Commit graph

1987 commits

Author SHA1 Message Date
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
Wim Taymans
b752470823 docs/plugins/gst-plugins-good-plugins-sections.txt: Fix docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Fix docs.
* gst/rtsp/URLS:
Add some more example urls.
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_chain_rtp):
Better debugging.
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_parse_rtpinfo):
Remove unused code.
2007-04-13 09:32:21 +00:00
Stefan Kost
3bf1b5ecf7 gst/wavparse/gstwavparse.c: Relax the audio/mpeg caps again and add FIXME: comment.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Relax the audio/mpeg caps again and add FIXME: comment.
2007-04-13 08:19:35 +00:00
Stefan Kost
0722106b57 gst/wavparse/gstwavparse.*: More sanity check for the header fields. Fix type for 'rate' header field.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
* gst/wavparse/gstwavparse.h:
More sanity check for the header fields. Fix type for 'rate' header
field.
2007-04-13 06:20:28 +00:00
Tim-Philipp Müller
ef7c18817f gst/icydemux/gsticydemux.c: If the metadata strings we get in the stream are not UTF-8, try to interpret them accordi...
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
(gst_icydemux_unicodify):
If the metadata strings we get in the stream are not UTF-8, try to
interpret them according to the character encodings specified in the
GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
only fall back to locale/ISO-8859-1 if those aren't set or don't
work. Should fix #428901.
2007-04-12 16:06:31 +00:00
Wim Taymans
f5e4a8b028 gst/rtp/gstrtph264depay.c: Use the proper sync word for SPS and PPS.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c:
Use the proper sync word for SPS and PPS.
2007-04-12 14:20:56 +00:00
Thomas Vander Stichele
2fc868841f gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
Add a simple hashing implementation that we can use to generate
a 24-bit ident value based on the codebooks for vorbis and theora.
* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
gst_rtp_theora_pay_handle_buffer):
* gst/rtp/gstrtpvorbisdepay.c
(gst_rtp_vorbis_depay_parse_configuration,
gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
Use the hashing function, ensuring that the same codebooks result
in the same ident and thus the same SDP description.
Various log fixes/changes.
2007-04-12 11:41:11 +00:00
Wim Taymans
eae68a64fa gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
Make timescale 32 bits again so we don't screw up the pts_offset
calculations.
2007-04-12 10:52:02 +00:00
Wim Taymans
86a4c1c6b0 gst/rtsp/gstrtpdec.*: Make backward compat with rtpbin by adding the request-pt-map signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
* gst/rtsp/gstrtpdec.h:
Make backward compat with rtpbin by adding the request-pt-map signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams):
* gst/rtsp/gstrtspsrc.h:
Implement request-pt-map signals instead of setting caps on the buffers
for the session manager.
2007-04-12 08:21:28 +00:00
Wim Taymans
bd11d3c9d2 gst/udp/gstudp.c: Register GstNetBuffer in plugin_init so that the type can be used from multiple threads without races.
Original commit message from CVS:
* gst/udp/gstudp.c: (plugin_init):
Register GstNetBuffer in plugin_init so that the type can be used from
multiple threads without races.
2007-04-11 10:25:25 +00:00
Wim Taymans
2f97f23897 gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(qtdemux_parse_samples), (qtdemux_parse_segments),
(qtdemux_parse_trak), (qtdemux_parse_tree):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd):
Handle version 1 mdhd atoms to get extended precision durations.
Fixes #426972.
2007-04-11 09:53:38 +00:00
Wim Taymans
acddbd83ff gst/rtp/gstrtpamrdepay.c: Fix depayloader clock_rate and some cleanups.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
Fix depayloader clock_rate and some cleanups.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Don't push codec_data in the adapter because it might get flushed when
we get a discont.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Handle multiple AU per packet.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
(gst_rtp_sv3v_depay_plugin_init):
Disable rank, this one does not work.
Remove timestamping, base class does that.
2007-04-10 17:06:05 +00:00
Stefan Kost
497d589d56 gst/auparse/gstauparse.c: limit caps to the formats we announce in the template
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
limit caps to the formats we announce in the template
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
fix some crashers/asserts when dealing with broken files
2007-04-10 12:01:33 +00:00
Peter Kjellerstedt
50f88db3ad gst/: Fix some compiler warnings. Fixes #428182.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
(gst_rtp_speex_depay_setcaps):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
Fix some compiler warnings. Fixes #428182.
2007-04-10 10:01:14 +00:00
Wim Taymans
f80444aaec gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
Wim Taymans
86b40a1c70 gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Try to recover from packet loss a little better.
2007-04-05 15:05:24 +00:00
Wim Taymans
f70206175f gst/rtp/gstrtpmp4adepay.c: This element is ready to be autoplugged.
Original commit message from CVS:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
This element is ready to be autoplugged.
2007-04-05 13:56:44 +00:00
Julien Moutte
d42fcc86cf gst/avi/gstavidemux.c: Don't leave the offsets defined by upstream element on the compressed data buffer we are pushi...
Original commit message from CVS:
2007-04-05  Julien MOUTTE  <julien@moutte.net>

* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Don't leave the offsets defined by upstream element on the
compressed data buffer we are pushing downstream. Make them
GST_BUFFER_OFFSET_NONE.
2007-04-05 11:26:25 +00:00
Stefan Kost
30df72ccb7 gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
Original commit message from CVS:
* gst/avi/README:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
2007-04-04 12:39:41 +00:00
Wim Taymans
9598d82c0c gst/smpte/barboxwipes.c:
Original commit message from CVS:
* gst/smpte/barboxwipes.c:
Fix error as spotted by Snaik <snaik32 at gmail dot com>
2007-04-03 09:55:45 +00:00
Sebastian Dröge
c11fefd494 gst/wavparse/gstwavparse.c: Support audio/x-raw-float in wav files. This only works with plugins-base CVS, using an o...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Support audio/x-raw-float in wav files. This only works with
plugins-base CVS, using an older version doesn't have any
disadvantages though.
2007-03-30 17:19:34 +00:00
Sebastian Dröge
6632cdb003 Revert last change as we don't want plugins-good to depend on plugins-base CVS now.
Original commit message from CVS:
* configure.ac:
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Revert last change as we don't want plugins-good to depend on
plugins-base CVS now.
2007-03-30 15:59:27 +00:00
René Stadler
bfd65c42d1 configure.ac: Require gst-plugins-base CVS for audioconvert with non-native float support and width/depth fix in libg...
Original commit message from CVS:
* configure.ac:
Require gst-plugins-base CVS for audioconvert with non-native
float support and width/depth fix in libgstriff.
Patch by: René Stadler <mail at renestadler dot de>
* gst/auparse/gstauparse.c: (gst_au_parse_reset),
(gst_au_parse_parse_header), (gst_au_parse_chain):
* gst/auparse/gstauparse.h:
Don't swap the floats ourself if they're not in native endianness.
Instead let audioconvert handle this. Fixes #339838.
2007-03-29 18:51:33 +00:00
Wim Taymans
a87260cb3b gst/rtp/: Flush adapter on disconts.
Original commit message from CVS:
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
(gst_rtp_h263p_depay_change_state):
* gst/rtp/gstrtph263pdepay.h:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_change_state):
* gst/rtp/gstrtph264depay.h:
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
Flush adapter on disconts.
2007-03-29 14:40:35 +00:00
Wim Taymans
da3e23d375 gst/rtp/: Use more efficient adapter and rtpbuffer methods when possible.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
Use more efficient adapter and rtpbuffer methods when possible.
2007-03-29 14:03:21 +00:00
Sebastian Dröge
d26cbc8c66 gst/wavenc/gstwavenc.c: Correctly handle width!=depth input.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps):
Correctly handle width!=depth input.
* gst/wavparse/gstwavparse.c:
Already export in the caps that width==8 uses unsigned samples and
everything else uses signed samples.
2007-03-29 12:14:22 +00:00
Laurent Glayal
112216c22f gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
(gst_dynudpsink_init), (gst_dynudpsink_set_property),
(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
(gst_dynudpsink_close):
* gst/udp/gstdynudpsink.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Rework the socket allocation a bit based on the sockfd argument so that
it becomes usable.
Add a closefd property to instruct the udp elements to close the custom
file descriptors when going to READY. Fixes #423304.
API:GstUDPSrc::closefd property
API:GstDynUDPSink::closefd property
2007-03-29 09:59:23 +00:00
Laurent Glayal
d94a696bcd gst/rtp/: Added H264 payloader. Fixes #423782.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
(gst_rtp_h264_pay_plugin_init):
* gst/rtp/gstrtph264pay.h:
Added H264 payloader. Fixes #423782.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Small fixes.
2007-03-29 08:08:49 +00:00
Sebastian Dröge
c76eea67cc gst/wavparse/gstwavparse.c: Actually support depths from 1 to 32, not only 8 to 32.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Actually support depths from 1 to 32, not only 8 to 32.
2007-03-28 22:27:36 +00:00
Sebastian Dröge
7add372a7a gst/wavparse/gstwavparse.c: Add support for wav files containing audio/x-raw-int with random depths between 1 and 32 ...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
Add support for wav files containing audio/x-raw-int with random
depths between 1 and 32 bits.
2007-03-28 22:23:43 +00:00
Stefan Kost
c0cdcae569 gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.
Original commit message from CVS:
Based on patch by: Stefan Kost  <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.
2007-03-28 18:40:12 +00:00
Edward Hervey
ab589bff3e gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video).
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (qtdemux_parse_samples):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Process 'ctts' atoms, which are present in AVC ISO files (.mov files
with h264 video).
