jp.liu [Wed, 14 Feb 2007 10:09:12 +0000 (10:09 +0000)]
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.
Wim Taymans [Wed, 14 Feb 2007 09:55:47 +0000 (09:55 +0000)]
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.
Jan Schmidt [Tue, 13 Feb 2007 16:01:29 +0000 (16:01 +0000)]
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.
Stefan Kost [Tue, 13 Feb 2007 11:57:18 +0000 (11:57 +0000)]
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
Stefan Kost [Tue, 13 Feb 2007 09:46:26 +0000 (09:46 +0000)]
Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
Tim-Philipp Müller [Mon, 12 Feb 2007 23:35:16 +0000 (23:35 +0000)]
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).
Jonathan Matthew [Mon, 12 Feb 2007 23:27:31 +0000 (23:27 +0000)]
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.
Stefan Kost [Mon, 12 Feb 2007 15:29:44 +0000 (15:29 +0000)]
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.
Stefan Kost [Mon, 12 Feb 2007 12:57:22 +0000 (12:57 +0000)]
sys/v4l2/: More FIXME comments and messaging changes.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
(gst_v4l2src_get_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init):
More FIXME comments and messaging changes.
Stefan Kost [Mon, 12 Feb 2007 12:43:00 +0000 (12:43 +0000)]
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.
Tim-Philipp Müller [Mon, 12 Feb 2007 10:29:57 +0000 (10:29 +0000)]
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.
Sébastien Moutte [Sun, 11 Feb 2007 15:26:49 +0000 (15:26 +0000)]
Makefile.am: Add win32 MANIFEST
Original commit message from CVS:
* Makefile.am:
Add win32 MANIFEST
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Clear unused code and add comments.
Remove yuv from template caps, it only supports RGB
actually.
Implement XOverlay interface and remove window and fullscreen
properties.
Add debug logs.
Test for blit capabilities to return only the current colorspace if
the hardware can't blit for one colorspace to another.
* sys/directsound/gstdirectsoundsink.c:
Add some debugs.
* win32/MANIFEST:
Add VS7 project files and solution.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectdraw.dsp:
* win32/vs6/libgstdirectsound.dsp:
* win32/vs6/libgstqtdemux.dsp:
Update project files.
Sébastien Moutte [Sun, 11 Feb 2007 12:57:47 +0000 (12:57 +0000)]
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.
Stefan Kost [Sun, 11 Feb 2007 10:53:21 +0000 (10:53 +0000)]
configure.ac: Activate monoscope when building with --enable-experimental. Fix
Original commit message from CVS:
* configure.ac:
Activate monoscope when building with --enable-experimental. Fix
--enable-external configure switch description.
* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
Help gst-indent.
Tim-Philipp Müller [Fri, 9 Feb 2007 09:24:58 +0000 (09:24 +0000)]
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.
Tim-Philipp Müller [Thu, 8 Feb 2007 11:09:15 +0000 (11:09 +0000)]
gst/debug/progressreport.c: Some more docs.
Original commit message from CVS:
* gst/debug/progressreport.c:
Some more docs.
Tim-Philipp Müller [Wed, 7 Feb 2007 21:09:45 +0000 (21:09 +0000)]
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.
Tim-Philipp Müller [Wed, 7 Feb 2007 20:39:16 +0000 (20:39 +0000)]
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.
Tim-Philipp Müller [Wed, 7 Feb 2007 13:08:34 +0000 (13:08 +0000)]
ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ALSA capabilities a bit better.
Original commit message from CVS:
* ext/hal/hal.c: (gst_hal_get_string):
* ext/hal/hal.h:
Some small cleanups; deal with errors when parsing the HAL ALSA
capabilities a bit better.
Tim-Philipp Müller [Tue, 6 Feb 2007 16:29:30 +0000 (16:29 +0000)]
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.
Tim-Philipp Müller [Tue, 6 Feb 2007 16:24:57 +0000 (16:24 +0000)]
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.
Tim-Philipp Müller [Tue, 6 Feb 2007 15:56:14 +0000 (15:56 +0000)]
ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle (ie. one dependent on the profile select...
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
(gst_gconf_render_bin_from_key),
(gst_gconf_get_default_audio_sink):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
(do_toggle_element), (gst_gconf_audio_sink_set_property),
(gst_gconf_audio_sink_get_property):
In gconfaudiosink, get the right key as the old key in do_toggle
(ie. one dependent on the profile selected). Log some more stuff so
we can see what's actually going on.
