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* gst/rtp/gstrtpmp2tdepay.c: Fix compilation.Wim Taymans2007-03-022-1/+7
| | | | | | Original commit message from CVS: * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process): Fix compilation.
* gst/rtp/gstrtpmp2tdepay.*: Add support to strip off proprietary headers. ↵Thijs Vermeir2007-03-023-11/+51
| | | | | | | | | | | | | Fixes #350278. Original commit message from CVS: Patch by: Thijs Vermeir <thijsvermeir at gmail dot com> * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process), (gst_rtp_mp2t_depay_set_property), (gst_rtp_mp2t_depay_get_property): * gst/rtp/gstrtpmp2tdepay.h: Add support to strip off proprietary headers. Fixes #350278.
* ext/hal/hal.c: Fix compilation.Wim Taymans2007-03-022-0/+10
| | | | | | Original commit message from CVS: * ext/hal/hal.c: Fix compilation.
* sys/sunaudio/gstsunaudiosrc.*: Remove device-name from GstSunAudioSrc. Fixes ↵Wim Taymans2007-03-023-15/+9
| | | | | | | | | | | #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.
* ext/hal/: Having NULL as UDI previously selected the default sink/src. ↵Sebastian Dröge2007-03-015-64/+295
| | | | | | | | | | | | | | | | | | | | | 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.
* gst/rtsp/gstrtspsrc.c: Errors from the udp sources are not fatal unless all ↵Wim Taymans2007-03-012-13/+59
| | | | | | | | | | 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.
* tests/check/Makefile.am: Disable aasink in the states test. I suspect this ↵Jan Schmidt2007-03-012-1/+8
| | | | | | | | | 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.
* tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, ↵Jan Schmidt2007-03-012-1/+7
| | | | | | | | | 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.
* sys/ximage/gstximagesrc.c: Error out correctly when getting xcontext fails.Zaheer Abbas Merali2007-03-012-0/+12
| | | | | | | | 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.
* gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's ↵Wim Taymans2007-03-013-4/+16
| | | | | | | | | | | | | 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.
* ext/hal/: Check if the device UDI is set before trying to query HAL about it ↵Sebastian Dröge2007-03-015-2/+27
| | | | | | | | | | | | | | 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.
* update config to trunkThomas Vander Stichele2007-02-281-4/+4
| | | | | Original commit message from CVS: update config to trunk
* configure.ac: Convert to new AG_GST style.Thomas Vander Stichele2007-02-283-77/+84
| | | | | | Original commit message from CVS: * configure.ac: Convert to new AG_GST style.
* tests/check/: add test for statesThomas Vander Stichele2007-02-283-0/+127
| | | | | | | Original commit message from CVS: * tests/check/Makefile.am: * tests/check/generic/states.c: (GST_START_TEST), (states_suite): add test for states
* tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore.Wim Taymans2007-02-282-0/+6
| | | | | | Original commit message from CVS: * tests/check/elements/.cvsignore: Add new videofilter check to .cvsignore.
* gst/avi/gstavidemux.c: Fix combined flow return. Fixes #412608.Wim Taymans2007-02-282-52/+48
| | | | | | | | 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.
* gst/videofilter/Makefile.am: Dist header..Wim Taymans2007-02-282-1/+6
| | | | | | Original commit message from CVS: * gst/videofilter/Makefile.am: Dist header..
* gst/videofilter/gstgamma.h: Add header too.Wim Taymans2007-02-282-0/+80
| | | | | | Original commit message from CVS: * gst/videofilter/gstgamma.h: Add header too.
* gst/videofilter/: Port gamma filter to 0.10. Fixes #412704.Mark Nauwelaerts2007-02-285-278/+409
| | | | | | | | | | | | | | | | | 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.
* gst/rtsp/URLS: Add another interesting test url.Wim Taymans2007-02-283-0/+12
| | | | | | | | 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.
* ext/shout2/gstshout2.*: Add a property for username.Michael Smith2007-02-273-2/+26
| | | | | | | | | 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.
