Vincent Torri <vtorri at univ-evry dot fr>
Fixes: #446981
2007-06-15 08:32:52 +00:00
Edward Hervey
98165a69ee gst/qtdemux/qtdemux.c: For AMR-NB streams, export the AMRSpecificBox as codec_data on the caps.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
For AMR-NB streams, export the AMRSpecificBox as codec_data on the
caps.
Fixes #447458
2007-06-14 10:23:20 +00:00
Wim Taymans
2826212827 gst/rtp/gstrtph264depay.c: Make sure we allocate enough memory for the codec_data.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
Make sure we allocate enough memory for the codec_data.
Fixes #447210.
2007-06-13 17:11:24 +00:00
Tommi Myöhänen
a3719a4a15 gst/rtp/: Add missing rate fields to caps. Fixes #441118.
Original commit message from CVS:
Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
Add missing rate fields to caps. Fixes #441118.
2007-06-11 10:21:13 +00:00
Thomas Vander Stichele
0444519131 gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details): Fix element description.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
Fix element description.
2007-06-08 20:20:56 +00:00
Tim-Philipp Müller
37e3097981 gst/videobox/gstvideobox.c: Printf fixes in debug statements; use LOG level for debug statements that are printed for...
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
Printf fixes in debug statements; use LOG level for debug statements
that are printed for each and every frame; convert c++ comments to
C-style comments; not much point using g_try_malloc() if we then not
even check the return value.
2007-06-06 08:53:12 +00:00
Tim-Philipp Müller
63fc719540 configure.ac: Bump requirements to released versions (core and base 0.10.13).
Original commit message from CVS:
* configure.ac:
Bump requirements to released versions (core and base 0.10.13).
* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
own implementation.
2007-06-05 16:32:19 +00:00
Wim Taymans
46ae6f904f gst/multipart/multipartmux.c: Add support for mapping gst structure names to the MIME type equivalent.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
Add support for mapping gst structure names to the MIME type equivalent.
Implemented for audio/x-mulaw->audio/basic. Fixes #442874.
2007-06-05 09:11:41 +00:00
Sebastian Dröge
10da08ace4 gst/wavenc/gstwavenc.*: Properly write wav files with width!=depth by having the depth most significant bytes set and...
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
(gst_wavenc_chain), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Properly write wav files with width!=depth by having the depth most
significant bytes set and all others zero. Fixes #442535.
2007-06-03 11:21:44 +00:00
Wim Taymans
6ce8b13eb4 gst/rtsp/rtspconnection.c: Add include to make buildbot happy.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c:
Add include to make buildbot happy.
2007-06-01 13:52:17 +00:00
Peter Kjellerstedt
f12fb76f70 gst/rtsp/: Improves version checking, allowing an RTSP server to reply with "505
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (add_date_header),
(rtsp_connection_send), (parse_response_status),
(parse_request_line), (parse_line), (rtsp_connection_receive):
* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (key_value_foreach),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_remove_header), (rtsp_message_append_headers),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Improves version checking, allowing an RTSP server to reply with "505
RTSP Version not supported.
Adds a Date header to all messages.
Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
want to be able to send a response even if something in the request was
invalid. EINVAL is only used when passing wrong arguments to functions.
Do not handle an invalid method in parse_request_line(). Defer this to
the caller so it can respond with "405 Method Not Allowed".
Improves parsing of the timeout parameter to the Session header,
allowing whitespace after the semicolon.
Avoids a compiler warning due to variables shadowing a function argument.
2007-06-01 13:07:11 +00:00
Daniel Charles
89ae9b40f9 gst/rtp/: Add support for AMR-WB.
Original commit message from CVS:
Based on Patch by: Daniel Charles <dcharles at ti dot com>
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpamrpay.h:
Add support for AMR-WB.
Small cleanups such as using BOILERPLATE.
2007-06-01 11:16:17 +00:00
Wim Taymans
0b2e6f1c90 gst/rtsp/rtspextwms.c: Fix compile warning when debug is disabled as spotted bu Saur on IRC.
Original commit message from CVS:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
Fix compile warning when debug is disabled as spotted bu Saur on IRC.