Sebastian Dröge [Tue, 6 Feb 2007 11:16:49 +0000 (11:16 +0000)]
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
Tim-Philipp Müller [Sat, 3 Feb 2007 23:35:26 +0000 (23:35 +0000)]
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.
Tim-Philipp Müller [Fri, 2 Feb 2007 18:36:28 +0000 (18:36 +0000)]
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.
James Doc Livingston [Fri, 2 Feb 2007 17:39:21 +0000 (17:39 +0000)]
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).
Sebastian Dröge [Wed, 31 Jan 2007 08:32:59 +0000 (08:32 +0000)]
ext/wavpack/gstwavpackparse.c: Fix a off by one that leads to the duration reported as one sample less than it is
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_create_src_pad):
Fix a off by one that leads to the duration reported as one
sample less than it is
Edward Hervey [Tue, 30 Jan 2007 17:19:33 +0000 (17:19 +0000)]
configure.ac: Check for an Objective C compiler
Original commit message from CVS:
* configure.ac:
Check for an Objective C compiler
* sys/Makefile.am:
* sys/osxvideo/Makefile.am:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Port of osxvideo plugin to 0.10. Do NOT consider 100% stable !
Fixes #402470
Wim Taymans [Mon, 29 Jan 2007 10:59:48 +0000 (10:59 +0000)]
tests/check/elements/.cvsignore: Some more ignores.
Original commit message from CVS:
* tests/check/elements/.cvsignore:
Some more ignores.
Tim-Philipp Müller [Sun, 28 Jan 2007 18:28:33 +0000 (18:28 +0000)]
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.
Tim-Philipp Müller [Sat, 27 Jan 2007 16:08:15 +0000 (16:08 +0000)]
tests/icles/videocrop-test.c: Catch errors while the test is running.
Original commit message from CVS:
* tests/icles/videocrop-test.c: (test_with_caps):
Catch errors while the test is running.
charles [Fri, 26 Jan 2007 12:21:41 +0000 (12:21 +0000)]
ext/shout2/gstshout2.*: Properly handle tags in shout2send. Fixes #399825.
Original commit message from CVS:
Patch by: charles <charlesg3 at gmail dot com>
* ext/shout2/gstshout2.c: (gst_shout2send_init),
(set_shout_metadata), (gst_shout2send_event):
* ext/shout2/gstshout2.h:
Properly handle tags in shout2send. Fixes #399825.
Sebastian Dröge [Thu, 25 Jan 2007 23:27:59 +0000 (23:27 +0000)]
ext/wavpack/gstwavpackparse.c: Fix the SEEKING query. We can seek if we are in pull mode, not the other way around. A...
Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_src_query):
Fix the SEEKING query. We can seek if we are in pull mode, not the
other way around. Also set the correct format in the seeking query and
handle the case where the headers are not read yet and we can't say
anything about our seeking capabilities.
Sebastian Dröge [Thu, 25 Jan 2007 21:55:49 +0000 (21:55 +0000)]
ext/wavpack/: Fix spelling in 2 places: It's called Wavpack, not WavePack.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
Fix spelling in 2 places: It's called Wavpack, not WavePack.
Wim Taymans [Thu, 25 Jan 2007 14:40:15 +0000 (14:40 +0000)]
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.
Wim Taymans [Thu, 25 Jan 2007 14:22:53 +0000 (14:22 +0000)]
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.
Edward Hervey [Thu, 25 Jan 2007 12:05:11 +0000 (12:05 +0000)]
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
Wim Taymans [Thu, 25 Jan 2007 11:02:01 +0000 (11:02 +0000)]
configure.ac: Bump required -core/-base to CVS
Original commit message from CVS:
* configure.ac:
Bump required -core/-base to CVS
Wim Taymans [Thu, 25 Jan 2007 10:54:19 +0000 (10:54 +0000)]
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.
Edward Hervey [Thu, 25 Jan 2007 10:36:35 +0000 (10:36 +0000)]
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.
Wim Taymans [Wed, 24 Jan 2007 18:20:14 +0000 (18:20 +0000)]
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.
Wim Taymans [Wed, 24 Jan 2007 16:25:55 +0000 (16:25 +0000)]
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.
Wim Taymans [Wed, 24 Jan 2007 15:18:34 +0000 (15:18 +0000)]
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.
Sebastian Dröge [Wed, 24 Jan 2007 12:41:03 +0000 (12:41 +0000)]
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
Wim Taymans [Wed, 24 Jan 2007 12:26:41 +0000 (12:26 +0000)]
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.
