1. 01 Mar, 2007 4 commits
    • Jan Schmidt's avatar
      tests/check/Makefile.am: Draw plugins in from the build tree sys/ dir, rather... · 8ed6a4ac
      Jan Schmidt authored
      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.
      8ed6a4ac
    • Zaheer Abbas Merali's avatar
      sys/ximage/gstximagesrc.c: Error out correctly when getting xcontext fails. · 3c316331
      Zaheer Abbas Merali authored
      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.
      3c316331
    • Wim Taymans's avatar
      gst/rtsp/gstrtpdec.c: Make state change to PAUSED NO_PREROLL because that's... · dc212cdb
      Wim Taymans authored
      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.
      dc212cdb
    • Sebastian Dröge's avatar
      ext/hal/: Check if the device UDI is set before trying to query HAL about it... · 16490dc0
      Sebastian Dröge authored
      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.
      16490dc0
  2. 28 Feb, 2007 9 commits
  3. 27 Feb, 2007 4 commits
    • Michael Smith's avatar
      ext/shout2/gstshout2.*: Add a property for username. · 570e2ffd
      Michael Smith authored
      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.
      570e2ffd
    • Christian Schaller's avatar
      update copyright statements · b9820ee9
      Christian Schaller authored
      Original commit message from CVS:
      update copyright statements
      b9820ee9
    • Christian Schaller's avatar
      update copyright statement · 83e7dadb
      Christian Schaller authored
      Original commit message from CVS:
      update copyright statement
      83e7dadb
    • Edward Hervey's avatar
      sys/osxvideo/: Disable the cocoa event loop since it's a huge memory leak.... · a471de34
      Edward Hervey authored
      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.
      a471de34
  4. 26 Feb, 2007 2 commits
  5. 24 Feb, 2007 2 commits
    • Loïc Minier's avatar
      Fix build with LDFLAGS='-Wl,-z,defs' (#410997) · 682312a2
      Loïc Minier authored and Tim-Philipp Müller's avatar Tim-Philipp Müller committed
      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)
      682312a2
    • Tim-Philipp Müller's avatar
      Fix build with LDFLAGS='-Wl,-z,defs'. · e854c41c
      Tim-Philipp Müller authored
      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'.
      e854c41c
  6. 23 Feb, 2007 2 commits
    • Jan Schmidt's avatar
      gst/rtsp/: g_base64_encode is a GLib 2.12 function. Use an equivalent taken... · 825cf238
      Jan Schmidt authored
      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.
      825cf238
    • Jan Schmidt's avatar
      gst/rtsp/: Implement simple Basic Authentication support so that urls like... · 66df66da
      Jan Schmidt authored
      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.
      66df66da
  7. 22 Feb, 2007 2 commits
  8. 21 Feb, 2007 1 commit
    • Stefan Kost's avatar
      gst/level/gstlevel.*: Use function pointer for process function and add... · 6e44a9c6
      Stefan Kost authored
      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.
      6e44a9c6
  9. 20 Feb, 2007 1 commit
    • Sebastien Moutte's avatar
      sys/directsound/gstdirectsoundsink.*: Remove include of unused headers. · 296687a3
      Sebastien Moutte authored
      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.
      296687a3
  10. 19 Feb, 2007 1 commit
    • Stefan Kost's avatar
      sys/v4l2/v4l2src_calls.c: Readd GST_ELEMENT_ERROR if we can't reenque buffers... · 2d1b4202
      Stefan Kost authored
      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
      2d1b4202
  11. 18 Feb, 2007 2 commits
    • Sebastien Moutte's avatar
      sys/directdraw/: Prepare the plugin to move to good: · 71cf071f
      Sebastien Moutte authored
      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
      71cf071f
    • Wim Taymans's avatar
      gst/rtp/: Added simple mpeg transport stream payloader. · bd4b1f68
      Wim Taymans authored
      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.
      bd4b1f68
  12. 16 Feb, 2007 2 commits
    • Wim Taymans's avatar
      gst/rtsp/URLS: Add example H264 rtsp url. · 7fd02504
      Wim Taymans authored
      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.
      7fd02504
    • Wim Taymans's avatar
      gst/rtp/README: Fix case of string params. · dc325990
      Wim Taymans authored
      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.
      dc325990
  13. 15 Feb, 2007 1 commit
  14. 14 Feb, 2007 6 commits
    • Wim Taymans's avatar
      gst/rtsp/sdpmessage.c: Clear stack allocated SDPMedia struct before calling _init() on it. · df5916db
      Wim Taymans authored
      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.
      df5916db
    • Jan Schmidt's avatar
      ext/gconf/gstgconfaudiosink.c: Don't reset the profile when going switching... · 3b5868a9
      Jan Schmidt authored
      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.
      3b5868a9
    • jp.liu's avatar
      gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793. · 6021b924
      jp.liu authored and Wim Taymans's avatar Wim Taymans committed
      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.
      6021b924
    • zhangfei gao's avatar
      gst/avi/gstavimux.c: Allow muxing video/x-h264 (was already in the caps). Fixes #407780. · d08a7da7
      zhangfei gao authored
      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.
      d08a7da7
    • jp.liu's avatar
      gst/rtsp/rtspurl.c: Fix parsing of password field in url. Fixes #407797. · a8f72c67
      jp.liu authored and Wim Taymans's avatar Wim Taymans committed
      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.
      a8f72c67
    • Wim Taymans's avatar
      gst/wavparse/gstwavparse.*: Update docs. · 2644d717
      Wim Taymans authored
      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.
      2644d717
  15. 13 Feb, 2007 1 commit
    • Jan Schmidt's avatar
      Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child... · b1aa8fef
      Jan Schmidt authored
      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.
      b1aa8fef