Wim Taymans [Fri, 2 Mar 2007 10:54:49 +0000 (10:54 +0000)]
sys/sunaudio/gstsunaudiosrc.*: Remove device-name from GstSunAudioSrc. Fixes #412597.
Original commit message from CVS:
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init),
(gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property),
(gst_sunaudiosrc_open):
* sys/sunaudio/gstsunaudiosrc.h:
Remove device-name from GstSunAudioSrc. Fixes #412597.
Sebastian Dröge [Thu, 1 Mar 2007 21:50:36 +0000 (21:50 +0000)]
ext/hal/: Having NULL as UDI previously selected the default sink/src. Change this back but mention it in the debug o...
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Having NULL as UDI previously selected the default sink/src. Change
this back but mention it in the debug output.
* ext/hal/hal.c: (gst_hal_get_alsa_element),
(gst_hal_get_oss_element), (gst_hal_get_string),
(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
(gst_hal_get_audio_src):
* ext/hal/hal.h:
Refactor a bit, check all error conditions, greatly improve debugging
and fix some possible memory leaks. Also implement OSS support
and allow specifying an UDI that points to a real device. For this the
child device which supports ALSA (preferred) or OSS is used.
As a side effect this makes it impossible now to get a alsasink in
halaudiosrc and a alsasrc in halaudiosink.
Wim Taymans [Thu, 1 Mar 2007 18:47:28 +0000 (18:47 +0000)]
gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all of them are in error.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
Errors from the udp sources are not fatal unless all of them are in
error.
Jan Schmidt [Thu, 1 Mar 2007 18:14:42 +0000 (18:14 +0000)]
tests/check/Makefile.am: Disable aasink in the states test. I suspect this is the element that is calling exit(1) whe...
Original commit message from CVS:
* tests/check/Makefile.am:
Disable aasink in the states test. I suspect this is the element that
is calling exit(1) when it can't proceed.
Jan Schmidt [Thu, 1 Mar 2007 17:26:30 +0000 (17:26 +0000)]
tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, rather than picking up the already installed v...
Original commit message from CVS:
* tests/check/Makefile.am:
Draw plugins in from the build tree sys/ dir, rather than picking
up the already installed versions.
Zaheer Abbas Merali [Thu, 1 Mar 2007 10:44:36 +0000 (10:44 +0000)]
sys/ximage/gstximagesrc.c: Error out correctly when getting xcontext fails.
Original commit message from CVS:
2007-03-01 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
Error out correctly when getting xcontext fails.
Wim Taymans [Thu, 1 Mar 2007 09:29:34 +0000 (09:29 +0000)]
gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's what it will be in the future and rtspsrc...
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
Make state change to PAUSED NO_PREROLL because that's what it will be in
the future and rtspsrc relies on it.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_change_state):
Don't error out when we don't get an error from the state change
function.
Sebastian Dröge [Thu, 1 Mar 2007 01:48:59 +0000 (01:48 +0000)]
ext/hal/: Check if the device UDI is set before trying to query HAL about it and give a useful error message if it wa...
Original commit message from CVS:
* ext/hal/gsthalaudiosink.c: (do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
Check if the device UDI is set before trying to query HAL
about it and give a useful error message if it wasn't set.
* ext/hal/hal.c: (gst_hal_get_string):
Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
gives an assertion failure in D-Bus when running with
DBUS_FATAL_WARNINGS=1.
Thomas Vander Stichele [Wed, 28 Feb 2007 19:29:42 +0000 (19:29 +0000)]
update config to trunk
Original commit message from CVS:
update config to trunk
Thomas Vander Stichele [Wed, 28 Feb 2007 19:29:25 +0000 (19:29 +0000)]
configure.ac: Convert to new AG_GST style.
Original commit message from CVS:
* configure.ac:
Convert to new AG_GST style.
Thomas Vander Stichele [Wed, 28 Feb 2007 12:59:43 +0000 (12:59 +0000)]
tests/check/: add test for states
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/generic/states.c: (GST_START_TEST), (states_suite):
add test for states
Wim Taymans [Wed, 28 Feb 2007 10:58:10 +0000 (10:58 +0000)]
tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore.
Original commit message from CVS:
* tests/check/elements/.cvsignore:
Add new videofilter check to .cvsignore.
Wim Taymans [Wed, 28 Feb 2007 10:54:55 +0000 (10:54 +0000)]
gst/avi/gstavidemux.c: Fix combined flow return. Fixes #412608.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop), (gst_avi_demux_chain):
Fix combined flow return. Fixes #412608.
Wim Taymans [Wed, 28 Feb 2007 10:41:14 +0000 (10:41 +0000)]
gst/videofilter/Makefile.am: Dist header..