Wim Taymans [Wed, 24 Jan 2007 12:22:51 +0000 (12:22 +0000)]
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.
Wim Taymans [Wed, 24 Jan 2007 11:29:00 +0000 (11:29 +0000)]
gst/qtdemux/gstrtpxqtdepay.c: Fix caps on the depayloader.
Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c:
Fix caps on the depayloader.
Sebastian Dröge [Tue, 23 Jan 2007 18:16:09 +0000 (18:16 +0000)]
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.
Wim Taymans [Tue, 23 Jan 2007 17:36:32 +0000 (17:36 +0000)]
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
Wim Taymans [Tue, 23 Jan 2007 17:27:39 +0000 (17:27 +0000)]
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.
Tim-Philipp Müller [Mon, 22 Jan 2007 13:06:43 +0000 (13:06 +0000)]
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.
Tim-Philipp Müller [Fri, 19 Jan 2007 13:06:07 +0000 (13:06 +0000)]
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.
Edward Hervey [Fri, 19 Jan 2007 10:35:13 +0000 (10:35 +0000)]
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.
Edward Hervey [Fri, 19 Jan 2007 09:48:47 +0000 (09:48 +0000)]
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
Tim-Philipp Müller [Thu, 18 Jan 2007 18:37:39 +0000 (18:37 +0000)]
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.
Tim-Philipp Müller [Thu, 18 Jan 2007 11:29:17 +0000 (11:29 +0000)]
docs/plugins/: Remove ladspa from docs; add hierarchy info for GstAudioPanorama; fix integer properties with -1 as mi...
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
fix integer properties with -1 as minimum value.
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-annodex.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cdio.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-efence.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-esdsink.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gconfelements.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-halelements.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-videobalance.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videoflip.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
Update to CVS.
Stefan Kost [Thu, 18 Jan 2007 11:23:36 +0000 (11:23 +0000)]
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)
Tim-Philipp Müller [Thu, 18 Jan 2007 10:33:50 +0000 (10:33 +0000)]
Remove bogus ChangeLog entry
Original commit message from CVS:
Remove bogus ChangeLog entry
Stefan Kost [Wed, 17 Jan 2007 14:30:50 +0000 (14:30 +0000)]
sys/v4l2/: Fix EIO handing when capturing. Add new property to specify the number of buffers to enque (and remove the...
Original commit message from CVS:
* sys/v4l2/gstv4l2object.c:
(gst_v4l2_object_install_properties_helper),
(gst_v4l2_object_set_property_helper),
(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_init), (gst_v4l2src_set_property),
(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
(gst_v4l2src_capture_deinit):
Fix EIO handing when capturing. Add new property to specify the number of
buffers to enque (and remove the borked num-buffers usage).
Sebastian Dröge [Tue, 16 Jan 2007 08:29:11 +0000 (08:29 +0000)]
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.
Mark Nauwelaerts [Sun, 14 Jan 2007 17:55:33 +0000 (17:55 +0000)]
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.
Andy Wingo [Sat, 13 Jan 2007 19:12:32 +0000 (19:12 +0000)]
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.
Andy Wingo [Sat, 13 Jan 2007 18:01:41 +0000 (18:01 +0000)]
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.
Sebastian Dröge [Sat, 13 Jan 2007 15:52:18 +0000 (15:52 +0000)]
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.
Christian Schaller [Fri, 12 Jan 2007 18:28:13 +0000 (18:28 +0000)]
comment out LADSPA plugin for now
Original commit message from CVS:
comment out LADSPA plugin for now
Wim Taymans [Fri, 12 Jan 2007 17:16:51 +0000 (17:16 +0000)]
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.
Tim-Philipp Müller [Fri, 12 Jan 2007 12:10:19 +0000 (12:10 +0000)]
gst/qtdemux/Makefile.am: Dist all new files.
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
Dist all new files.
Wim Taymans [Fri, 12 Jan 2007 10:27:25 +0000 (10:27 +0000)]
docs/plugins/: Activate docs for jack, sdl and qtdemux.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-qtdemux.xml:
Activate docs for jack, sdl and qtdemux.
Wim Taymans [Fri, 12 Jan 2007 10:22:16 +0000 (10:22 +0000)]
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.
Sebastian Dröge [Thu, 11 Jan 2007 19:51:04 +0000 (19:51 +0000)]
ext/wavpack/gstwavpackenc.c: Minor clean-up: use enum values instead of hardcoded constants (#395536).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
(gst_wavpack_enc_correction_mode_get_type),
(gst_wavpack_enc_joint_stereo_mode_get_type):
Minor clean-up: use enum values instead of hardcoded constants (#395536).