Use the offset present in 'ctts' to calculate the PTS for each packet
and set the PTS on outgoing buffers.
Fixes #423283
2007-03-28 15:17:27 +00:00
Wim Taymans
8f5fb88b5a gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Handle default clock-rates for static payload types, rearrange stuff so
that the rtpmap field in the sdp can override the defaults.
Parse RTP-Info field to get the seqnum and timebase fields that should
go in the caps.
Delay configuring caps after we got the RTP-Info from the PLAY reply from
the server.
2007-03-25 15:34:42 +00:00
Tim-Philipp Müller
c53ad3009d gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
Remove 'channel-positions' field when munging input caps into
1-channel output caps (I guess technically we should set the
position for each channel on the output caps if it's non-NONE,
but I'll save that as a task for another day).
2007-03-24 19:46:59 +00:00
Tim-Philipp Müller
56b1a888fd gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_remove_pads), (gst_deinterleave_process),
(gst_deinterleave_chain):
Don't leak input buffer in chain function; maintain our own list of
source pads - there are no guarantees about the order of the list
in the GstElement struct, and we want a very specific order; lastly,
some more debugging.
2007-03-22 22:14:29 +00:00
Tim-Philipp Müller
a227a885c9 gst/apetag/gsttagdemux.c: Rename registered type in preparation of GstTagDemux moving to
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
Rename registered type in preparation of GstTagDemux moving to
-base at some point in the future.
2007-03-21 11:49:32 +00:00
Tim-Philipp Müller
61b44790c4 gst/wavparse/gstwavparse.c: Streaming mode fixes: don't unref buffer we don't own any longer; remove bogus adapter fl...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
Streaming mode fixes: don't unref buffer we don't own any longer;
remove bogus adapter flush. Fixes #419338.
2007-03-19 10:29:19 +00:00
Stefan Kost
a0a99f1096 gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized to 1 bands and not to 3.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_init):
A 10 band EQ should be initialized to 1 bands and not to 3.
2007-03-16 09:57:40 +00:00
Philippe Kalaf
1be3219c70 gst/rtp/: Ported mulaw and alaw payloaders to use new base class
Original commit message from CVS:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
2007-03-14 22:21:26 +00:00
Stefan Kost
7ce779f579 gst/equalizer/: Add 3 and 10 band version and add missing gst_object_sync_values.
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(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_transform_ip), (plugin_init):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_base_init),
(gst_iir_equalizer_10bands_class_init),
(gst_iir_equalizer_10bands_init),
(gst_iir_equalizer_10bands_set_property),
(gst_iir_equalizer_10bands_get_property):
* gst/equalizer/gstiirequalizer10bands.h:
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_base_init),
(gst_iir_equalizer_3bands_class_init),
(gst_iir_equalizer_3bands_init),
(gst_iir_equalizer_3bands_set_property),
(gst_iir_equalizer_3bands_get_property):
* gst/equalizer/gstiirequalizer3bands.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_init):
Add 3 and 10 band version and add missing gst_object_sync_values.
* gst/spectrum/gstspectrum.c: (gst_spectrum_event),
(gst_spectrum_transform_ip):
Add some comments about float support.
2007-03-14 14:48:08 +00:00
Tim-Philipp Müller
dbe62aba11 gst/apetag/gsttagdemux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END her...
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END here as well.
2007-03-12 17:56:54 +00:00
Jan Schmidt
56fbcb6766 gst/id3demux/gstid3demux.c: Fix handling of -1 values for start and stop values when seeking, and SEEK_CUR+SEEK_END.
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
Fix handling of -1 values for start and stop values when seeking,
and SEEK_CUR+SEEK_END.
2007-03-12 17:24:23 +00:00
Tim-Philipp Müller
2354b65a9e gst/id3demux/id3v2frames.c: Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is the image format a vari...
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
the image format a variable-length NUL-terminated string; in
versions before that the image format is a fixed-length string of
3 characters (see #348644 for a sample tag).
Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
2007-03-12 13:28:29 +00:00
Tim-Philipp Müller
7236a2f8b3 Printf format fixes; also add some missing quotes in translated strings. Fixes #416728 and #416727.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index):
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
Printf format fixes; also add some missing quotes in translated
strings. Fixes #416728 and #416727.
2007-03-10 12:30:48 +00:00
Jan Schmidt
647934baf9 gst/autodetect/gstautoaudiosink.c: Tim and I can't think of any reason the child audio sink needs to be set back to N...
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
Tim and I can't think of any reason the child audio sink needs to
be set back to NULL after successfully determining that it can
reach READY - it gets immediately set back to READY by the caller
anyway, causing an unnecessary close/open of any audio devices
involved.
2007-03-09 20:12:08 +00:00
Wim Taymans
beef8e0136 gst/rtsp/gstrtspsrc.c: When activated, remove the udpsrc timeout, we have dataflow and timeouts will later be handled...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
When activated, remove the udpsrc timeout, we have dataflow and timeouts
will later be handled by the jitterbuffer.
2007-03-09 17:05:17 +00:00
Wim Taymans
a98caaeb67 gst/avi/gstavidemux.c: Fix stream position reporting after a seek. Fixes #416445.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
Fix stream position reporting after a seek. Fixes #416445.
2007-03-09 15:04:45 +00:00
Stefan Kost
44e09dddc4 gst/equalizer/: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy an...
Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_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),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_finalize), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_setup), (plugin_init):
* gst/equalizer/gstiirequalizer.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_class_init),
(gst_iir_equalizer_nbands_init),
(gst_iir_equalizer_nbands_set_property),
(gst_iir_equalizer_nbands_get_property):
* gst/equalizer/gstiirequalizernbands.h:
Refactor plugin into a base class and a first subclass (nband eq). The
nband eq uses GstChildProxy and is controlable. More subclasses will
follow.
2007-03-09 08:58:26 +00:00
René Stadler
654ad41f25 gst/avi/gstavidemux.c: Make avidemux accept optional header chunks in any order.
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_chain):
Make avidemux accept optional header chunks in any order.
Fixes #415446.
2007-03-08 16:01:42 +00:00
Sebastian Dröge
dbd1b8490f gst/audiofx/: Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and sof...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
2007-03-08 10:02:12 +00:00
Edward Hervey
816404ac41 gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, otherwise all debugging in files other than qtd...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
Share qtdemux debug category across all files, otherwise all debugging
in files other than qtdemux.c would end up in the default category.
2007-03-07 11:37:23 +00:00
Stefan Kost
143708a433 gst/level/gstlevel.*: Resolve message timestamps against the playback segment.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init),
(gst_level_set_caps), (gst_level_start), (gst_level_event),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Resolve message timestamps against the playback segment.
2007-03-07 11:24:05 +00:00
Stefan Kost
28114d571f gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps against the playback segment.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_event), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
One FIXME less, by resolving message timestamps against the playback
segment.
2007-03-07 11:23:20 +00:00
Tim-Philipp Müller
009c9750ea gst/id3demux/gstid3demux.c: Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the caps passed to ...
Original commit message from CVS:
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
(gst_id3demux_sink_activate):
Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
caps passed to it (previouslly one code path assumes it takes ownership
while another one assumes it doesn't).
* configure.ac:
* tests/files/Makefile.am:
* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
Add directory where data for unit tests can be stored.
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
(read_tags_from_file), (run_check_for_file),
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
Add unit test for id3demux, and in particular for bug #407349. Only
testing pull-mode for now; push mode doesn't work yet because the test
files are smaller than ID3_TYPE_FIND_MIN_SIZE.
2007-03-06 23:19:30 +00:00
Tim-Philipp Müller
8ffc1761b3 gst/id3demux/: Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise the four-digit number will be interp...
Original commit message from CVS:
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_obsolete_tdat_frame):
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
the four-digit number will be interpreted as a year, whereas it is
month and day in DDMM format. Instead, parse TDAT frames and fix up
the date in the GST_TAG_DATE tag later if we also extracted a year.
Fixes #407349.
2007-03-06 18:16:49 +00:00
Wim Taymans
57145cecf3 gst/spectrum/gstspectrum.c: Fix and cleanup default property values.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_transform_ip):
Fix and cleanup default property values.
Add FIXMEs for stuff that looks rather wrong.
2007-03-06 13:57:55 +00:00
Wim Taymans
0dcf0cebb9 gst/goom/gstgoom.*: Document, fix and improve goom adapter behaviour.
Original commit message from CVS:
* gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
(gst_goom_chain):
* gst/goom/gstgoom.h:
Document, fix and improve goom adapter behaviour.
Fixes #407006.
2007-03-06 13:21:23 +00:00
Wim Taymans
20f18abf72 gst/rtp/: Fix encoding-name case.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Fix encoding-name case.
2007-03-05 17:08:32 +00:00
Wim Taymans
d3948d2323 gst/rtp/: Fix speex (de)payloader. Fixes #358040.
Original commit message from CVS:
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
(gst_rtp_speex_depay_process):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
(gst_rtp_speex_pay_change_state):
* gst/rtp/gstrtpspeexpay.h:
Fix speex (de)payloader. Fixes #358040.
2007-03-05 16:39:29 +00:00
Jan Schmidt
2229ae3f98 gst/multipart/multipartdemux.c: Use gst_pad_new_from_static_template instead of static_pad_template_get+pad_new.
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
(gst_multipart_find_pad_by_mime):
Use gst_pad_new_from_static_template instead of
static_pad_template_get+pad_new.