2007-05-31 15:57:07 +00:00
Tim-Philipp Müller
3127a32c1c gst/avi/gstavidemux.*: Parse subtitle text streams instead of erroring out (#442034). Still needs a parser for the su...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
* gst/avi/gstavidemux.h:
Parse subtitle text streams instead of erroring out (#442034). Still
needs a parser for the subtitles to actually show up.
2007-05-30 14:38:59 +00:00
Tim-Philipp Müller
f91649edb8 gst/avi/gstavidemux.c: Make _push_event() return TRUE if the event could be pushed on at least one pad and not only i...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
(gst_avi_demux_loop):
Make _push_event() return TRUE if the event could be pushed on at
least one pad and not only if it could be pushed on all pads,
otherwise we'll end up posting an error message on EOS if one or
more source pads are not connected.
2007-05-30 12:46:32 +00:00
Wim Taymans
0ea8d875a2 gst/rtsp/rtsptransport.c: Use renamed RTP bin.
Original commit message from CVS:
* gst/rtsp/rtsptransport.c:
Use renamed RTP bin.
2007-05-28 16:39:09 +00:00
Dejan Sakelšak
82a509fdfd gst/videobox/gstvideobox.c: Add AYUV->AYUV and AYUV->I420 formats.
Original commit message from CVS:
Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_set_property), (gst_video_box_transform_caps),
(video_box_recalc_transform), (gst_video_box_set_caps),
(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
(gst_video_box_i420_i420), (gst_video_box_transform),
(plugin_init):
Add AYUV->AYUV and AYUV->I420 formats.
Fix negotiation and I420->AYUV conversion.
Fixes #429329.
2007-05-28 15:01:33 +00:00
Tim-Philipp Müller
da0da24565 gst/id3demux/gstid3demux.c: Don't leak newsegment events.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
Don't leak newsegment events.
2007-05-25 20:51:36 +00:00
Tim-Philipp Müller
fefb7bfa6d gst/wavparse/Makefile.am: Add '-lm' to LIBS for ceil(), don't assume one of our dependencies drags it in.
Original commit message from CVS:
* gst/wavparse/Makefile.am:
Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
drags it in.
2007-05-25 20:33:10 +00:00
Jan Schmidt
4a7ecfb814 gst/: Handle and adjust new-segment events so that downstream really sees a stream with the tag pieces stripped off t...
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
(gst_id3demux_send_new_segment), (gst_id3demux_chain),
(gst_id3demux_sink_event):
* gst/id3demux/gstid3demux.h:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
(gst_tag_demux_chain), (gst_tag_demux_sink_event),
(gst_tag_demux_send_new_segment):
Handle and adjust new-segment events so that downstream really
sees a stream with the tag pieces stripped off the front and back.
Fixes strangeness in seeking when mp3 decoders use the new-segment
byte position to estimate their current playback position timestamp
and then the arriving buffers don't match up.
2007-05-25 10:44:12 +00:00
Jan Schmidt
465a740bbf gst/autodetect/gstautoaudiosink.c: Don't unnecessarily perform a READY->NULL->READY transition on the detected audio ...
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
Don't unnecessarily perform a READY->NULL->READY transition on the
detected audio sink when starting up. Fixes: #440127
2007-05-25 10:23:49 +00:00
Wim Taymans
587d209252 gst/rtsp/gstrtspsrc.c: Init value to avoid infinte loops.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
Init value to avoid infinte loops.
2007-05-24 08:14:00 +00:00
Peter Kjellerstedt
77cc870bbc gst/rtsp/: Fix for new API.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_play):
(rtsp_connection_send), (rtsp_connection_receive):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
Fix for new API.
* gst/rtsp/rtspconnection.c: (add_auth_header),
Only add authorisation and session headers when sending messages.
* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_unset), (rtsp_message_add_header),
(rtsp_message_remove_header), (rtsp_message_get_header),
(rtsp_message_append_headers), (dump_key_value),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Add support for multiple headers of the same type by storing the parsed
headers in a GArray instaed of a hashtable.
2007-05-24 08:10:42 +00:00
Stefan Kost
ab92670d13 configure.ac: Depend on gstreamer-0.10.12.1. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _Gs...