Tim-Philipp Müller [Thu, 11 Jan 2007 16:59:40 +0000 (16:59 +0000)]
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.
Jonas Holmberg [Thu, 11 Jan 2007 11:05:04 +0000 (11:05 +0000)]
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.
Lutz Mueller [Thu, 11 Jan 2007 09:30:59 +0000 (09:30 +0000)]
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.
Peter Kjellerstedt [Wed, 10 Jan 2007 15:19:48 +0000 (15:19 +0000)]
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.
Sebastian Dröge [Wed, 10 Jan 2007 09:47:43 +0000 (09:47 +0000)]
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).
Wim Taymans [Tue, 9 Jan 2007 12:25:26 +0000 (12:25 +0000)]
gst/avi/gstavidemux.c: Fix docs.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Fix docs.
Wim Taymans [Tue, 9 Jan 2007 12:23:48 +0000 (12:23 +0000)]
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.
Vincent Torri [Mon, 8 Jan 2007 12:45:10 +0000 (12:45 +0000)]
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.
Andy Wingo [Sun, 7 Jan 2007 22:03:54 +0000 (22:03 +0000)]
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.
Sébastien Moutte [Sun, 7 Jan 2007 10:44:12 +0000 (10:44 +0000)]
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).
Tim-Philipp Müller [Fri, 5 Jan 2007 18:32:03 +0000 (18:32 +0000)]
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).
Tim-Philipp Müller [Fri, 5 Jan 2007 17:23:04 +0000 (17:23 +0000)]
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).
Tim-Philipp Müller [Fri, 5 Jan 2007 16:07:12 +0000 (16:07 +0000)]
tests/icles/videocrop-test.c: Call g_thread_init() right at the beginning. Remove superfluous gst_init() - we've alre...
Original commit message from CVS:
* tests/icles/videocrop-test.c: (main):
Call g_thread_init() right at the beginning. Remove superfluous
gst_init() - we've already been inited via the GOption stuff.
Tim-Philipp Müller [Thu, 4 Jan 2007 11:02:29 +0000 (11:02 +0000)]
ext/esd/esdsink.c: Don't return bogus values when esd_get_delay() fails for some reason (#392189).
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_delay):
Don't return bogus values when esd_get_delay() fails for some
reason (#392189).
Vincent Torri [Thu, 4 Jan 2007 09:44:57 +0000 (09:44 +0000)]
Add directsoundsink to build and dist it, so it gets built when compiling with MingW on win32 and the required header...
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* configure.ac:
* sys/Makefile.am:
* sys/directsound/Makefile.am:
* sys/directsound/gstdirectsoundsink.c:
(gst_directsoundsink_reset):
Add directsoundsink to build and dist it, so it gets built when
compiling with MingW on win32 and the required headers and libraries
are available (fixes: #392638). Also simplify DirectDraw check a bit.
* tests/check/elements/.cvsignore:
Fix CVS ignore for neonhttpsrc test binary.
Vincent Torri [Wed, 3 Jan 2007 19:54:33 +0000 (19:54 +0000)]
Add directdrawsink to build and dist it, so it gets built when compiling with MingW on win32 and the required headers...
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry fr>
* configure.ac:
* sys/Makefile.am:
* sys/directdraw/Makefile.am:
Add directdrawsink to build and dist it, so it gets built when
compiling with MingW on win32 and the required headers and libraries
are available (fixes: #392313).
* sys/directdraw/gstdirectdrawsink.c:
(gst_directdrawsink_center_rect), (gst_directdrawsink_show_frame),
(gst_directdrawsink_setup_ddraw),
(gst_directdrawsink_surface_create):
Comment out some unused things and fix some printf format issues in
order to avoid warnings when buildling with MingW (#392313).
Jens Granseuer [Wed, 3 Jan 2007 16:41:10 +0000 (16:41 +0000)]
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.
Tim-Philipp Müller [Sat, 30 Dec 2006 12:44:01 +0000 (12:44 +0000)]
tests/check/elements/videocrop.c: When we can't create an element needed for the test, print a message detailing whic...
Original commit message from CVS:
* tests/check/elements/videocrop.c: (GST_START_TEST),
(videocrop_test_cropping_init_context):
When we can't create an element needed for the test, print a message
detailing which element it actually is that's missing (#390673).
Tim-Philipp Müller [Sun, 24 Dec 2006 11:36:31 +0000 (11:36 +0000)]
sys/ximage/gstximagesrc.c: Fix presumably copy'n'pasto for 16bpp depth.
