Sebastian Dröge [Sat, 9 Jun 2007 15:33:32 +0000 (15:33 +0000)]
ext/wavpack/: Improve discont handling by checking if the next Wavpack block has the expected, following block index.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_push_buffer):
* ext/wavpack/gstwavpackparse.h:
Improve discont handling by checking if the next Wavpack block has
the expected, following block index.
Thomas Vander Stichele [Fri, 8 Jun 2007 20:23:07 +0000 (20:23 +0000)]
moap ignore
Original commit message from CVS:
moap ignore
Thomas Vander Stichele [Fri, 8 Jun 2007 20:20:56 +0000 (20:20 +0000)]
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.
Thomas Vander Stichele [Fri, 8 Jun 2007 20:19:55 +0000 (20:19 +0000)]
move wavpack plugin. See #352605.
Original commit message from CVS:
* configure.ac:
* 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/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.signals:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-gconfelements.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* ext/Makefile.am:
* tests/check/Makefile.am:
move wavpack plugin. See #352605.
Thomas Vander Stichele [Fri, 8 Jun 2007 19:45:43 +0000 (19:45 +0000)]
the alphabet tripping up people since 10929BC
Original commit message from CVS:
the alphabet
tripping up people since 10929BC
Jan Schmidt [Fri, 8 Jun 2007 17:37:02 +0000 (17:37 +0000)]
Add DirectDraw & DirectSound plugins to the build and docs.
Original commit message from CVS:
* configure.ac:
* 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:
* sys/Makefile.am:
* win32/MANIFEST:
Add DirectDraw & DirectSound plugins to the build and docs.
Jan Schmidt [Fri, 8 Jun 2007 16:31:15 +0000 (16:31 +0000)]
Rename the keep-aspect-ratio property to force-aspect-ratio to make it consistent with xvimagesink and ximagesink.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_class_init):
Rename the keep-aspect-ratio property to force-aspect-ratio to make
it consistent with xvimagesink and ximagesink.
Tim-Philipp Müller [Fri, 8 Jun 2007 10:43:26 +0000 (10:43 +0000)]
ext/: When operating in pull mode, error out correct on not-linked.
Original commit message from CVS:
* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
When operating in pull mode, error out correct on not-linked.
Tim-Philipp Müller [Fri, 8 Jun 2007 08:12:43 +0000 (08:12 +0000)]
tests/icles/videocrop-test.c: Default to xvimagesink instead of autovideosink while autovideosink/ghostpads/whatever ...
Original commit message from CVS:
* tests/icles/videocrop-test.c: (main):
Default to xvimagesink instead of autovideosink while
autovideosink/ghostpads/whatever don't handle the way we use it in
the way we expect it to.
Andy Wingo [Wed, 6 Jun 2007 10:19:17 +0000 (10:19 +0000)]
sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
Original commit message from CVS:
2007-06-06 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
format and size if the ioctls are defined; should fix compilation
on Linux < 2.16.19.
Tim-Philipp Müller [Wed, 6 Jun 2007 08:53:12 +0000 (08:53 +0000)]
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.
Tim-Philipp Müller [Tue, 5 Jun 2007 16:32:19 +0000 (16:32 +0000)]
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.
Andy Wingo [Tue, 5 Jun 2007 14:17:25 +0000 (14:17 +0000)]
sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add some useless comments.
Original commit message from CVS:
2007-06-05 Andy Wingo <wingo@pobox.com>
* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
some useless comments.
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
frames before calling STREAMON, that might leave them in a state
where they can't be dequeued if we go back to NULL without calling
STREAMON, according to the docs.
(gst_v4l2src_capture_start): Enqueue buffers here instead, right
before we call STREAMON.
(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
failures. (For me this code hung.) The pool refcounting is still
crack; added a note to that effect.
Wim Taymans [Tue, 5 Jun 2007 09:11:41 +0000 (09:11 +0000)]
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.
Sebastian Dröge [Sun, 3 Jun 2007 11:21:44 +0000 (11:21 +0000)]
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.
Wim Taymans [Fri, 1 Jun 2007 13:52:17 +0000 (13:52 +0000)]
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.
Peter Kjellerstedt [Fri, 1 Jun 2007 13:07:11 +0000 (13:07 +0000)]
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.
Daniel Charles [Fri, 1 Jun 2007 11:16:17 +0000 (11:16 +0000)]
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.