2007-03-04 15:07:15 +00:00
Jan Schmidt
de1357a407 Fix a bunch of leaks shown by the newly-added states test.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
(gst_gconf_audio_src_finalize), (do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
(gst_gconf_video_src_finalize), (do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
(gst_switch_sink_reset), (gst_switch_sink_set_child):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_finalize):
* gst/debug/testplugin.c: (gst_test_class_init),
(gst_test_finalize):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(gst_flxdec_dispose):
* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
* gst/rtsp/rtspextwms.h:
* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
(gst_smpte_finalize):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
(gst_udpsink_finalize):
* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
(gst_wavparse_sink_activate):
* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
(gst_oss_src_finalize):
* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_finalize):
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
Fix a bunch of leaks shown by the newly-added states test.
2007-03-04 13:52:03 +00:00
Loïc Minier
63886c8b3c Don't mix tabs and spaces (#414168).
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* ext/libcaca/Makefile.am:
* gst/debug/Makefile.am:
Don't mix tabs and spaces (#414168).
2007-03-03 13:06:21 +00:00
Stefan Kost
c0b3a18684 gst/wavparse/gstwavparse.c: Unbreak my previous commit (swapped nominator & denominator). Tim, thanks for spotting.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Unbreak my previous commit (swapped nominator & denominator). Tim,
thanks for spotting.
2007-03-02 21:01:19 +00:00
Wim Taymans
823b49268f gst/udp/gstudpsrc.c: Fix doc.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create):
Fix doc.
2007-03-02 13:40:06 +00:00
René Stadler
a8b5f90ed0 gst/wavparse/gstwavparse.c: Handle rounding better to not drop last sample frame. Fixes #356692
Original commit message from CVS:
Patch by: René Stadler <mail@renestadler.de>
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Handle rounding better to not drop last sample frame. Fixes #356692
2007-03-02 13:29:25 +00:00
Thijs Vermeir
afd829326e gst/udp/gstudpsrc.*: Add support to strip proprietary headers. Fixes #350296.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property):
* gst/udp/gstudpsrc.h:
Add support to strip proprietary headers. Fixes #350296.
2007-03-02 12:56:13 +00:00
Wim Taymans
2bd9964f12 gst/rtp/gstrtpmp2tdepay.c: Fix compilation.
Original commit message from CVS:
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
Fix compilation.
2007-03-02 12:52:56 +00:00
Thijs Vermeir
fe901ccec7 gst/rtp/gstrtpmp2tdepay.*: Add support to strip off proprietary headers. Fixes #350278.
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init),
(gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process),
(gst_rtp_mp2t_depay_set_property),
(gst_rtp_mp2t_depay_get_property):
* gst/rtp/gstrtpmp2tdepay.h:
Add support to strip off proprietary headers. Fixes #350278.
2007-03-02 12:16:16 +00:00
Wim Taymans
84c6cb989a gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all of them are in error.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
Errors from the udp sources are not fatal unless all of them are in
error.
2007-03-01 18:47:28 +00:00
Wim Taymans
dc212cdb3d gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc...
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
2007-03-01 09:29:34 +00:00
Wim Taymans
83676ebd17 gst/avi/gstavidemux.c: Fix combined flow return. Fixes #412608.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop), (gst_avi_demux_chain):
Fix combined flow return. Fixes #412608.
2007-02-28 10:54:55 +00:00
Wim Taymans
dcdaf922c4 gst/videofilter/Makefile.am: Dist header..
Original commit message from CVS:
* gst/videofilter/Makefile.am:
Dist header..
2007-02-28 10:41:14 +00:00
Wim Taymans
3ed5e28e20 gst/videofilter/gstgamma.h: Add header too.
Original commit message from CVS:
* gst/videofilter/gstgamma.h:
Add header too.
2007-02-28 10:29:08 +00:00
Mark Nauwelaerts
18f3209f29 gst/videofilter/: Port gamma filter to 0.10. Fixes #412704.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videofilter/Makefile.am:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
(gst_gamma_get_property), (gst_gamma_calculate_tables),
(oil_tablelookup_u8), (gst_gamma_set_caps),
(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
Port gamma filter to 0.10. Fixes #412704.
* tests/check/Makefile.am:
* tests/check/elements/videofilter.c: (setup_filter),
(cleanup_filter), (check_filter), (GST_START_TEST),
(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
Add unit tests for videofilters.
2007-02-28 10:17:15 +00:00
Wim Taymans
3a6dd1e4bf gst/rtsp/URLS: Add another interesting test url.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another interesting test url.
* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
Don't allow getting header fields from data packets.
2007-02-28 10:06:27 +00:00
Jan Schmidt
08470e221b gst/rtsp/Makefile.am: Fix make check too.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
Fix make check too.
2007-02-26 12:07:14 +00:00
Jan Schmidt
ff1a71edf9 gst/rtsp/base64.*: Commit missing files for base64 encoding.
Original commit message from CVS:
* gst/rtsp/base64.c: (util_base64_encode):
* gst/rtsp/base64.h:
Commit missing files for base64 encoding.
2007-02-26 10:00:28 +00:00
Loïc Minier
682312a296 Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/annodex/Makefile.am:
* ext/jpeg/Makefile.am:
* ext/speex/Makefile.am:
* gst/alpha/Makefile.am:
* gst/cutter/Makefile.am:
* gst/debug/Makefile.am:
* gst/effectv/Makefile.am:
* gst/goom/Makefile.am:
* gst/level/Makefile.am:
* gst/smpte/Makefile.am:
* gst/videofilter/Makefile.am:
Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
2007-02-24 22:57:49 +00:00
Tim-Philipp Müller
e854c41c2f Fix build with LDFLAGS='-Wl,-z,defs'.
Original commit message from CVS:
* configure.ac:
* ext/gsm/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/wavpack/Makefile.am:
* gst/equalizer/Makefile.am:
* gst/filter/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/replaygain/Makefile.am:
* gst/speed/Makefile.am:
Fix build with LDFLAGS='-Wl,-z,defs'.
2007-02-24 22:52:47 +00:00
Jan Schmidt
825cf238bb gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (append_auth_header),
(rtsp_connection_send), (rtsp_connection_set_auth):
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
from icecast to replace it. Relicensed from GPL courtesy of Mike
Smith.
2007-02-23 19:12:52 +00:00
Jan Schmidt
66df66daa2 gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
Stefan Kost
5c1b116dc8 Fix level for multi-channel case.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_caps),
(gst_level_transform_ip):
* sys/v4l2/README:
* tests/check/elements/level.c: (GST_START_TEST):
Fix level for multi-channel case.
2007-02-22 14:35:28 +00:00
Stefan Kost
6e44a9c618 gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
2007-02-21 10:18:12 +00:00
Wim Taymans
bd4b1f680c gst/rtp/: Added simple mpeg transport stream payloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
(gst_rtp_mp2t_pay_plugin_init):
* gst/rtp/gstrtpmp2tpay.h:
Added simple mpeg transport stream payloader.
2007-02-18 13:24:26 +00:00
Wim Taymans
7fd025043d gst/rtsp/URLS: Add example H264 rtsp url.
Original commit message from CVS:
* gst/rtsp/URLS:
Add example H264 rtsp url.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
Don't convert values to lowercase or we might mess up base64 encoded
properties.
2007-02-16 12:32:01 +00:00
Wim Taymans
dc325990e0 gst/rtp/README: Fix case of string params.
Original commit message from CVS:
* gst/rtp/README:
Fix case of string params.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Fix depayloader, support more packet types.
Add sync codes to make sure the packetizer can do its job.
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
Fix caps case again.
2007-02-16 12:30:22 +00:00
Wim Taymans
e4b3dce677 gst/rtp/gstrtph264depay.c: Set right caps on output buffers.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Set right caps on output buffers.
2007-02-15 12:26:28 +00:00
Wim Taymans
df5916db2f gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
As spotted by: Peter Kjellerstedt  <pkj at axis com>:
Clear stack allocated SDPMedia struct before calling _init() on it.
Clarify this in the docs as well.
2007-02-14 17:04:47 +00:00
jp.liu
6021b92465 gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
(sdp_parse_line):
* gst/rtsp/sdpmessage.h:
Based on patch by: jp.liu <jp_liu at astrocom dot cn>
Fix memory management of SDP messages. Fixes #407793.
2007-02-14 15:24:50 +00:00
zhangfei gao
d08a7da76b gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
Original commit message from CVS:
Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>
* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
2007-02-14 12:07:01 +00:00
jp.liu
a8f72c67d1 gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.
Original commit message from CVS:
Patch by: jp.liu <jp_liu at astrocom dot cn>
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix parsing of password field in url. Fixes #407797.
2007-02-14 10:09:12 +00:00
Wim Taymans
2644d7178b gst/wavparse/gstwavparse.*: Update docs.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes #405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
2007-02-14 09:55:47 +00:00
Jan Schmidt
b1aa8fef18 Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. ...
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
2007-02-13 16:01:29 +00:00
Stefan Kost
5116ff603e gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
2007-02-13 11:57:18 +00:00
Tim-Philipp Müller
ecc16f3e31 gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS (but no LIBS, since we only use define...
Original commit message from CVS:
* gst/monoscope/Makefile.am:
* gst/monoscope/gstmonoscope.c:
Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
(but no LIBS, since we only use defines from the headers).
2007-02-12 23:35:16 +00:00
Jonathan Matthew
9c49fa7113 gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode due to
Original commit message from CVS:
Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
(gst_wavparse_stream_data):
Fix massive memory leak when operating in streaming mode due to
GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
Fixes #407057.