Original commit message from CVS:
* sys/ximage/gstximagesrc.c: (composite_pixel):
Fix presumably copy'n'pasto for 16bpp depth.
Tim-Philipp Müller [Sun, 24 Dec 2006 11:24:59 +0000 (11:24 +0000)]
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.
Stefan Kost [Fri, 22 Dec 2006 10:15:24 +0000 (10:15 +0000)]
tests/check/elements/: reapply consistent pad (de)activation
Original commit message from CVS:
* tests/check/elements/mpeg2enc.c: (setup_mpeg2enc),
(cleanup_mpeg2enc):
* tests/check/elements/rganalysis.c: (cleanup_rganalysis):
* tests/check/elements/wavpackdec.c: (setup_wavpackdec),
(cleanup_wavpackdec):
* tests/check/elements/wavpackenc.c: (setup_wavpackenc),
(cleanup_wavpackenc):
* tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
reapply consistent pad (de)activation
Stefan Kost [Fri, 22 Dec 2006 10:15:23 +0000 (10:15 +0000)]
tests/check/elements/: reapply consistent pad (de)activation
Original commit message from CVS:
* tests/check/elements/audiopanorama.c: (cleanup_panorama):
* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc):
* tests/check/elements/level.c: (setup_level), (cleanup_level):
reapply consistent pad (de)activation
Jan Schmidt [Thu, 21 Dec 2006 17:03:39 +0000 (17:03 +0000)]
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
* gst-plugins-good.doap:
Add 0.10.5 doap entry
Jan Schmidt [Thu, 21 Dec 2006 15:45:02 +0000 (15:45 +0000)]
configure.ac: releasing 0.10.4, "Black Bugs"
Original commit message from CVS:
=== release 0.10.4 ===
2006-12-21 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
releasing 0.10.4, "Black Bugs"
Jan Schmidt [Thu, 21 Dec 2006 15:40:55 +0000 (15:40 +0000)]
configure.ac: releasing 0.10.5, "The Path of Thorns"
Original commit message from CVS:
=== release 0.10.5 ===
2006-12-21 Jan Schmidt <thaytan@mad.scientist.com>
* configure.ac:
releasing 0.10.5, "The Path of Thorns"
Stefan Kost [Thu, 21 Dec 2006 14:03:42 +0000 (14:03 +0000)]
tests/check/elements/mpeg2enc.c: (setup_mpeg2enc)
Original commit message from CVS:
* tests/check/elements/mpeg2enc.c: (setup_mpeg2enc)
(cleanup_mpeg2enc):
* tests/check/elements/rganalysis.c: (cleanup_rganalysis):
* tests/check/elements/wavpackdec.c: (setup_wavpackdec),
(cleanup_wavpackdec):
* tests/check/elements/wavpackenc.c: (setup_wavpackenc),
(cleanup_wavpackenc):
* tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
revert my freeze breakage
Stefan Kost [Thu, 21 Dec 2006 12:48:32 +0000 (12:48 +0000)]
tests/check/elements/: revert my freeze breakage
Original commit message from CVS:
* tests/check/elements/audiopanorama.c: (cleanup_panorama):
* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc):
* tests/check/elements/level.c: (setup_level), (cleanup_level):
revert my freeze breakage
Stefan Kost [Thu, 21 Dec 2006 08:20:10 +0000 (08:20 +0000)]
tests/check/elements/: consistent pad (de)activation
Original commit message from CVS:
* tests/check/elements/mpeg2enc.c: (setup_mpeg2enc),
(cleanup_mpeg2enc):
* tests/check/elements/rganalysis.c: (cleanup_rganalysis):
* tests/check/elements/wavpackdec.c: (setup_wavpackdec),
(cleanup_wavpackdec):
* tests/check/elements/wavpackenc.c: (setup_wavpackenc),
(cleanup_wavpackenc):
* tests/check/elements/y4menc.c: (setup_y4menc), (cleanup_y4menc):
consistent pad (de)activation
Stefan Kost [Thu, 21 Dec 2006 08:15:23 +0000 (08:15 +0000)]
tests/check/elements/: consistent pad (de)activation
Original commit message from CVS:
* tests/check/elements/audiopanorama.c: (cleanup_panorama):
* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc):
* tests/check/elements/level.c: (setup_level), (cleanup_level):
consistent pad (de)activation
Tim-Philipp Müller [Mon, 18 Dec 2006 17:11:49 +0000 (17:11 +0000)]
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).
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