Wim Taymans [Thu, 31 May 2007 15:57:07 +0000 (15:57 +0000)]
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.
Andy Wingo [Wed, 30 May 2007 14:57:44 +0000 (14:57 +0000)]
sys/v4l2/gstv4l2object.*: Revert some unintended changes.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
unintended changes.
Andy Wingo [Wed, 30 May 2007 14:40:53 +0000 (14:40 +0000)]
sys/v4l2/v4l2src_calls.*: Store the format list in the order that the driver gives it to us.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.h:
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
the format list in the order that the driver gives it to us.
(gst_v4l2src_probe_caps_for_format_and_size)
(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
based on the capabilities of the device.
(gst_v4l2src_grab_frame): Update for object variable renaming.
(gst_v4l2src_set_capture): Update to be strict in its parameters,
as in the set_caps below.
(gst_v4l2src_capture_init): Update for object variable renaming,
and reflow.
(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
(gst_v4l2src_capture_deinit): Update for object variable renaming.
(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
(gst_v4l2src_get_fps): Remove; these functions don't have much
meaning outside of an atomic set_caps method.
(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
known.
* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
call to update_fps; not sure about this change.
(gst_v4l2_tuner_set_norm): Work around the fact that for the
moment we don't have an update_fps_func.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
structures in the object, just store what we need. Do store the
probed caps of the device. Don't store the current frame rate.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
update_fps_function, for now. Update for new object variable
naming.
(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
new object variable naming.
(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
(gst_v4l2src_get_caps): Rework to probe the device for supported
frame sizes and frame rates.
(gst_v4l2src_set_caps): Rework to be strict in the given
parameters: if someone asks us to have a certain size and rate,
that is what we configure.
(gst_v4l2src_get_read): Update for object variable naming. Don't
leak buffers on short reads.
(gst_v4l2src_get_mmap): Update for object variable naming, and add
comments.
(gst_v4l2src_create): Update for object variable naming.
Tim-Philipp Müller [Wed, 30 May 2007 14:38:59 +0000 (14:38 +0000)]
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.
Tim-Philipp Müller [Wed, 30 May 2007 12:46:32 +0000 (12:46 +0000)]
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.
Wim Taymans [Mon, 28 May 2007 16:39:09 +0000 (16:39 +0000)]
gst/rtsp/rtsptransport.c: Use renamed RTP bin.
Original commit message from CVS:
* gst/rtsp/rtsptransport.c:
Use renamed RTP bin.
Dejan Sakelšak [Mon, 28 May 2007 15:01:33 +0000 (15:01 +0000)]
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.
Wim Taymans [Sat, 26 May 2007 15:25:18 +0000 (15:25 +0000)]
ext/speex/gstspeexdec.c: Use different variables for nested for loops so that the outer loop functions properly and s...
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
Use different variables for nested for loops so that the outer loop
functions properly and speex files with multiple frames per buffer work
properly.
Fixes #441408.
Tim-Philipp Müller [Fri, 25 May 2007 20:51:36 +0000 (20:51 +0000)]
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.
Tim-Philipp Müller [Fri, 25 May 2007 20:33:10 +0000 (20:33 +0000)]
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.
Tim-Philipp Müller [Fri, 25 May 2007 16:02:51 +0000 (16:02 +0000)]
ext/flac/gstflacenc.*: Collect headers, add "streamheader" field to output caps and set
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_init),
(notgst_value_array_append_buffer),
(gst_flac_enc_process_stream_headers),
(gst_flac_enc_write_callback), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Collect headers, add "streamheader" field to output caps and set
BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
produces output according to the official FLAC-to-Ogg mapping
instead of completely broken files. Fixes #426044.
Jan Schmidt [Fri, 25 May 2007 10:44:12 +0000 (10:44 +0000)]
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.
Jan Schmidt [Fri, 25 May 2007 10:23:49 +0000 (10:23 +0000)]
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
Tim-Philipp Müller [Thu, 24 May 2007 17:00:21 +0000 (17:00 +0000)]
ext/flac/gstflacenc.c: Don't crash in chain function if setcaps hasn't been called.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
(gst_flac_enc_chain):
Don't crash in chain function if setcaps hasn't been called.