2007-02-12 23:27:31 +00:00
Stefan Kost
114afecd8d gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs t...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Save some memory (8%) by repacking the index entry structure (more to
come). Add more FIXMEs to questionable parts.
2007-02-12 15:29:44 +00:00
Stefan Kost
14d79a36f3 gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR.
Original commit message from CVS:
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
(gst_goom_change_state):
* gst/goom/gstgoom.h:
Improved docs and use GST_DEBUG_FUNCPTR.
* gst/level/gstlevel.c: (gst_level_class_init):
Use GST_DEBUG_FUNCPTR.
* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
(gst_monoscope_chain), (gst_monoscope_change_state):
Improved docs source cleanups.
2007-02-12 12:43:00 +00:00
Tim-Philipp Müller
84c6815cf7 gst/debug/: Add code for a pushfilesrc element that implements a pushfile:// URI handler, to make debugging push-mode...
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/gstdebug.c: (plugin_init):
* gst/debug/gstpushfilesrc.c:
* gst/debug/gstpushfilesrc.h:
Add code for a pushfilesrc element that implements a pushfile:// URI
handler, to make debugging push-mode operation of demuxer/decoders
that support both easier in connection with seek/playbin/etc.
The element isn't registered at the moment.
2007-02-12 10:29:57 +00:00
Sébastien Moutte
9c8ea35617 gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Comment a #if 0 in caps template definition as VS6 seems to
do not support it.
* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
Use gst_guint64_to_gdouble for conversion.
* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
Move variables declaration before the first instruction.
* gst/rtsp/rtspdefs.c:(rtsp_strresult):
Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
And don't include netdb.h for G_OS_WIN32
* gst/rtsp/sdpmessage.c:(sdp_parse_line):
This initialization SDPMedia nmedia = {.media = NULL }; is not supported
by VS6 then use an other way to initialize SDPMedia structure.
* gst/udp/gstdynudpsink.h:
* gst/udp/gstdynudpnetutils.h:
Do not include <sys/time.h> for G_OS_WIN32
* gst/udp/gstudpsrc.c:
Define socklen_t as int for G_OS_WIN32
* win/common/config.h.in:
Undef HAVE_NETINET_IN_H
* win32/vs6/gst_plugins_good.dsw:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgstrtsp.dsp:
* win32/vs6/libgstautogen.dsp:
* win32/vs6/libgstaudiofx.dsp:
* win32/vs6/libgstudp.dsp:
Add and update project files.
* win32/common/gstudp-enumtypes.c:
* win32/common/gstudp-enumtypes.h:
Add a copy of udp enumtypes to win32/common as in core
and base.
2007-02-11 12:57:47 +00:00
Tim-Philipp Müller
d8f5483d85 gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer in order to avoid compiler warnings on s...
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
Explicitly cast result of pointer arithmetic to integer in order to
avoid compiler warnings on some 64-bit systems. Should fix #406018.
2007-02-09 09:24:58 +00:00
Tim-Philipp Müller
6bbee3202a gst/debug/progressreport.c: Some more docs.
Original commit message from CVS:
* gst/debug/progressreport.c:
Some more docs.
2007-02-08 11:09:15 +00:00
Tim-Philipp Müller
ba2af9fa12 docs/plugins/inspect/plugin-rtp.xml: Update for new elements.
Original commit message from CVS:
* docs/plugins/inspect/plugin-rtp.xml:
Update for new elements.
* gst/debug/progressreport.h:
Commit newly-created header file as well.
2007-02-07 21:09:45 +00:00
Tim-Philipp Müller
b5ee422546 Make progressreport element post messages with the current progress on the bus. Also add some basic docs for it.
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/gst-plugins-good-plugins.hierarchy:
* gst/debug/Makefile.am:
* gst/debug/progressreport.c: (gst_progress_report_post_progress),
(gst_progress_report_do_query), (gst_progress_report_report):
Make progressreport element post messages with the current progress
on the bus. Also add some basic docs for it.
2007-02-07 20:39:16 +00:00
Tim-Philipp Müller
2a873dd98e gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
Let's try this again and use the right cast this time.
2007-02-06 16:29:30 +00:00
Tim-Philipp Müller
7dd530e6c4 gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib versions where the nick/name members in GEn...
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
Add cast to avoid compiler warnings with older GLib versions
where the nick/name members in GEnumValue are not declared as
constant strings.
2007-02-06 16:24:57 +00:00
Sebastian Dröge
cdba2c4219 gst/audiofx/: Some small cleanups and port both elements to the new GstAudioFilter base class to save a few lines of ...
Original commit message from CVS:
* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
(gst_audio_amplify_class_init), (gst_audio_amplify_init),
(gst_audio_amplify_set_process_function),
(gst_audio_amplify_setup):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_setup):
* gst/audiofx/audioinvert.h:
Some small cleanups and port both elements to the new GstAudioFilter
base class to save a few lines of common code.
* gst/audiofx/Makefile.am:
Link against libgstaudio for the above changes
2007-02-06 11:16:49 +00:00
Tim-Philipp Müller
f7935f9a40 Fix up to use the newly ported (actually working) GstAudioFilter.
Original commit message from CVS:
* configure.ac:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init),
(gst_iir_equalizer_class_init), (gst_iir_equalizer_init),
(setup_filter), (gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup),
(plugin_init):
* gst/equalizer/gstiirequalizer.h:
Fix up to use the newly ported (actually working) GstAudioFilter.
Bump core/base requirements to CVS for this.
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/equalizer-test.c: (check_bus),
(equalizer_set_band_value), (equalizer_set_all_band_values),
(equalizer_set_band_value_and_wait),
(equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
(main):
Add brain-dead interactive test for equalizer.
2007-02-03 23:35:26 +00:00
Tim-Philipp Müller
8996dbb3f9 gst/equalizer/gstiirequalizer.c: Rename "values" property to "band-values" and change type into a
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace):
Rename "values" property to "band-values" and change type into a
GValueArray, so it's more easily bindable and the range of the
values passed in is defined and checked etc.; also do some
locking.
2007-02-02 18:36:28 +00:00
James Doc Livingston
4655cbd45d Port equalizer plugin to 0.10 (#403572).
Original commit message from CVS:
Patch by: James "Doc" Livingston  <doclivingston at gmail com>
* configure.ac:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
Port equalizer plugin to 0.10 (#403572).
2007-02-02 17:39:21 +00:00
Tim-Philipp Müller
726254bdde gst/videocrop/gstvideocrop.c: Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps),
(gst_video_crop_transform_packed_complex):
Fix cropping for packed 4:2:2 formats YUYV/YUY2 and UYVY.
* tests/icles/videocrop-test.c: (check_bus_for_errors),
(test_with_caps), (main):
Block streaming thread before changing filter caps while the
pipeline is running so that we don't get random not-negotiated
errors just because GStreamer can't handle that yet.
2007-01-28 18:28:33 +00:00
Wim Taymans
2de7376aaf gst/rtsp/gstrtspsrc.c: Convert SDP fields to upper/lowercase following the rules in the SDP to caps document.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_activate_streams):
Convert SDP fields to upper/lowercase following the rules in the SDP to
caps document.
2007-01-25 14:40:15 +00:00
Wim Taymans
22eb34e2fe gst/rtp/: Fix case of encoding-name and key/value pairs to match the document.
Original commit message from CVS:
* gst/rtp/README:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpilbcpay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix case of encoding-name and key/value pairs to match the document.
This is to make interoperation with SDP case-insensitive as required by
the relevant RFCs.
2007-01-25 14:22:53 +00:00
Edward Hervey
a02af52f4e gst/: Use proper print statements.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
(gst_multi_file_sink_class_init):
* gst/multifile/gstmultifilesrc.c: (gst_multi_file_src_class_init):
* gst/mve/gstmvedemux.c: (gst_mve_video_create_buffer),
(gst_mve_video_palette), (gst_mve_video_code_map),
(gst_mve_audio_init), (gst_mve_audio_data), (gst_mve_timer_create),
(gst_mve_demux_chain):
* gst/mve/gstmvemux.c: (gst_mve_mux_push_chunk):
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/mve/mvevideodec16.c: (ipvideo_copy_block):
* gst/mve/mvevideodec8.c: (ipvideo_copy_block):
* gst/mve/mvevideoenc16.c: (mve_encode_frame16):
* gst/mve/mvevideoenc8.c: (mve_encode_frame8):
Use proper print statements.
Fixes build on mac os x.
<wingo> oo look at me my name is edward i'm hacking on macos wooo
2007-01-25 12:05:11 +00:00
Wim Taymans
40d06b6a55 gst/rtp/gstrtpL16pay.*: Fill up to MTU using adapter.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
* gst/rtp/gstrtpL16pay.h:
Fill up to MTU using adapter.
Timestamp rtp packets.
2007-01-25 10:54:19 +00:00
Edward Hervey
d7666d033c Use G_GSIZE_FORMAT in print statements for portability.
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
Use G_GSIZE_FORMAT in print statements for portability.
Fixes build on macosx.
2007-01-25 10:36:35 +00:00
Wim Taymans
85420195b2 gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
(gst_rtp_L16_depay_plugin_init):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
(gst_rtp_L16_pay_plugin_init):
* gst/rtp/gstrtpL16pay.h:
Port and enable raw audio payloader/depayloader. Needs a bit more work
on the payloader side.
2007-01-24 18:20:14 +00:00
Wim Taymans
a6a9207c42 gst/rtsp/gstrtspsrc.*: Only unblock the udp pads when we linked and activated them all.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
* gst/rtsp/gstrtspsrc.h:
Only unblock the udp pads when we linked and activated them all.