* update copyright statementsChristian Schaller2007-02-2710-3/+57
| | | | | Original commit message from CVS: update copyright statements
* update copyright statementChristian Schaller2007-02-276-2/+29
| | | | | Original commit message from CVS: update copyright statement
* sys/osxvideo/: Disable the cocoa event loop since it's a huge memory leak. ↵Edward Hervey2007-02-274-149/+2
| | | | | | | | | | | | | | 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.
* gst/rtsp/Makefile.am: Fix make check too.Jan Schmidt2007-02-262-1/+6
| | | | | | Original commit message from CVS: * gst/rtsp/Makefile.am: Fix make check too.
* gst/rtsp/base64.*: Commit missing files for base64 encoding.Jan Schmidt2007-02-263-0/+107
| | | | | | | Original commit message from CVS: * gst/rtsp/base64.c: (util_base64_encode): * gst/rtsp/base64.h: Commit missing files for base64 encoding.
* Fix build with LDFLAGS='-Wl,-z,defs' (#410997)Loïc Minier2007-02-2413-12/+40
| | | | | | | | | | | | | | | | | | 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)
* Fix build with LDFLAGS='-Wl,-z,defs'.Tim-Philipp Müller2007-02-243-5/+5
| | | | | | | | | | | | | | | 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'.
* gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken ↵Jan Schmidt2007-02-233-4/+14
| | | | | | | | | | | | 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.
* gst/rtsp/: Implement simple Basic Authentication support so that urls like ↵Jan Schmidt2007-02-238-37/+343
| | | | | | | | | | | | | | | | | | | | | | | 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.
* Fix segfault when oppening a radio device.Edgard Lima2007-02-224-25/+40
| | | | | Original commit message from CVS: Fix segfault when oppening a radio device.
* Fix level for multi-channel case.Stefan Kost2007-02-224-2/+15
| | | | | | | | | 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.
* gst/level/gstlevel.*: Use function pointer for process function and add ↵Stefan Kost2007-02-213-67/+140
| | | | | | | | | | | 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.
* sys/directsound/gstdirectsoundsink.*: Remove include of unused headers.Sébastien Moutte2007-02-206-17/+866
| | | | | | | | | | | | | | | 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.
* sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque ↵Stefan Kost2007-02-192-3/+18
| | | | | | | | | | | 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
* sys/directdraw/: Prepare the plugin to move to good:Sébastien Moutte2007-02-186-1020/+635
| | | | | | | | | | | | | | | | | | | | | | | 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
* gst/rtp/: Added simple mpeg transport stream payloader.Wim Taymans2007-02-185-0/+233
| | | | | | | | | | | | 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.
* gst/rtsp/URLS: Add example H264 rtsp url.Wim Taymans2007-02-163-18/+36
| | | | | | | | | | 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.
* gst/rtp/README: Fix case of string params.Wim Taymans2007-02-166-67/+133
| | | | | | | | | | | | | | 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.
* gst/rtp/gstrtph264depay.c: Set right caps on output buffers.Wim Taymans2007-02-152-4/+7
| | | | | | Original commit message from CVS: * gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process): Set right caps on output buffers.
* gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling ↵Wim Taymans2007-02-142-0/+14
| | | | | | | | | | _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.
* ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching ↵Jan Schmidt2007-02-142-2/+10
| | | | | | | | | | 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.
* gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.jp.liu2007-02-143-40/+216
| | | | | | | | | | | | | 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.
* gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). ↵zhangfei gao2007-02-142-2/+11
| | | | | | | | | 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.
* gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797.jp.liu2007-02-142-1/+8
| | | | | | | 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.
* gst/wavparse/gstwavparse.*: Update docs.Wim Taymans2007-02-143-240/+281
| | | | | | | | | | | | | | | | | | | | | | | | 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.
* Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child ↵Jan Schmidt2007-02-1310-156/+533
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endifStefan Kost2007-02-132-7/+26
| | | | | | | | | | | | 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
* Add crossreferences to glib/gobject/gstream docs.Stefan Kost2007-02-133-3/+20
| | | | | | | Original commit message from CVS: * configure.ac: * docs/plugins/Makefile.am: Add crossreferences to glib/gobject/gstream docs.
* gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing ↵Tim-Philipp Müller2007-02-123-3/+10
| | | | | | | | | | 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).
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