Vincent Torri [Thu, 24 May 2007 08:35:23 +0000 (08:35 +0000)]
sys/directdraw/gstdirectdrawsink.*: Fix more warnings when compiling with MingW (#439914).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_show_frame),
(gst_directdraw_sink_check_primary_surface),
(gst_directdraw_sink_check_offscreen_surface),
(EnumModesCallback2), (gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
* sys/directdraw/gstdirectdrawsink.h:
Fix more warnings when compiling with MingW (#439914).
Wim Taymans [Thu, 24 May 2007 08:14:00 +0000 (08:14 +0000)]
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.
Peter Kjellerstedt [Thu, 24 May 2007 08:10:42 +0000 (08:10 +0000)]
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.
Sébastien Moutte [Wed, 23 May 2007 22:44:12 +0000 (22:44 +0000)]
docs/plugins/gst-plugins-bad-plugins.args: Remove directsoundsink property doc as this sink use the mixer interface now.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
Stefan Kost [Tue, 22 May 2007 11:14:13 +0000 (11:14 +0000)]
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.
Edward Hervey [Tue, 22 May 2007 11:03:30 +0000 (11:03 +0000)]
docs/plugins/Makefile.am: Also look for .m (objectivec) files.
Original commit message from CVS:
* docs/plugins/Makefile.am:
Also look for .m (objectivec) files.
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* sys/osxvideo/osxvideosink.m:
Add documentation for element and properties.
Stefan Kost [Mon, 21 May 2007 14:01:16 +0000 (14:01 +0000)]
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.
Stefan Kost [Mon, 21 May 2007 12:43:37 +0000 (12:43 +0000)]
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.
Sebastian Dröge [Mon, 21 May 2007 11:37:16 +0000 (11:37 +0000)]
ext/wavpack/gstwavpackenc.c: Specify and use properties as unsigned int that are an unsigned int.
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
Specify and use properties as unsigned int that are an unsigned int.
Sebastian Dröge [Mon, 21 May 2007 11:17:21 +0000 (11:17 +0000)]
ext/wavpack/gstwavpackenc.*: Fixup docs, make the bitrate property an int as it should be and allow to set the differ...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property):
* ext/wavpack/gstwavpackenc.h:
Fixup docs, make the bitrate property an int as it should be and
allow to set the different extra processing modes instead of only
allowing none and the default one.
Wim Taymans [Mon, 21 May 2007 10:07:05 +0000 (10:07 +0000)]
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.
Wim Taymans [Mon, 21 May 2007 10:03:42 +0000 (10:03 +0000)]
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.
René Stadler [Mon, 21 May 2007 09:32:26 +0000 (09:32 +0000)]
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.
Wim Taymans [Mon, 21 May 2007 09:29:30 +0000 (09:29 +0000)]
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.
Antoine Tremblay [Mon, 21 May 2007 08:57:18 +0000 (08:57 +0000)]
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.
Sebastian Dröge [Sun, 20 May 2007 21:31:58 +0000 (21:31 +0000)]
ext/wavpack/gstwavpackenc.c: Add missing audioconverts in the example pipelines of wavpackenc. As the wavpack stuff n...
Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c:
Add missing audioconverts in the example pipelines of wavpackenc. As
the wavpack stuff now needs input with 32 bit width (and random depth)
this is needed now. The example pipelines for the parser and decoder
are still fine.
Tim-Philipp Müller [Sun, 20 May 2007 14:59:46 +0000 (14:59 +0000)]
sys/directdraw/gstdirectdrawsink.c: Bunch of small fixes: remove static function that doesn't exist; declare another ...
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c: (gst_ddrawsurface_finalize),
(gst_directdraw_sink_buffer_alloc),
(gst_directdraw_sink_get_ddrawcaps),
(gst_directdraw_sink_surface_create):
Bunch of small fixes: remove static function that doesn't exist;
declare another one that does; printf format fix; use right macro
when specifying debug category; remove a bunch of unused variables;
#if 0 out an unused chunk of code (partially fixes #439914).
Tim-Philipp Müller [Sun, 20 May 2007 14:14:49 +0000 (14:14 +0000)]
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).
Tim-Philipp Müller [Sun, 20 May 2007 14:05:42 +0000 (14:05 +0000)]
gst/rtsp/gstrtspsrc.c: Printf format fix.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Printf format fix.
René Stadler [Sat, 19 May 2007 10:01:45 +0000 (10:01 +0000)]
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).