Fixes #395688.
2007-01-24 16:25:55 +00:00
Wim Taymans
f083178741 gst/rtp/: Added simple AC3 depayloader (RFC 4184).
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
* gst/rtp/gstrtpac3depay.h:
Added simple AC3 depayloader (RFC 4184).
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
Fix a leak.
2007-01-24 15:18:34 +00:00
Sebastian Dröge
54b10ebf2a gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
2007-01-24 12:41:03 +00:00
Wim Taymans
1f51fd9785 gst/rtsp/gstrtspsrc.c: Unblock pads after adding the pads to the element so that autopluggers get a change to link so...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
Unblock pads after adding the pads to the element so that autopluggers
get a change to link something. Possibly fixes #395688.
2007-01-24 12:26:41 +00:00
Wim Taymans
3df533de2c gst/rtp/: Fix caps with payload numbers.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp2tdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
(gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtp/gstrtpsv3vdepay.c:
* gst/rtp/gstrtptheoradepay.c:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix caps with payload numbers.
Add some fixed payload numbers to caps when possible.
2007-01-24 12:22:51 +00:00
Wim Taymans
1cf20feb6e gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c:
Fix caps on the depayloader.
2007-01-24 11:29:00 +00:00
Sebastian Dröge
447ae144c2 gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes #396057
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
2007-01-23 18:16:09 +00:00
Wim Taymans
60054f479a gst/rtp/gstrtpmp4gdepay.c: Parse config params as string and int.
Original commit message from CVS:
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
Parse config params as string and int.
Parse and use AU header length
2007-01-23 17:36:32 +00:00
Wim Taymans
168db53bf4 gst/smpte/: constify some static structs.
Original commit message from CVS:
* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
* gst/smpte/gstmask.c: (_gst_mask_register):
* gst/smpte/gstmask.h:
* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
(gst_smpte_paint_triangle_clock):
constify some static structs.
Don't update the mask if nothing changed to the params.
Make sure we never draw outside of the picture. Fixes #398325.
2007-01-23 17:27:39 +00:00
Tim-Philipp Müller
a10f2478bb gst/avi/gstavidemux.c: Error out properly when pull_range fails while we're reading the headers, instead of just paus...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
Error out properly when pull_range fails while we're reading the
headers, instead of just pausing the task silently. Fixes #399338.
2007-01-22 13:06:43 +00:00
Tim-Philipp Müller
813c331abd gst/smpte/gstsmpte.c: Some more sanity checks to make sure the input formats match and the input pads are actually ne...
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Some more sanity checks to make sure the input formats match and the
input pads are actually negotiated, in case someone tries to feed
buffers from fakesrc or filesrc. Fixes #398299.
Also const-ify an array, just because we can.
2007-01-19 13:06:07 +00:00
Edward Hervey
3206d6ee5e gst/smpte/gstsmpte.c: Ignore previous commit, that was only valid for widths and heights that are multiples of 4.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
Ignore previous commit, that was only valid for widths and heights
that are multiples of 4.
Copy over size/stride macros from jpegdec. This allows the element
to work with any width,height...
... but puts in evidence that the actual transformations only work
with width/height that are multiples of 4.
2007-01-19 10:35:13 +00:00
Edward Hervey
5d45f48fca gst/smpte/gstsmpte.c: Allocate buffers of the right size.
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_collected):
Allocate buffers of the right size.
The proper size of a I420 buffer in bytes is:
width * height * 3
------------------
2
2007-01-19 09:48:47 +00:00
Tim-Philipp Müller
914b79faa6 gst/smpte/gstsmpte.c: Proxy getcaps on sink pads too, so that we either end up with the same dimensions on all pads o...
Original commit message from CVS:
* gst/smpte/gstsmpte.c: (gst_smpte_init):
Proxy getcaps on sink pads too, so that we either end up with the
same dimensions on all pads or error out if that's not possible
(seems to work even!). Fixes #398086, I think.
2007-01-18 18:37:39 +00:00
Stefan Kost
8000e45c5b gst/audiofx/audiopanorama.c: Fix doc section name (Fixes #397946)
Original commit message from CVS:
* gst/audiofx/audiopanorama.c:
Fix doc section name (Fixes #397946)
2007-01-18 11:23:36 +00:00
Sebastian Dröge
703a0d00d8 gst/audiofx/audiopanorama.c: Use a function array for process methods, add more docs and define the startindex of enums.
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo circular-chaos org>
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
(gst_audio_panorama_set_process_function):
Use a function array for process methods, add more docs and define the
startindex of enums.
2007-01-16 08:29:11 +00:00
Mark Nauwelaerts
36dfafcda9 Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps...
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
* tests/check/elements/avimux.c: (teardown_src_pad):
Add support for more than one audio stream; write better AVIX
header; refactor code a bit; don't announce vorbis caps on our audio
sink pads since we don't support it anyway. Closes #379298.
2007-01-14 17:55:33 +00:00
Andy Wingo
1509c2efcc gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads): Use fixed caps on src pads.
Original commit message from CVS:
2007-01-13  Andy Wingo  <wingo@pobox.com>

* gst/interleave/deinterleave.c (gst_deinterleave_add_new_pads):
Use fixed caps on src pads.
(gst_deinterleave_remove_pads): Remove src pads, not sink pads. I
seem to have reverse midas disease!
(gst_deinterleave_process): Proxy timestamps, offsets, durations,
and set caps on outgoing buffers. Fixes #395597, I think.
2007-01-13 19:12:32 +00:00
Andy Wingo
10a685a940 gst/interleave/interleave.c (gst_interleave_init): Init the activation mode properly.
Original commit message from CVS:
2007-01-13  Andy Wingo  <wingo@pobox.com>

* gst/interleave/interleave.c (gst_interleave_init): Init the
activation mode properly.
(gst_interleave_src_setcaps, gst_interleave_src_getcaps)
(gst_interleave_init): Set a setcaps and getcaps function on the
src pad, so that we can implement pull-mode negotiation.
(gst_interleave_sink_setcaps): Renamed from
gst_interleave_setcaps, as it only does the sink logic now.
Implement both for pull-mode and push-mode.
(gst_interleave_process): Set caps on our outgoing buffer.
(gst_interleave_src_activate_pull): Fix some more bogus casts.
What is up with this.
2007-01-13 18:01:41 +00:00
Sebastian Dröge
22ebbb6912 gst/audiofx/audiopanorama.*: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo circular-chaos org>
* gst/audiofx/audiopanorama.c:
(gst_audio_panorama_method_get_type),
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
(gst_audio_panorama_set_process_function),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s_int_simple),
(gst_audio_panorama_transform_s2s_int_simple),
(gst_audio_panorama_transform_m2s_float_simple),
(gst_audio_panorama_transform_s2s_float_simple):
* gst/audiofx/audiopanorama.h:
Add 'method' property and provide a simple (non-psychoacustic)
processing method (#394859).
* tests/check/elements/audiopanorama.c: (GST_START_TEST),
(panorama_suite):
Tests for new method.
2007-01-13 15:52:18 +00:00
Wim Taymans
c7839a6aa7 gst/qtdemux/: Add X-QT depayloader that will eventually share code with the demuxer.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_base_init),
(gst_rtp_xqt_depay_class_init), (gst_rtp_xqt_depay_init),
(gst_rtp_xqt_depay_finalize), (gst_rtp_quicktime_parse_sd),
(gst_rtp_xqt_depay_setcaps), (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_set_property), (gst_rtp_xqt_depay_get_property),
(gst_rtp_xqt_depay_change_state), (gst_rtp_xqt_depay_plugin_init):
* gst/qtdemux/gstrtpxqtdepay.h:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_base_init),
(gst_qtdemux_loop_state_header), (gst_qtdemux_loop),
(qtdemux_parse_moov), (qtdemux_parse_container),
(qtdemux_parse_node), (gst_qtdemux_add_stream),
(qtdemux_parse_trak), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/quicktime.c: (plugin_init):
Add X-QT depayloader that will eventually share code with the demuxer.
Make new plugin entry point with quicktime releated stuff.
2007-01-12 17:16:51 +00:00
Tim-Philipp Müller
9003c60563 gst/qtdemux/Makefile.am: Dist all new files.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
Dist all new files.
2007-01-12 12:10:19 +00:00
Wim Taymans
a09ea6cce4 gst/qtdemux/: Cleanup and refactor to make the code more readable.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (extract_initial_length_and_fourcc),
(gst_qtdemux_loop_state_header), (gst_qtdemux_combine_flows),
(gst_qtdemux_loop_state_movie), (gst_qtdemux_loop),
(gst_qtdemux_chain), (qtdemux_sink_activate_pull),
(qtdemux_inflate), (qtdemux_parse_moov), (qtdemux_parse_container),
(qtdemux_parse_node), (qtdemux_tree_get_child_by_type),
(qtdemux_tree_get_sibling_by_type), (gst_qtdemux_add_stream),
(qtdemux_parse_samples), (qtdemux_parse_segments),
(qtdemux_parse_trak), (qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_date), (qtdemux_tag_add_gnre),
(qtdemux_parse_udta), (qtdemux_redirects_sort_func),
(qtdemux_process_redirects), (qtdemux_parse_redirects),
(qtdemux_parse_tree), (gst_qtdemux_handle_esds),
(qtdemux_video_caps), (qtdemux_audio_caps):
* gst/qtdemux/qtdemux.h:
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mvhd),
(qtdemux_dump_tkhd), (qtdemux_dump_elst), (qtdemux_dump_mdhd),
(qtdemux_dump_hdlr), (qtdemux_dump_vmhd), (qtdemux_dump_dref),
(qtdemux_dump_stsd), (qtdemux_dump_stts), (qtdemux_dump_stss),
(qtdemux_dump_stsc), (qtdemux_dump_stsz), (qtdemux_dump_stco),
(qtdemux_dump_co64), (qtdemux_dump_dcom), (qtdemux_dump_cmvd),
(qtdemux_dump_unknown), (qtdemux_node_dump_foreach),
(qtdemux_node_dump):
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c: (qtdemux_type_get):
* gst/qtdemux/qtdemux_types.h:
* gst/qtdemux/qtpalette.h:
Cleanup and refactor to make the code more readable.