Original commit message from CVS:
* configure.ac:
Depend on gstreamer-0.10.12.1.
* gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
_GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
parent_class, gst_iir_equalizer_band_set_property,
gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
gst_iir_equalizer_child_proxy_get_child_by_index,
gst_iir_equalizer_child_proxy_get_children_count,
gst_iir_equalizer_child_proxy_interface_init, setup_filter,
gst_iir_equalizer_compute_frequencies,
gst_iir_equalizer_set_property, gst_iir_equalizer_get_property,
plugin_init):
* gst/equalizer/gstiirequalizer.h (audiofilter):
* gst/equalizer/gstiirequalizernbands.c (ARG_NUM_BANDS,
gst_iir_equalizer_nbands_base_init, gst_iir_equalizer_nbands_init,
gst_iir_equalizer_nbands_set_property):
Use new locking macros.
* gst/filter/gstbpwsinc.c (bpwsinc_set_caps):
Add fixme.
* gst/spectrum/gstspectrum.c (SPECTRUM_WINDOW_BASE,
SPECTRUM_WINDOW_LEN, gst_spectrum_init, gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use new locking macros. Turn two fixed values into #defines.
2007-05-22 11:14:13 +00:00
Stefan Kost
161e49b62e ChangeLog: ChangeLog surgery. gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN, _GstIirEqualizerBa...
Original commit message from CVS:
* ChangeLog:
ChangeLog surgery.
* gst/equalizer/gstiirequalizer.c (ARG_BAND_WIDTH, _do_init, ARG_GAIN,
_GstIirEqualizerBand, object, _GstIirEqualizerBandClass,
parent_class, gst_iir_equalizer_band_set_property,
gst_iir_equalizer_band_class_init, gst_iir_equalizer_band_get_type,
gst_iir_equalizer_child_proxy_get_child_by_index,
gst_iir_equalizer_child_proxy_get_children_count,
gst_iir_equalizer_child_proxy_interface_init, setup_filter,
gst_iir_equalizer_compute_frequencies, plugin_init):
* tests/icles/equalizer-test.c:
Add fixme and comment for example.
2007-05-21 14:01:16 +00:00
Stefan Kost
5e9e882543 gst/spectrum/gstspectrum.c (gst_spectrum_set_property, gst_spectrum_event, gst_spectrum_transform_ip):
Original commit message from CVS:
* gst/spectrum/gstspectrum.c (gst_spectrum_set_property,
gst_spectrum_event, gst_spectrum_transform_ip):
Use lock to protect from concurrent access.
2007-05-21 12:43:37 +00:00
Wim Taymans
127d233104 gst/udp/gstudpsrc.c: Since we depend on 0.10.13 -core, override the unlock_stop vmethod for safer shutdown.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
safer shutdown.
2007-05-21 10:07:05 +00:00
Wim Taymans
321a79d484 gst/rtsp/gstrtpdec.*: Added signal for backwards compat.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Added signal for backwards compat.
2007-05-21 10:03:42 +00:00
René Stadler
4bd1140630 Use audioconvert for converting from non-native endianness floats in auparse instead of doing it ourself. Fixes #424527.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Use audioconvert for converting from non-native endianness floats
in auparse instead of doing it ourself. Fixes #424527.
This needs the audioconvert from plugins-base CVS.
2007-05-21 09:32:26 +00:00
Wim Taymans
20dc422e40 gst/rtp/gstrtph263ppay.c: Fix enum registration.
Original commit message from CVS:
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_flush):
Fix enum registration.
2007-05-21 09:29:30 +00:00
Antoine Tremblay
0ff05f8195 gst/rtp/gstrtph263ppay.*: Add new fragmentation mode base on GOB headers. Fixes #438940.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
(gst_rtp_h263p_pay_flush):
* gst/rtp/gstrtph263ppay.h:
Add new fragmentation mode base on GOB headers. Fixes #438940.
2007-05-21 08:57:18 +00:00
Tim-Philipp Müller
798b78630f gst/: Printf format fixes (#439910, #439911).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample):
* gst/switch/gstswitch.c: (gst_switch_chain):
Printf format fixes (#439910, #439911).