Wim Taymans [Fri, 18 May 2007 13:27:39 +0000 (13:27 +0000)]
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.
Wim Taymans [Fri, 18 May 2007 11:39:12 +0000 (11:39 +0000)]
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.
Wim Taymans [Fri, 18 May 2007 10:36:12 +0000 (10:36 +0000)]
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.
Wim Taymans [Thu, 17 May 2007 14:56:39 +0000 (14:56 +0000)]
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.
Edward Hervey [Thu, 17 May 2007 10:59:00 +0000 (10:59 +0000)]
sys/osxvideo/osxvideosink.*: Remove the event-loop-in-separate-thread modifications, because MacOSX is $#@(*%$# ! For...
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Remove the event-loop-in-separate-thread modifications, because MacOSX
is $#@(*%$# ! For those wondering, the event handling needs to be done
in the main thread after all..
Edward Hervey [Thu, 17 May 2007 09:41:48 +0000 (09:41 +0000)]
sys/osxvideo/osxvideosink.*: Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
Original commit message from CVS:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix a stupid #if vs #ifdef bug. Should use the proper colorspace now.
Use a separate thread/task for the cocoa event_loop, else it wouldn't
stop.
Edward Hervey [Wed, 16 May 2007 16:50:23 +0000 (16:50 +0000)]
ext/libpng/gstpngdec.c: Fix build on macosx.
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
Fix build on macosx.
Sebastian Dröge [Wed, 16 May 2007 16:30:03 +0000 (16:30 +0000)]
ext/raw1394/gstdv1394src.c: Replace direct comparison of a string with the string literal "" with a comparison of the...
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
Replace direct comparison of a string with the string literal "" with
a comparison of the first character with '\0'. Fixes #438926.
Tim-Philipp Müller [Tue, 15 May 2007 17:22:58 +0000 (17:22 +0000)]
Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save and restore the various flags in the directdraw/dir...
Original commit message from CVS:
* configure.ac:
* sys/directdraw/Makefile.am:
* sys/directsound/Makefile.am:
Add DIRECTDRAW_CFLAGS and DIRECTSOUND_CFLAGS to Makefile.am; save
and restore the various flags in the directdraw/directsound
detection section. Apparently improves cross-compiling for win32
with mingw32 under some circumstances (#437539).
Stefan Kost [Tue, 15 May 2007 11:18:33 +0000 (11:18 +0000)]
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.
Stefan Kost [Tue, 15 May 2007 06:41:58 +0000 (06:41 +0000)]
gst/debug/breakmydata.c: Ooops, no // comments please.
Original commit message from CVS:
* gst/debug/breakmydata.c:
Ooops, no // comments please.
Stefan Kost [Tue, 15 May 2007 06:34:48 +0000 (06:34 +0000)]
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.
David Schleef [Tue, 15 May 2007 02:56:23 +0000 (02:56 +0000)]
sys/v4l2/gstv4l2src.c: Add support for Bayer images as video/x-raw-bayer. Fixes #314160.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: Add support for Bayer images as
video/x-raw-bayer. Fixes #314160.
Wim Taymans [Mon, 14 May 2007 17:10:12 +0000 (17:10 +0000)]
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.
Wim Taymans [Mon, 14 May 2007 16:19:58 +0000 (16:19 +0000)]
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.
Wim Taymans [Mon, 14 May 2007 14:47:26 +0000 (14:47 +0000)]
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.
Wim Taymans [Mon, 14 May 2007 11:11:42 +0000 (11:11 +0000)]
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.
Wim Taymans [Mon, 14 May 2007 09:01:05 +0000 (09:01 +0000)]
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.
David Schleef [Sun, 13 May 2007 19:57:45 +0000 (19:57 +0000)]
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>.
Sébastien Moutte [Sun, 13 May 2007 15:47:13 +0000 (15:47 +0000)]
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).
Peter Kjellerstedt [Sat, 12 May 2007 16:37:50 +0000 (16:37 +0000)]
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.
Peter Kjellerstedt [Sat, 12 May 2007 16:27:51 +0000 (16:27 +0000)]
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.
Peter Kjellerstedt [Sat, 12 May 2007 16:26:06 +0000 (16:26 +0000)]
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.
Eric Anholt [Fri, 11 May 2007 16:11:04 +0000 (16:11 +0000)]
sys/ximage/gstximagesrc.c (gst_ximage_src_open_display, gst_ximage_src_ximage_get):
Original commit message from CVS:
Patch by: Eric Anholt
* sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
gst_ximage_src_ximage_get):
Use union of all damage between frames to make it faster.