Move debugging/tables into separate files.
Add 2/4/16 color palletee support.
Fix raw 15 bit RGB handling.
Use more FOURCC constants.
Add some docs.
2007-01-12 10:22:16 +00:00
Tim-Philipp Müller
1e364d04f5 gst/: Set correct caps on outgoing pulled buffers, or things blow up after recent core changes.
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
Set correct caps on outgoing pulled buffers, or things blow up
after recent core changes.
2007-01-11 16:59:40 +00:00
Jonas Holmberg
5c1a7a9260 gst/multipart/multipartmux.c: Return FLOW errors ASAP. Fixes #394977.
Original commit message from CVS:
Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
* gst/multipart/multipartmux.c: (gst_multipart_mux_init),
(gst_multipart_mux_request_new_pad),
(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
(gst_multipart_mux_change_state):
Return FLOW errors ASAP. Fixes #394977.
Misc cleanups.
2007-01-11 11:05:04 +00:00
Lutz Mueller
cfed610d01 gst/rtsp/gstrtspsrc.c: Check for stream pad before activating.
Original commit message from CVS:
Patch by: Lutz Mueller <lutz at topfrose dot de>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
Check for stream pad before activating.
2007-01-11 09:30:59 +00:00
Peter Kjellerstedt
12ab127d12 gst/rtsp/: Allow url to be NULL to be able to use it for server connections.
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst/rtsp/COPYING.MIT:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_open), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_send), (read_line),
(parse_request_line), (parse_line), (rtsp_connection_read),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
(rtsp_method_as_text), (rtsp_header_as_text),
(rtsp_status_as_text), (rtsp_find_header_field),
(rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
(rtsp_ext_wms_configure_stream):
* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
(rtsp_message_new_request), (rtsp_message_init_request),
(rtsp_message_new_response), (rtsp_message_init_response),
(rtsp_message_init_data), (rtsp_message_unset),
(rtsp_message_free), (rtsp_message_add_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
(sdp_message_dump):
Allow url to be NULL to be able to use it for server connections.
Can now send responses as well as requests.
No longer hangs in an endless loop if EOF is received.
Can now convert a status code to a text string.
Return RTSP_HDR_INVALID for unknown headers.
Return RTSP_INVALID for unknown methods.
Copy CSeq and Session headers from the request.
Only free memory corresponding to the currently set message type.
Added const to function arguments as appropriate.
Avoid a compiler warning when initializing nmedia.
Use guint rather than gint to avoid compiler warnings.
Fix crasher in wms extension.
Factor out stream setup from open_connection.
Delay activation of streams when actual data is received from the
server, this prepares us to do proper protocol switching.
Added new license.
Fixes #380895.
2007-01-10 15:19:48 +00:00
Sebastian Dröge
8f7c1775d9 Some small docs fixes (#394851).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo ubuntu com>
* docs/plugins/Makefile.am:
* gst/audiofx/audiopanorama.c:
Some small docs fixes (#394851).
2007-01-10 09:47:43 +00:00
Wim Taymans
5aadb77a1d gst/avi/gstavidemux.c: Fix docs.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Fix docs.
2007-01-09 12:25:26 +00:00
Wim Taymans
42b8b3a37f gst/rtp/: Added RFC 2250 MPEG Video Depayloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
* gst/rtp/gstrtpmpvdepay.h:
Added RFC 2250 MPEG Video Depayloader.
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
Fix Header file. Small cleanups.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
(gst_rtp_mp4v_depay_change_state):
Remove usused code. Remove Adapter from state Change. Added debug.
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpadepay.h:
Subclass base depayloader.
Added debug.
Support static payload type assignment as well.
* gst/rtp/gstrtpmpapay.c:
Fix caps.
2007-01-09 12:23:48 +00:00
Vincent Torri
fd18506657 ext/jpeg/: These libjpeg callbacks should return a 'boolean' (unsigned char apparently) and not a 'gboolean' (which m...
Original commit message from CVS:
Patch by: Vincent Torri  <vtorri at univ-evry fr>
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/smokecodec.c:
These libjpeg callbacks should return a 'boolean' (unsigned char
apparently) and not a 'gboolean' (which maps to gint). Fixes
warnings when compiling with MingW (#393427).
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
Use ioctlsocket on win32.
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Some printf format fixes for win32.
2007-01-08 12:45:10 +00:00
Andy Wingo
12359919d3 New elements interleave and deinterleave, implement channel interleaving and deinterleaving.
Original commit message from CVS:
2007-01-07  Andy Wingo  <wingo@pobox.com>

* configure.ac:
* gst/interleave/Makefile.am:
* gst/interleave/plugin.h:
* gst/interleave/plugin.c:
* gst/interleave/interleave.c:
* gst/interleave/deinterleave.c: New elements interleave and
deinterleave, implement channel interleaving and deinterleaving.
The interleaver can operate in pull or push mode but the
deinterleaver is more like a demuxer and can only operate in push
mode.
2007-01-07 22:03:54 +00:00
Sébastien Moutte
8d2ac1002c gst/cutter/gstcutter.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst/cutter/gstcutter.c: (gst_cutter_chain):
Use gst_guint64_to_gdouble for conversion.
* win32/vs6/libgstmatroska.dsp:
Add zlib to the link.
* win32/vs6/libgstvideobox.dsp:
Update liboil library name (project is linked to liboil-0.3-0.lib now).
2007-01-07 10:44:12 +00:00
Tim-Philipp Müller
9445ca84f5 Check for zlib and if available pass it explicitly to the linker when linking qtdemux. If not available (or --disable...
Original commit message from CVS:
* configure.ac:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_moov):
Check for zlib and if available pass it explicitly to the linker
when linking qtdemux. If not available (or --disable-external has
been specified!), disable the bits in qtdemux that use it. Fixes
build on MingW (#392856).
2007-01-05 18:32:03 +00:00
Tim-Philipp Müller
5d78ae0a1c gst/matroska/Makefile.am: If zlib is available and used, we must link it explicitly for things to work on MingW (fixe...
Original commit message from CVS:
* gst/matroska/Makefile.am:
If zlib is available and used, we must link it explicitly for
things to work on MingW (fixes #392855).
2007-01-05 17:23:04 +00:00
Jens Granseuer
fa57a52f69 Fix build with gcc-2.x (declare variables at the beginning of a block etc.). Fixes #391971.
Original commit message from CVS:
Patch by: Jens Granseuer  <jensgr at gmx net>
* ext/xvid/gstxvidenc.c: (gst_xvidenc_encode),
(gst_xvidenc_get_property):
* gst/filter/gstbpwsinc.c: (bpwsinc_transform_ip):
* gst/filter/gstfilter.c: (plugin_init):
* gst/filter/gstiir.c: (iir_transform_ip):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform_ip):
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c: (gst_nuv_demux_header_load),
(gst_nuv_demux_stream_extend_header):
Fix build with gcc-2.x (declare variables at the beginning of a
block etc.). Fixes #391971.
2007-01-03 16:41:10 +00:00
Tim-Philipp Müller
7735292ec2 gst/matroska/matroska-mux.c: The "signed" field in audio caps is of boolean type, trying to use gst_structure_get_int...
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_audio_pad_setcaps):
The "signed" field in audio caps is of boolean type, trying to use
gst_structure_get_int() to extract it will fail. Fixing this makes
matroskamux accept raw audio input (#387121) (use at your own risk
though, due to the matroska spec being not entirely useful in this
respect).
Also fix up raw audio structures in template caps so that they
represent what our setcaps function will actually accept, so that
converters know what to convert to.
Finally, don't fail if there isn't an "endianness" field in 8-bit
PCM caps.
2006-12-24 11:24:59 +00:00
Tim-Philipp Müller
2f353d7379 gst/qtdemux/qtdemux.c: Don't post BUFFERING messages in streaming mode if the stream headers are behind the movie dat...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_post_progress),
(gst_qtdemux_chain):
Don't post BUFFERING messages in streaming mode if the stream
headers are behind the movie data; instead, post "progress" element
messages as a temporary solution. Apps might get confused and do
silly things to the pipeline state if they see buffering messages
from different sources and don't realize they come from different
sources (#387160).
2006-12-18 17:11:49 +00:00
Jan Schmidt
7966c804ab gst/qtdemux/qtdemux.c: Don't output g_warning for an unsupported format, just send a
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_chain),
(gst_qtdemux_add_stream):
Don't output g_warning for an unsupported format, just send a
GST_ELEMENT_WARNING and don't add the pad.
Fix the case where it doesn't check for a NULL pad in streaming mode.
Fixes #387137
2006-12-18 13:40:34 +00:00
Tim-Philipp Müller
ef691f3827 gst/qtdemux/qtdemux.c: Fix crash dereferencing NULL pointer if there's no stco atom.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Fix crash dereferencing NULL pointer if there's no stco atom.