2007-05-20 14:14:49 +00:00
Tim-Philipp Müller
263e0458f1 gst/rtsp/gstrtspsrc.c: Printf format fix.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Printf format fix.
2007-05-20 14:05:42 +00:00
René Stadler
4e45e0a269 Add replaygain playback elements (#412710).
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-replaygain.xml:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
(gst_rg_analysis_start), (gst_rg_analysis_set_caps),
(gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
(gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
(gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
(gst_rg_analysis_album_result):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
(gst_rg_limiter_class_init), (gst_rg_limiter_init),
(gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
(gst_rg_limiter_transform_ip):
* gst/replaygain/gstrglimiter.h:
* gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
(gst_rg_volume_class_init), (gst_rg_volume_init),
(gst_rg_volume_set_property), (gst_rg_volume_get_property),
(gst_rg_volume_dispose), (gst_rg_volume_change_state),
(gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
(gst_rg_volume_reset), (gst_rg_volume_update_gain),
(gst_rg_volume_determine_gain):
* gst/replaygain/gstrgvolume.h:
* gst/replaygain/replaygain.c: (plugin_init):
* gst/replaygain/replaygain.h:
* gst/replaygain/rganalysis.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (send_eos_event),
(GST_START_TEST):
* tests/check/elements/rglimiter.c: (setup_rglimiter),
(cleanup_rglimiter), (set_playing_state), (create_test_buffer),
(verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
* tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
(cleanup_rgvolume), (set_playing_state), (set_null_state),
(send_eos_event), (send_tag_event), (test_buffer_new),
(fail_unless_target_gain), (fail_unless_result_gain),
(fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
Add replaygain playback elements (#412710).
2007-05-19 10:01:45 +00:00
Wim Taymans
fc99abef7f gst/rtsp/gstrtspsrc.c: Don't crash when an unsupported transport error was returned by the server, just try to config...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
Don't crash when an unsupported transport error was returned by the
server, just try to configure the next stream. Fixes #439255.
2007-05-18 13:27:39 +00:00
Wim Taymans
e04f7a828f gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Add TCP timeout property and use it for all TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_write), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
Make connect and writes cancelable and make them use the timeout.
2007-05-18 11:39:12 +00:00
Wim Taymans
e4720e286c gst/rtsp/gstrtspsrc.c: Refactor timeout handling.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Refactor timeout handling.
Also send keep-alive when dealing with TCP transport.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_free), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
* gst/rtsp/rtspconnection.h:
Use a timer to handle the session timeouts, add some methods to deal
with timeouts.
2007-05-18 10:36:12 +00:00
Wim Taymans
ccd7a136a9 gst/rtsp/gstrtspsrc.c: Ignore streams that fail the setup command, we will retry with a different transport later on.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams):
Ignore streams that fail the setup command, we will retry with a
different transport later on.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_configure_stream):
Fix encoding name case.
2007-05-17 14:56:39 +00:00
Stefan Kost
0434640bc1 gst/debug/breakmydata.c (gst_break_my_data_init): One more try. This should be the proper fix now.
Original commit message from CVS:
* gst/debug/breakmydata.c (gst_break_my_data_init):
One more try. This should be the proper fix now.
2007-05-15 11:18:33 +00:00
Stefan Kost
e4abba63b0 gst/debug/breakmydata.c: Ooops, no // comments please.
Original commit message from CVS:
* gst/debug/breakmydata.c:
Ooops, no // comments please.
2007-05-15 06:41:58 +00:00
Stefan Kost
c7ecf8c9a8 gst/debug/breakmydata.c: Fix gst_buffer_is_writable() assertion.
Original commit message from CVS:
* gst/debug/breakmydata.c: (gst_break_my_data_class_init),
(gst_break_my_data_init):
Fix gst_buffer_is_writable() assertion.
2007-05-15 06:34:48 +00:00
Wim Taymans
4da361f94c gst/rtp/: Update theora pay/depayloader in a similar to vorbis.