Original commit message from CVS:
* gst/videofilter/Makefile.am:
Dist header..
Wim Taymans [Wed, 28 Feb 2007 10:29:08 +0000 (10:29 +0000)]
gst/videofilter/gstgamma.h: Add header too.
Original commit message from CVS:
* gst/videofilter/gstgamma.h:
Add header too.
Mark Nauwelaerts [Wed, 28 Feb 2007 10:17:15 +0000 (10:17 +0000)]
gst/videofilter/: Port gamma filter to 0.10. Fixes #412704.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videofilter/Makefile.am:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
(gst_gamma_get_property), (gst_gamma_calculate_tables),
(oil_tablelookup_u8), (gst_gamma_set_caps),
(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
Port gamma filter to 0.10. Fixes #412704.
* tests/check/Makefile.am:
* tests/check/elements/videofilter.c: (setup_filter),
(cleanup_filter), (check_filter), (GST_START_TEST),
(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
Add unit tests for videofilters.
Wim Taymans [Wed, 28 Feb 2007 10:06:27 +0000 (10:06 +0000)]
gst/rtsp/URLS: Add another interesting test url.
Original commit message from CVS:
* gst/rtsp/URLS:
Add another interesting test url.
* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
Don't allow getting header fields from data packets.
Michael Smith [Tue, 27 Feb 2007 23:43:08 +0000 (23:43 +0000)]
ext/shout2/gstshout2.*: Add a property for username.
Original commit message from CVS:
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_start),
(gst_shout2send_set_property), (gst_shout2send_get_property):
* ext/shout2/gstshout2.h:
Add a property for username.
Christian Schaller [Tue, 27 Feb 2007 12:02:03 +0000 (12:02 +0000)]
update copyright statements
Original commit message from CVS:
update copyright statements
Christian Schaller [Tue, 27 Feb 2007 11:59:21 +0000 (11:59 +0000)]
update copyright statement
Original commit message from CVS:
update copyright statement
Edward Hervey [Tue, 27 Feb 2007 11:30:19 +0000 (11:30 +0000)]
sys/osxvideo/: Disable the cocoa event loop since it's a huge memory leak. Should only matter if the sink isn't used ...
Original commit message from CVS:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Disable the cocoa event loop since it's a huge memory leak. Should only
matter if the sink isn't used within an NSApp (which has already got
a coca event loop).
Remove all unused code.
Jan Schmidt [Mon, 26 Feb 2007 12:07:14 +0000 (12:07 +0000)]
gst/rtsp/Makefile.am: Fix make check too.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
Fix make check too.
Jan Schmidt [Mon, 26 Feb 2007 10:00:28 +0000 (10:00 +0000)]
gst/rtsp/base64.*: Commit missing files for base64 encoding.
Original commit message from CVS:
* gst/rtsp/base64.c: (util_base64_encode):
* gst/rtsp/base64.h:
Commit missing files for base64 encoding.
Loïc Minier [Sat, 24 Feb 2007 22:57:49 +0000 (22:57 +0000)]
Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
Original commit message from CVS:
Patch by: Loïc Minier <lool+gnome at via ecp fr>
* configure.ac:
* ext/annodex/Makefile.am:
* ext/jpeg/Makefile.am:
* ext/speex/Makefile.am:
* gst/alpha/Makefile.am:
* gst/cutter/Makefile.am:
* gst/debug/Makefile.am:
* gst/effectv/Makefile.am:
* gst/goom/Makefile.am:
* gst/level/Makefile.am:
* gst/smpte/Makefile.am:
* gst/videofilter/Makefile.am:
Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
Tim-Philipp Müller [Sat, 24 Feb 2007 22:52:47 +0000 (22:52 +0000)]
Fix build with LDFLAGS='-Wl,-z,defs'.
Original commit message from CVS:
* configure.ac:
* ext/gsm/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/wavpack/Makefile.am:
* gst/equalizer/Makefile.am:
* gst/filter/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/replaygain/Makefile.am:
* gst/speed/Makefile.am:
Fix build with LDFLAGS='-Wl,-z,defs'.
Jan Schmidt [Fri, 23 Feb 2007 19:12:52 +0000 (19:12 +0000)]
gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken from icecast to replace it. Relicensed fr...
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/rtspconnection.c: (append_auth_header),
(rtsp_connection_send), (rtsp_connection_set_auth):
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
from icecast to replace it. Relicensed from GPL courtesy of Mike
Smith.
Jan Schmidt [Fri, 23 Feb 2007 18:12:27 +0000 (18:12 +0000)]
gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
Edgard Lima [Thu, 22 Feb 2007 17:53:26 +0000 (17:53 +0000)]
Fix segfault when oppening a radio device.