Fixes #387122.
2006-12-18 12:27:32 +00:00
Sjoerd Simons
e2f1b66fb2 gst/videomixer/videomixer.c: Introduce some locking around the videomixer state so that it does not crash when adding...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
(gst_videomixer_reset), (gst_videomixer_init),
(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
(gst_videomixer_release_pad), (gst_videomixer_collected),
(gst_videomixer_change_state):
Introduce some locking around the videomixer state so that it does not
crash when adding/removing pads. Fixes #383043.
2006-12-16 16:21:26 +00:00
Tim-Philipp Müller
40d3caa168 gst/qtdemux/qtdemux.c: We don't support seeking in streaming mode, so don't even try.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query), (gst_qtdemux_handle_src_event):
We don't support seeking in streaming mode, so don't even try.
Implement seeking query so apps can query seekability properly
(see #365414). Fix duration query.
2006-12-16 15:25:23 +00:00
Tim-Philipp Müller
59c1122481 gst/effectv/gstquark.c: Add some NULL pointer checks (possibly related to #385623).
Original commit message from CVS:
* gst/effectv/gstquark.c: (gst_quarktv_transform),
(gst_quarktv_planetable_clear):
Add some NULL pointer checks (possibly related to #385623).
2006-12-14 14:25:17 +00:00
Wim Taymans
f4dd37e871 gst/qtdemux/qtdemux.c: Add AMR-WB to the list of supported formats.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add AMR-WB to the list of supported formats.
2006-12-13 17:12:22 +00:00
Tim-Philipp Müller
173ee367e4 gst/: In streaming mode, if the first buffer we get doesn't have an offset, fix it up to be 0, otherwise trimming won...
Original commit message from CVS:
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
(gst_tag_demux_chain):
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
In streaming mode, if the first buffer we get doesn't have an
offset, fix it up to be 0, otherwise trimming won't work later on
and we'll be typefinding application/x-id3, which may result in
decodebin plugging an endless number of id3demux elements as a
consequence. Fixes #385031.
2006-12-12 18:45:58 +00:00
Tim-Philipp Müller
81c7f2c4a7 gst/qtdemux/qtdemux.c: Fix non-working redirects from inetfilm.com (handle 'alis' reference data type as well). Fixes...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_tree):
Fix non-working redirects from inetfilm.com (handle 'alis' reference
data type as well). Fixes #378613.
2006-12-11 17:33:26 +00:00
Tim-Philipp Müller
0d3b023699 gst/matroska/: Try harder to extract the framerate for video tracks correctly and save it directly instead of convert...
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_video_caps):
* gst/matroska/matroska-ids.c:
(gst_matroska_track_init_video_context):
* gst/matroska/matroska-ids.h:
Try harder to extract the framerate for video tracks correctly and
save it directly instead of converting it back and forth a few
times. Mostly makes a difference for very small framerates (<1).
Fixes #380199.
2006-12-11 13:59:33 +00:00
Sebastian Dröge
14999998d4 gst/apetag/gstapedemux.c: We need to be able to read and parse any possible floating point string format ("1,234" or ...
Original commit message from CVS:
Patch by: Sebastian Dröge  <mail at slomosnail de>
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
We need to be able to read and parse any possible floating point string
format ("1,234" or "1.234") irrespective of the current locale. g_strod()
will parse the former only in certain locales though, so we really need
to canonicalise the separator to '.' and then use g_ascii_strtod() to
make sure we can parse either version at all times.
Fixes #382982 for real.
2006-12-09 19:27:28 +00:00
René Stadler
2214d0b5bb gst/qtdemux/qtdemux.c: Fix caps for 24 bit raw PCM audio (2).
Original commit message from CVS:
Patch by: René Stadler  <mail at renestadler de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Fix caps for 24 bit raw PCM audio (2).
Fixes #383471.
2006-12-08 17:06:43 +00:00
Sebastian Dröge
6a016876c8 gst/audiofx/audiopanorama.*: Fix audiopanorame with float samples. Fixes #383726.
Original commit message from CVS:
Patch by: Sebastian Dröge  <mail at slomosnail de >
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
(gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
* gst/audiofx/audiopanorama.h:
Fix audiopanorame with float samples. Fixes #383726.
2006-12-08 16:38:18 +00:00
Wim Taymans
6b01538bca gst/smpte/: Port to 0.10 some more.
Original commit message from CVS:
* gst/smpte/gstmask.h:
* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
(gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset),
(gst_smpte_collected), (gst_smpte_set_property),
(gst_smpte_get_property), (gst_smpte_change_state), (plugin_init):
* gst/smpte/gstsmpte.h:
Port to 0.10 some more.
Added duration property to specify the duration of the transition.
Make framerate a fraction.
Deprecate fps property, we only use negotiated fps.
Added docs.
Fix collectpad usage.
Reset state in READY.
Send NEWSEGMENT event.
Fix racy updates of object properties.
Added debug category.
Fixes #383323.
2006-12-07 17:30:03 +00:00
Wim Taymans
c37faa76a9 gst/qtdemux/qtdemux.c: Handle more H263 variants.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak),
(qtdemux_video_caps):
Handle more H263 variants.
2006-12-07 11:35:41 +00:00
Sjoerd Simons
fd47c4fbf1 gst/videomixer/videomixer.c: Don't reset xpos and ypos in the setcaps function because causes unexpected behaviour.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free):
Don't reset xpos and ypos in the setcaps function because causes
unexpected behaviour.
Fixes #382179.
2006-12-06 15:06:04 +00:00
Wim Taymans
1444040b98 gst/multipart/multipartmux.c: Keep track of the buffer timestamp in the collectdata member instead of modifying the b...
Original commit message from CVS:
* gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads),
(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected):
Keep track of the buffer timestamp in the collectdata member instead
of modifying the buffer without making the metadata writable first.
Fixes #382277.
2006-12-06 14:45:30 +00:00
Rob Taylor
21c9ac41a9 gst/udp/gstudpsrc.c: If using multicast in udpsrc, bind to the multicast address rather than
Original commit message from CVS:
Patch by: Rob Taylor <robtaylor at floopily dot org>
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
If using multicast in udpsrc, bind to the multicast address rather than
IN_ADDR_ANY.
This allows the simultanous use of multiple udpsrcs listening on
different multicat addresses. Without this all udpsrcs will receive all
packets from all subscribed multicast addresses.
Fixes #383001.
2006-12-06 14:33:54 +00:00
Sebastian Dröge
4f799e000f gst/apetag/gstapedemux.c: Use g_strtod() instead of sscanf to parse doubles, so that it will try parsing in the C loc...
Original commit message from CVS:
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
Use g_strtod() instead of sscanf to parse doubles, so that it will
try parsing in the C locale if the current locale fails.
Fixes: #382982
Patch by: Sebastian Dröge  <mail at slomosnail de >
2006-12-06 13:16:59 +00:00
Stefan Kost
541a13d0e2 gst/avi/gstavimux.c: accept all mpegversions,fixes #380825 spotted by: Jerome Alet
Original commit message from CVS:
* gst/avi/gstavimux.c:
accept all mpegversions,fixes #380825
spotted by: Jerome Alet
2006-11-30 16:48:51 +00:00
René Stadler
a7f31a2117 gst/replaygain/gstrganalysis.c: Call the base class handler. Fixes #380610.
Original commit message from CVS:
Patch by: René Stadler  <mail at renestadler de>
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_event):
Call the base class handler.  Fixes #380610.
2006-11-30 15:08:08 +00:00
Wim Taymans
f249d639f8 gst/rtsp/: Add method so that extensions can choose to disable the setup of a stream.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
(rtsp_ext_wms_get_context):
Add method so that extensions can choose to disable the setup of
a stream.
Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.
2006-11-28 11:52:27 +00:00
Wim Taymans
4b8597668d gst/qtdemux/qtdemux.c: Remove some asserts and replace them with a proper error message. Fixes #379261.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Remove some asserts and replace them with a proper error
message. Fixes #379261.
2006-11-27 17:16:26 +00:00
Jonas Holmberg
6d91770165 gst/multipart/multipartmux.c: Push header in a separate buffer instead of memcpy:ing all data
Original commit message from CVS:
Patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Push header in a separate buffer instead of memcpy:ing all data
Change LF => CRLF in headers
Move trailing LF to header
2006-11-27 16:29:07 +00:00
Wim Taymans
9e782fdc7e gst/rtp/gstrtpmpadepay.c: Small buffer overflow fix and improve debugging.
Original commit message from CVS:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_chain):
Small buffer overflow fix and improve debugging.
2006-11-27 16:26:50 +00:00
Stefan Kost
90431d788e gst/avi/gstavidemux.c: remove dead code, tweak debugs statements, add comments, use _uint64_scale instead _uint64_sca...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex),
(gst_avi_demux_read_subindexes_push),
(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
remove dead code, tweak debugs statements, add comments, use
_uint64_scale instead _uint64_scale_int when using guint64 values,
small optimizations, reflow some error handling
2006-11-24 07:46:54 +00:00
Tim-Philipp Müller
f41376624b gst/videobox/gstvideobox.c: Minor clean-ups: const-ify static array, remove trailing comma from use GST_DEBUG_FUNCPTR.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
Minor clean-ups: const-ify static array, remove trailing comma from
last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR.
2006-11-19 16:30:19 +00:00
René Stadler
3b4c2b054a gst/id3demux/id3v2frames.c: Make sure that g_free always gets called on the same pointer that was returned by g_mallo...