Original commit message from CVS:
* gst/rtp/gstrtptheoradepay.c: (decode_base64),
(gst_rtp_theora_depay_parse_configuration):
* gst/rtp/gstrtptheorapay.c: (encode_base64),
(gst_rtp_theora_pay_finish_headers),
(gst_rtp_theora_pay_handle_buffer):
Update theora pay/depayloader in a similar to vorbis.
* gst/rtp/gstrtpvorbisdepay.c:
(gst_rtp_vorbis_depay_parse_configuration):
Update docs.
2007-05-14 17:10:12 +00:00
Wim Taymans
789ef04027 gst/rtsp/gstrtspsrc.c: When we try to execute a method that is not supported by the server, don't error out but remov...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
When we try to execute a method that is not supported by the server,
don't error out but remove the method from the accepted methods so that
we never try to perform this method again.
2007-05-14 16:19:58 +00:00
Wim Taymans
4333477d0c gst/rtp/gstrtpvorbisdepay.c: Remove annoying _dump_mem.
Original commit message from CVS:
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
Remove annoying _dump_mem.
2007-05-14 14:47:26 +00:00
Wim Taymans
63b73eff7d gst/rtsp/gstrtspsrc.c: Parse range correctly.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
Parse range correctly.
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
The baseurl now always has a '/' at the start.
2007-05-14 11:11:42 +00:00
Wim Taymans
fc2f6baf0d gst/rtsp/gstrtspsrc.c: Factor out caps configuration and configure more stuff such as the time ranges and speed/scale...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
Factor out caps configuration and configure more stuff such as the time
ranges and speed/scale values.
* gst/rtsp/rtsptransport.c:
Add Copyright after non-trival fixes.
2007-05-14 09:01:05 +00:00
David Schleef
bcbbda0b80 gst/replaygain/rganalysis.c: Fix wrong ifdef for visual C++. Fixes: #437403.
Original commit message from CVS:
* gst/replaygain/rganalysis.c:
Fix wrong ifdef for visual C++.  Fixes: #437403.
By Ali Sabil <ali.sabil@gmail.com>.
2007-05-13 19:57:45 +00:00
Sébastien Moutte
603656d1bf gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 can build in_data += (filter->width / 8).
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_transform_ip):
Use guint8 * instead of gpointer then vs6 can build
in_data += (filter->width / 8).
2007-05-13 15:47:13 +00:00
Peter Kjellerstedt
7ef62aac45 gst/rtsp/: Make channel guint8 where possible.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
(rtsp_message_get_header):
* gst/rtsp/rtspmessage.h:
Make channel guint8 where possible.
Make rtsp_message_init_data() take the channel as a guint8.
* gst/rtsp/rtspdefs.c:
Fixed a typo: Timout -> Timeout
* gst/rtsp/rtspdefs.h:
Make RTSP_CHECK() behave as a statement.
* gst/rtsp/sdpmessage.c:
Avoid a compiler warning in INIT_ARRAY().
Fixes #437692.
2007-05-12 16:37:50 +00:00
Peter Kjellerstedt
02a64fe5ad gst/rtsp/rtspurl.*: Add support for query parameters to RTSP URLs.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
(rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Add support for query parameters to RTSP URLs.
2007-05-12 16:27:51 +00:00
Peter Kjellerstedt
5f9984e866 gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
Tim-Philipp Müller
4128e375f1 gst/wavparse/gstwavparse.c: Skip LIST chunks before the fmt chunk (fixes #437499). Also fix streaming mode regression...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
streaming mode regression for file from #343837 with 'bext' chunk
before the 'fmt' chunk.
2007-05-11 16:01:45 +00:00
Wim Taymans
02fa0a7992 gst/rtsp/: Preliminary seek support.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_handle_src_event),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.h:
Preliminary seek support.
Activate internal pads so that we can receive events on them.
Don't try to parse a range string when it's NULL.
2007-05-11 15:09:39 +00:00
Wim Taymans
5bc71b661d gst/rtp/README: Update README with new RTP variables that will be used for synchronisation.
Original commit message from CVS:
* gst/rtp/README:
Update README with new RTP variables that will be used for
synchronisation.
* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
(gst_rtp_vorbis_depay_parse_configuration),
(gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c: (encode_base64),
(gst_rtp_vorbis_pay_finish_headers),
(gst_rtp_vorbis_pay_handle_buffer):
Update vorbis pay and depayloader to draft-04.
2007-05-11 15:04:38 +00:00
Wim Taymans
3e1fd61201 gst/rtsp/rtsptransport.c: UDP MCAST is actually the default for RTP/AVP.
Original commit message from CVS:
* gst/rtsp/rtsptransport.c:
UDP MCAST is actually the default for RTP/AVP.
2007-05-11 11:24:13 +00:00
Wim Taymans
4b69fc4466 gst/rtsp/rtsptransport.c: Make UDP the default transport when not specified.
Original commit message from CVS:
* gst/rtsp/rtsptransport.c:
Make UDP the default transport when not specified.
2007-05-11 09:12:55 +00:00
Stefan Kost
eb5b5a8400 gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
2007-05-10 14:02:07 +00:00
David Schleef
7ab6d2b0b0 gst/level/gstlevel.c: Revert last change.
Original commit message from CVS:
* gst/level/gstlevel.c:
Revert last change.
2007-05-10 01:21:19 +00:00
Sébastien Moutte
f636fb8b34 gst/level/gstlevel.c: Use guint8 * instead of gpointer then vs6 know the size of data pointed when moving the pointer.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
(gst_level_transform_ip):
Use guint8 * instead of gpointer then vs6 know the size of data
pointed when moving the pointer.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
Move instructions after variables declaration.
* win32/vs6/autogen.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
Update vs6 project files.
2007-05-09 21:30:53 +00:00
Wim Taymans
d29215b257 gst/rtsp/: Add code to parse time ranges.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
(rtsp_range_free):
* gst/rtsp/rtsprange.h:
Add code to parse time ranges.
Report DURATION on the stream when possible.
2007-05-09 11:23:39 +00:00
Tim-Philipp Müller
e38b5e7590 gst/videomixer/videomixer.c: Fix strides calculation for AYUV (it's just width*4) (#436910).
Original commit message from CVS:
* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
(gst_videomixer_collected):
Fix strides calculation for AYUV (it's just width*4) (#436910).
2007-05-08 15:49:01 +00:00
Sebastian Dröge
3d7b6f15b8 gst/audiofx/: Sync the GObject properties before each processing step to properly work with the controller.
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
Sync the GObject properties before each processing step to properly
work with the controller.
2007-05-06 21:32:40 +00:00
Wim Taymans
9e37243eca gst/rtsp/gstrtspsrc.c: Let more error state trickle down so that we can catch more error cases.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
Let more error state trickle down so that we can catch more error
cases.
Handle keep-alive a little smarter by selecting a method the server
actually supports.
Fix a race in UDP streaming shutdown.
2007-05-04 15:17:14 +00:00
Wim Taymans
5f2fbbd76b gst/rtsp/gstrtspsrc.c: Ignore errors when trying to use the keep-alive messages.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
Ignore errors when trying to use the keep-alive messages.
2007-05-04 13:04:31 +00:00
Wim Taymans
fb80e57990 gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport):
Send RTCP messages back to the server over the TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Factor out and expose lowlevel _write and _read methods.
Implement sending data messages to the server.
2007-05-04 12:31:32 +00:00
Wim Taymans
4d42c097a6 gst/multipart/multipartmux.c: Fix timestamps on outgoing buffers.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
(gst_multipart_mux_collected):
Fix timestamps on outgoing buffers.
2007-05-03 15:55:06 +00:00
Wim Taymans
5ba2fa6e3f gst/multipart/multipartmux.c: Emit NEWSEGMENT events before pushing the first buffer.
Original commit message from CVS:
* gst/multipart/multipartmux.c:
(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
(gst_multipart_mux_change_state):
Emit NEWSEGMENT events before pushing the first buffer.
2007-05-03 14:39:09 +00:00
Wim Taymans
17011e9a41 gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
Refactor transport configuration code.
Create internal pads for TCP transport so that we can implement events
and queries.
Handle events and queries.
Parse range from the SDP.