Original commit message from CVS:
Fix segfault when oppening a radio device.
Stefan Kost [Thu, 22 Feb 2007 14:35:28 +0000 (14:35 +0000)]
Fix level for multi-channel case.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_set_caps),
(gst_level_transform_ip):
* sys/v4l2/README:
* tests/check/elements/level.c: (GST_START_TEST):
Fix level for multi-channel case.
Stefan Kost [Wed, 21 Feb 2007 10:18:12 +0000 (10:18 +0000)]
gst/level/gstlevel.*: Use function pointer for process function and add process functions for float audio.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use function pointer for process function and add process functions
for float audio.
Sébastien Moutte [Tue, 20 Feb 2007 21:34:00 +0000 (21:34 +0000)]
sys/directsound/gstdirectsoundsink.*: Remove include of unused headers.
Original commit message from CVS:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove include of unused headers.
* sys/waveform/gstwaveformplugin.c:
* sys/waveform/gstwaveformsink.c:
* sys/waveform/gstwaveformsink.h:
* win32/vs6/libgstwaveform.dsp:
Add a new waveform plugin which includes an audio sink
element using the WaveForm win32 API.
* win32/MANIFEST:
Add the new project file form waveform plugin.
Stefan Kost [Mon, 19 Feb 2007 12:22:43 +0000 (12:22 +0000)]
sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO, fixes #407369
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init):
Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
fixes #407369
Sébastien Moutte [Sun, 18 Feb 2007 18:00:51 +0000 (18:00 +0000)]
sys/directdraw/: Prepare the plugin to move to good:
Original commit message from CVS:
* sys/directdraw/gstdirectdrawplugin.c:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Prepare the plugin to move to good:
Remove unused/untested code (rendering to an extern surface,
yuv format rendering).Use GST_(DEBUG/*)_OBJECT macros
Rename all functions from gst_directdrawsink to gst_directdraw_sink.
Add gtk doc section
Fix a bug in gst_directdraw_sink_show_frame, memcpy line by line
respecting destination surface stride.
* sys/directsound/gstdirectsoundplugin.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Prepare the plugin to move to good:
Rename all functions from gst_directsoundsink to gst_directsound_sink.
Add gtk doc section
* win32/common/config.h.in:
* win32/MANIFEST:
Add config.h.in
Wim Taymans [Sun, 18 Feb 2007 13:24:26 +0000 (13:24 +0000)]
gst/rtp/: Added simple mpeg transport stream payloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
(gst_rtp_mp2t_pay_plugin_init):
* gst/rtp/gstrtpmp2tpay.h:
Added simple mpeg transport stream payloader.
Wim Taymans [Fri, 16 Feb 2007 12:32:01 +0000 (12:32 +0000)]
gst/rtsp/URLS: Add example H264 rtsp url.
Original commit message from CVS:
* gst/rtsp/URLS:
Add example H264 rtsp url.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
Don't convert values to lowercase or we might mess up base64 encoded
properties.
Wim Taymans [Fri, 16 Feb 2007 12:30:22 +0000 (12:30 +0000)]
gst/rtp/README: Fix case of string params.
Original commit message from CVS:
* gst/rtp/README:
Fix case of string params.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Fix depayloader, support more packet types.
Add sync codes to make sure the packetizer can do its job.
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
Fix caps case again.
Wim Taymans [Thu, 15 Feb 2007 12:26:28 +0000 (12:26 +0000)]
gst/rtp/gstrtph264depay.c: Set right caps on output buffers.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Set right caps on output buffers.
Wim Taymans [Wed, 14 Feb 2007 17:04:47 +0000 (17:04 +0000)]
gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
As spotted by: Peter Kjellerstedt <pkj at axis com>:
Clear stack allocated SDPMedia struct before calling _init() on it.
Clarify this in the docs as well.
Jan Schmidt [Wed, 14 Feb 2007 17:01:25 +0000 (17:01 +0000)]
ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching states, as it makes the element non-reusa...
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
(do_change_child):
Don't reset the profile when going switching states, as it makes
the element non-reusable.
jp.liu [Wed, 14 Feb 2007 15:24:50 +0000 (15:24 +0000)]
gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
(sdp_parse_line):
* gst/rtsp/sdpmessage.h:
Based on patch by: jp.liu <jp_liu at astrocom dot cn>
Fix memory management of SDP messages. Fixes #407793.
zhangfei gao [Wed, 14 Feb 2007 12:07:01 +0000 (12:07 +0000)]
gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
Original commit message from CVS:
Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>
* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
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).
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