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
Make sure that g_free always gets called on the same pointer that was
returned by g_malloc.  Fixes #376594.
Do not leak memory if decompressed size is wrong.
Remove unneeded check of return value of g_malloc.
Patch by: René Stadler <mail@renestadler.de>
2006-11-19 13:41:53 +00:00
Tim-Philipp Müller
f441440a43 gst/matroska/matroska-mux.c: Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_request_new_pad):
Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
* tests/check/elements/matroskamux.c: (setup_src_pad),
(setup_sink_pad), (GST_START_TEST):
Activate pads before using them.
2006-11-16 15:36:48 +00:00
Tim-Philipp Müller
6b16453ab4 gst/avi/gstavidemux.c: Initialise variable to get rid of bogus compiler warning.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
Initialise variable to get rid of bogus compiler warning.
2006-11-16 15:04:55 +00:00
Ville Syrjala
77fe86dc4e gst/rtp/: Specify H.263 variant and version in the caps (fixes #361637)
Original commit message from CVS:
Patch by: Ville Syrjala <ville.syrjala@movial.fi>
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
Specify H.263 variant and version in the caps (fixes #361637)
2006-11-16 07:26:17 +00:00
Wim Taymans
0cbacacba3 gst/rtsp/rtspconnection.c: Don't set a data pointer to NULL and a size > 0 when we deal with empty packets.
Original commit message from CVS:
* gst/rtsp/rtspconnection.c: (read_body):
Don't set a data pointer to NULL and a size > 0 when we deal
with empty packets.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_take_body):
Check that we can't create invalid empty packets.
2006-11-15 17:44:01 +00:00
Mark Nauwelaerts
5c446dc4f5 gst/matroska/matroska-mux.*: Add basic tag writing support; implement releasing pads (#374658).
Original commit message from CVS:
Patch by: Mark Nauwelaerts  <manauw at skynet be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces),
(gst_matroska_mux_class_init), (gst_matroska_pad_free),
(gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
(gst_matroska_mux_track_header), (gst_matroska_mux_start),
(gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish):
* gst/matroska/matroska-mux.h:
Add basic tag writing support; implement releasing pads (#374658).
2006-11-15 12:08:20 +00:00
Tim-Philipp Müller
f5dda5f00b gst/matroska/matroska-demux.c: Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_audio_caps):
Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).
2006-11-15 11:19:13 +00:00
David Schleef
4207d2fe74 gst/matroska/matroska-mux.c: Add Dirac fourcc.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: Add Dirac fourcc.
2006-11-15 00:12:19 +00:00
Wim Taymans
90ef72e564 gst/avi/gstavidemux.c: Disable init_frames delay timestamp adjustment, it does not seem to be needed at all. Fixes #3...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Disable init_frames delay timestamp adjustment, it does not
seem to be needed at all. Fixes #369621.
2006-11-14 15:55:32 +00:00
Wim Taymans
f8cb2ce5c3 gst/qtdemux/qtdemux.c: Don't parse extra sample params for raw pcm. Fixes #374914.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_trak):
Don't parse extra sample params for raw pcm. Fixes #374914.
2006-11-14 11:43:40 +00:00
Mark Nauwelaerts
27d42165b7 gst/videomixer/videomixer.c: Fix memleak by unref'ing collectpads instance (when finalizing)
Original commit message from CVS:
Patch by: Mark Nauwelaerts  <manauw at skynet be>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init),
(gst_videomixer_collect_free), (gst_videomixer_reset),
(gst_videomixer_init), (gst_videomixer_finalize),
(gst_videomixer_request_new_pad), (gst_videomixer_release_pad),
(gst_videomixer_collected), (gst_videomixer_change_state):
Fix memleak by unref'ing collectpads instance (when finalizing)
Implement releasing a request pad. Fixes #374479.
2006-11-13 18:31:18 +00:00
David Schleef
a7af574186 gst/multifile/Makefile.am: Let's not depend on a file that doesn't exist.
Original commit message from CVS:
* gst/multifile/Makefile.am:
Let's not depend on a file that doesn't exist.
2006-11-10 19:18:33 +00:00
David Schleef
1126a8863d Revive multifile[src|sink].
Original commit message from CVS:
* configure.ac:
* gst/multifile/Makefile.am:
* gst/multifile/gstmultifile.c:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/multifile.vproj:
Revive multifile[src|sink].
2006-11-10 18:51:10 +00:00
Stefan Kost
0a345cbb28 gst/flx/gstflxdec.c: fix categorisation, make short desc more explicit, remove unused code
Original commit message from CVS:
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
fix categorisation, make short desc more explicit, remove unused code
Fixes #372021
2006-11-08 12:04:03 +00:00
Christian Schaller
8707ff4347 gst/rtp/: Fix element descriptions.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtph264depay.c:
* gst/rtp/gstrtpmp2tdepay.c:
* gst/rtp/gstrtpmp4gdepay.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/rtp/gstrtpsv3vdepay.c:
Fix element descriptions.
2006-11-08 01:30:39 +00:00
Christian Schaller
5109e0d6cb gst/rtp/: Fix description.
Original commit message from CVS:
* gst/rtp/gstrtpvorbisdepay.c:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer):
Fix description.
Small cleanup in the payloader.
2006-11-08 01:29:51 +00:00
Christian Schaller
2ed1cb8ac2 gst/rtp/: Add theora pay/depayloaders.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
(gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
(gst_rtp_theora_depay_finalize),
(gst_rtp_theora_depay_parse_configuration),
(gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_switch_codebook),
(gst_rtp_theora_depay_process),
(gst_rtp_theora_depay_set_property),
(gst_rtp_theora_depay_get_property),
(gst_rtp_theora_depay_change_state),
(gst_rtp_theora_depay_plugin_init):
* gst/rtp/gstrtptheoradepay.h:
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
(gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
(gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
(gst_rtp_theora_pay_init_packet),
(gst_rtp_theora_pay_flush_packet),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer),
(gst_rtp_theora_pay_plugin_init):
* gst/rtp/gstrtptheorapay.h:
Add theora pay/depayloaders.
2006-11-08 01:28:00 +00:00
Christian Schaller
e0e75f715e gst/rtp/Makefile.am: We depend on gsttag to generate the vorbis comments.
Original commit message from CVS:
* gst/rtp/Makefile.am:
We depend on gsttag to generate the vorbis comments.
* gst/rtp/gstrtpvorbisdepay.c:
(gst_rtp_vorbis_depay_parse_configuration),
(gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_switch_codebook),
(gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbisdepay.h:
Parse configuration string in the depayloader.
Implement selecting and switching to a new codebook.
Receiving vorbis over RTP now works.
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
(gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_finish_headers),
(gst_rtp_vorbis_pay_handle_buffer):
* gst/rtp/gstrtpvorbispay.h:
Set timestamps on outgoing buffers and RTP packets.
Fix configuration string, prepend number of Packet headers.
Fix encoding of ident string.
Add delivery-method to caps.
Streaming vorbis over RTP now works.
2006-11-07 01:43:06 +00:00
Christian Schaller
53ddac062d gst/rtp/gstrtpvorbispay.*: Generate a valid configuration string in the caps based on the vorbis headers.
Original commit message from CVS:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
(gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
(gst_rtp_vorbis_pay_handle_buffer):
* gst/rtp/gstrtpvorbispay.h:
Generate a valid configuration string in the caps based on the
vorbis headers.
2006-11-06 20:52:10 +00:00
Tim-Philipp Müller
93c5c6fcba gst/id3demux/id3v2frames.c: We require a -base more recent than 0.10.9, so it's safe to use
Original commit message from CVS:
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
We require a -base more recent than 0.10.9, so it's safe to use
GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
Use _newsegment_full() now that we depend on a recent enough core.
* gst/wavparse/gstwavparse.c:
Remove cruft that we don't need any longer now that we depend on
a recent enough -base.
2006-11-01 13:59:49 +00:00
Wim Taymans
eead7e23b7 gst/rtp/: Fix and activate ILBC pay and depayloaders. Fixes #368162.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
(gst_rtpilbcpay_setcaps):
Fix and activate ILBC pay and depayloaders. Fixes #368162.
2006-10-31 10:52:31 +00:00
Wim Taymans
8ac0c0e09a gst/qtdemux/qtdemux.c: Handle unbounded length streams a bit better. Fixes #367696.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(gst_qtdemux_handle_src_query), (qtdemux_parse_tree),
(qtdemux_parse_trak):
Handle unbounded length streams a bit better. Fixes #367696.
2006-10-31 10:31:18 +00:00
Wim Taymans
92172d85bb gst/avi/gstavidemux.c: Use higher precision scale function.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
Use higher precision scale function.
2006-10-31 09:29:36 +00:00
Michal Benes
737df8258b gst/matroska/matroska-demux.c: Fix several issues with encoded/compressed/encrypted/signed tracks; also, remove super...
Original commit message from CVS:
Patch by: Michal Benes  <michal dot benes at itonis tv>
* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
(gst_matroska_demux_read_track_encodings),
(gst_matroska_decode_buffer):
Fix several issues with encoded/compressed/encrypted/signed tracks;
also, remove superfluous newline characters from some debug
statements. (#366155)
2006-10-30 16:18:18 +00:00
Mark Nauwelaerts
ccce64b293 gst/videomixer/videomixer.c: Fix videomixer so that it can handle any combination of framerates.
Original commit message from CVS:
Patch by: Mark Nauwelaerts  <manauw at skynet be>
* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
Fix videomixer so that it can handle any combination of framerates.
Fixes #367221.
2006-10-30 08:17:08 +00:00