Fix race in pause handler where the connection could still be flushing.
2007-05-03 13:48:54 +00:00
Wim Taymans
24e51b3c73 gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
2007-05-02 19:32:58 +00:00
Wim Taymans
6991907036 gst/wavparse/gstwavparse.c: Only set DISCONT when there actually is a discont or when we just started.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
Only set DISCONT when there actually is a discont or when we just
started.
2007-05-02 18:25:09 +00:00
Wim Taymans
64e0ee90f6 gst/wavparse/gstwavparse.*: Be a bit more clever when dealing with VBR files with FACT tags, we don't want to timesta...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
Be a bit more clever when dealing with VBR files with FACT tags, we
don't want to timestamp buffers in that case but the estimated BPS can
be used for seeking.
Only send close segment in the streaming thread.
2007-05-02 17:19:36 +00:00
Wim Taymans
8281f6c054 gst/rtsp/test.c: Fix compilation of deprecated test just because I'm too lazy to delete it.
Original commit message from CVS:
* gst/rtsp/test.c: (main):
Fix compilation of deprecated test just because I'm too lazy to delete
it.
2007-05-02 14:27:28 +00:00
Wim Taymans
92396be152 gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
Sjoerd Simons
f34fce9df4 gst/rtp/gstrtpmp4vpay.*: Handle NEWSEGMENT and FLUSH events. Fixes #434824.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
* gst/rtp/gstrtpmp4vpay.h:
Handle NEWSEGMENT and FLUSH events. Fixes #434824.
2007-05-01 16:13:58 +00:00
Wim Taymans
066598d8de gst/udp/gstmultiudpsink.c: Add code to drop membership of a multicast group.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (leave_multicast),
(gst_multiudpsink_add), (gst_multiudpsink_remove):
Add code to drop membership of a multicast group.
* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
(gst_udpsink_set_uri):
Implement URI handler.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
Use URI handler to make udpsink instace.
Improve code to configure port and destination.
2007-04-29 14:43:37 +00:00
Wim Taymans
589b8282e8 gst/udp/gstmultiudpsink.c: Fix multicast detection.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
Fix multicast detection.
Don't try to join a multicast group if the address is not multicast.
* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
Small debug improvement.
2007-04-29 12:19:21 +00:00
Wim Taymans
6a790cb75a gst/rtsp/gstrtspsrc.c: Ignore ASYNC state messages from the udpsink, it's irrelevant for the parent.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_handle_message):
Ignore ASYNC state messages from the udpsink, it's irrelevant for the
parent.
2007-04-27 16:44:17 +00:00
Wim Taymans
7fe2138eea gst/rtp/gstrtpilbcdepay.h: Fix mode property when specified as an arg.
Original commit message from CVS:
* gst/rtp/gstrtpilbcdepay.h:
Fix mode property when specified as an arg.
2007-04-27 15:30:39 +00:00
Wim Taymans
530f214bd5 gst/rtsp/gstrtspsrc.*: Protect state changes with a lock.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect state changes with a lock.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(parse_line):
* gst/rtsp/rtspconnection.h:
Remove some unused stuff.
2007-04-26 10:08:27 +00:00
Wim Taymans
45b77c57b4 gst/udp/gstudpsrc.c: Handle the case where there are exactly 0 bytes to read and the ioctl did not report an error. F...
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Handle the case where there are exactly 0 bytes to read and the ioctl
did not report an error. Fixes #433530.
2007-04-26 08:48:30 +00:00
Wim Taymans
88bf47c911 gst/wavparse/gstwavparse.*: Apply DISCONT to buffers.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
Apply DISCONT to buffers.
Only apply timestamp to the first sample after a DISCONT, too many VBR
files cause random jitter in the timestamps. Fixes #433119.
2007-04-26 08:39:49 +00:00
Wim Taymans
6937be1a09 gst/rtsp/gstrtpdec.*: Add dummy latency property to be backwards compat with rtpbin.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property):
* gst/rtsp/gstrtpdec.h:
Add dummy latency property to be backwards compat with rtpbin.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Add latency property and configure in the session manager.
Don't set invalid clock-base and seqnum-base on caps, some servers
sometimes don't send them.