Fixes bug #342463.
Also fix crasher when cursor is at bottom right of window.
Tim-Philipp Müller [Fri, 11 May 2007 16:01:45 +0000 (16:01 +0000)]
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.
Wim Taymans [Fri, 11 May 2007 15:09:39 +0000 (15:09 +0000)]
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.
Wim Taymans [Fri, 11 May 2007 15:04:38 +0000 (15:04 +0000)]
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.
Wim Taymans [Fri, 11 May 2007 11:24:13 +0000 (11:24 +0000)]
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.
Zaheer Abbas Merali [Fri, 11 May 2007 10:31:27 +0000 (10:31 +0000)]
sys/ximage/gstximagesrc.c (gst_ximage_src_start, gst_ximage_src_ximage_get):
Original commit message from CVS:
* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
gst_ximage_src_ximage_get):
* sys/ximage/gstximagesrc.h (last_ximage):
When using Damage actually keep the last frame, and not assume
that the buffer we get already has the last frame on it.
Copy the cursor over if we specify a non-zero start x and
start y.
Wim Taymans [Fri, 11 May 2007 09:12:55 +0000 (09:12 +0000)]
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.
Stefan Kost [Thu, 10 May 2007 14:02:07 +0000 (14:02 +0000)]
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>).
David Schleef [Thu, 10 May 2007 01:21:19 +0000 (01:21 +0000)]
gst/level/gstlevel.c: Revert last change.
Original commit message from CVS:
* gst/level/gstlevel.c:
Revert last change.
Sébastien Moutte [Wed, 9 May 2007 21:30:53 +0000 (21:30 +0000)]
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.
Wim Taymans [Wed, 9 May 2007 11:23:39 +0000 (11:23 +0000)]
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.
Tim-Philipp Müller [Tue, 8 May 2007 15:49:01 +0000 (15:49 +0000)]
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).
Sebastian Dröge [Sun, 6 May 2007 21:32:40 +0000 (21:32 +0000)]
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.
Wim Taymans [Fri, 4 May 2007 15:17:14 +0000 (15:17 +0000)]
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.
Wim Taymans [Fri, 4 May 2007 13:04:31 +0000 (13:04 +0000)]
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.
Wim Taymans [Fri, 4 May 2007 12:31:32 +0000 (12:31 +0000)]
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.
Wim Taymans [Thu, 3 May 2007 15:55:06 +0000 (15:55 +0000)]
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.
Wim Taymans [Thu, 3 May 2007 14:39:09 +0000 (14:39 +0000)]
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.
Wim Taymans [Thu, 3 May 2007 13:48:54 +0000 (13:48 +0000)]
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.
Wim Taymans [Wed, 2 May 2007 19:32:58 +0000 (19:32 +0000)]
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.
Sebastian Dröge [Wed, 2 May 2007 18:31:16 +0000 (18:31 +0000)]
ext/wavpack/gstwavpack.c: Call bindtextdomain() to get localized strings.
Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
Call bindtextdomain() to get localized strings.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_chain):
* ext/wavpack/gstwavpackparse.h:
Handle DISCONT buffers by correctly setting the DISCONT flag
on outgoing buffers when necessary.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_handle_seek_event)
Send newsegment from the streaming thread.
Wim Taymans [Wed, 2 May 2007 18:25:09 +0000 (18:25 +0000)]
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.
Sebastian Dröge [Wed, 2 May 2007 18:01:52 +0000 (18:01 +0000)]
ext/flac/gstflac.c: Call bindtextdomain() to get localized strings.
Original commit message from CVS:
* ext/flac/gstflac.c: (plugin_init):
Call bindtextdomain() to get localized strings.
Wim Taymans [Wed, 2 May 2007 17:19:36 +0000 (17:19 +0000)]
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.
Sebastian Dröge [Wed, 2 May 2007 17:08:09 +0000 (17:08 +0000)]
ext/flac/gstflacdec.c: Correctly post an error on the bus if something went wrong in the loop function. This fixes a ...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
Correctly post an error on the bus if something went wrong in the loop
function. This fixes a few cases where the task was paused and nothing
happened anymore.
projects / platform / upstream / gst-plugins-good.git / log