2007-04-25 15:55:32 +00:00
Tim-Philipp Müller
e53a24511b gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ...
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
Double-check that RGB input caps are really RGBA caps (apparently
the core doesn't always catch it if those caps aren't a subset of
our template caps, also see #421543). Fixes #429319 in a way.
Also, don't leak the pad template in the transform_caps function.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/alphacolor.c: (setup_alphacolor),
(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
(GST_START_TEST), (alphacolor_suite):
Add some basic unit tests for alphacolor.
2007-04-25 15:31:53 +00:00
Wim Taymans
a7531984c3 gst/rtsp/rtspconnection.c: Read the channel byte as an unsigned byte.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
Read the channel byte as an unsigned byte.
2007-04-25 10:07:12 +00:00
Wim Taymans
24c5812d65 gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
(gst_rtp_gsm_depay_setcaps):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
(gst_ilbc_depay_get_property):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
(gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
(gst_rtp_pcmu_depay_setcaps):
Make sure we configure the clock_rate in the baseclass in the setcaps
function. Fixes #431282.
2007-04-25 09:47:48 +00:00
Wim Taymans
1beeda3ff2 gst/rtsp/gstrtspsrc.*: Parse server address from SDP.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_stream_free), (request_pt_map),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Parse server address from SDP.
Hook up a udpsink to send RTCP back to the server.
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/rtsp/rtsptransport.h:
Add some docs.
2007-04-25 08:36:46 +00:00
Stefan Kost
fa7454bda2 gst/wavparse/gstwavparse.c: Make header field check conditional. Fixes #433135
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Make header field check conditional. Fixes #433135
2007-04-25 06:52:09 +00:00
Tim-Philipp Müller
7002f0336b Add minimal docs blurb to alphacolor; split out headers into separate header file for gtk-doc.
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:
* docs/plugins/inspect/plugin-alphacolor.xml:
* gst/alpha/Makefile.am:
* gst/alpha/gstalphacolor.c:
* gst/alpha/gstalphacolor.h:
Add minimal docs blurb to alphacolor; split out headers into
separate header file for gtk-doc.
2007-04-24 09:12:42 +00:00
Tim-Philipp Müller
106db1b2eb gst/debug/progressreport.c: Don't try to post NULL message (in case we can't query upstream position or duration).
Original commit message from CVS:
* gst/debug/progressreport.c: (gst_progress_report_report):
Don't try to post NULL message (in case we can't query upstream
position or duration).
2007-04-20 17:25:50 +00:00
Michael Smith
4a1ceda8df gst/cutter/gstcutter.*: Fix some of the most obvious bugs in cutter. Now doesn't leak everything if input is silent.
Original commit message from CVS:
* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
(gst_cutter_get_caps):
* gst/cutter/gstcutter.h:
Fix some of the most obvious bugs in cutter. Now doesn't leak
everything if input is silent.
2007-04-18 12:36:37 +00:00
Sebastian Dröge
1723d916dd gst/wavenc/gstwavenc.*: everything else results in a invalid block align and invalid files.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Wav apparently only supports width==GST_ROUND_UP(depth), everything
else results in a invalid block align and invalid files.
2007-04-18 09:48:25 +00:00
Snaik
b5cfe36ab7 gst/smpte/barboxwipes.c: Add missing break statement for BOX_HORIZONTAL case.
Original commit message from CVS:
Patch by: Snaik <snaik32 gmail com>
* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
Add missing break statement for BOX_HORIZONTAL case.
2007-04-17 16:39:02 +00:00
Vincent Torri
188cc7a9e0 gst/wavparse/gstwavparse.c: Use correct format strings for integer types.
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Use correct format strings for integer types.
2007-04-17 10:14:43 +00:00
Sebastian Dröge
c383f21c10 gst/wavparse/gstwavparse.c: Use gst_riff_create_audio_template_caps () instead of the local caps.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_create_sourcepad):
Use gst_riff_create_audio_template_caps () instead of the local caps.
This makes updates of the local caps unecessary whenever libgstriff
gets support for new formats.
2007-04-17 02:51:02 +00:00