1. 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
  2. 22 Feb, 2007 2 commits
  3. 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
  4. 20 Feb, 2007 1 commit
    • Sébastien Moutte's avatar
      sys/directsound/gstdirectsoundsink.*: Remove include of unused headers. · 296687a3
      Sébastien 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
  5. 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
  6. 18 Feb, 2007 2 commits
    • Sébastien Moutte's avatar
      sys/directdraw/: Prepare the plugin to move to good: · 71cf071f
      Sébastien 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
  7. 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
  8. 15 Feb, 2007 1 commit
  9. 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
      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
      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
  10. 13 Feb, 2007 3 commits
    • 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
    • Stefan Kost's avatar
      gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif · 5116ff60
      Stefan Kost authored
      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
      5116ff60
    • Stefan Kost's avatar
      Add crossreferences to glib/gobject/gstream docs. · 15075ede
      Stefan Kost authored
      Original commit message from CVS:
      * configure.ac:
      * docs/plugins/Makefile.am:
      Add crossreferences to glib/gobject/gstream docs.
      15075ede
  11. 12 Feb, 2007 6 commits
    • Tim-Philipp Müller's avatar
      gst/monoscope/: Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS... · ecc16f3e
      Tim-Philipp Müller authored
      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).
      ecc16f3e
    • Jonathan Matthew's avatar
      gst/wavparse/gstwavparse.c: Fix massive memory leak when operating in streaming mode due to · 9c49fa71
      Jonathan Matthew authored
      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.
      9c49fa71
    • Stefan Kost's avatar
      gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry... · 114afecd
      Stefan Kost authored
      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.
      114afecd
    • Stefan Kost's avatar
      sys/v4l2/: More FIXME comments and messaging changes. · 77790aa2
      Stefan Kost authored
      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.
      77790aa2
    • Stefan Kost's avatar
      gst/goom/gstgoom.*: Improved docs and use GST_DEBUG_FUNCPTR. · 14d79a36
      Stefan Kost authored
      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.
      14d79a36
    • Tim-Philipp Müller's avatar
      gst/debug/: Add code for a pushfilesrc element that implements a pushfile://... · 84c6815c
      Tim-Philipp Müller authored
      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.
      84c6815c
  12. 11 Feb, 2007 3 commits
    • Sébastien Moutte's avatar
      Makefile.am: Add win32 MANIFEST · 4b58be7f
      Sébastien Moutte authored
      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.
      4b58be7f
    • Sébastien Moutte's avatar
      gst/avi/gstavimux.c: Comment a #if 0 in caps template definition as VS6 seems to do not support it. · 9c8ea356
      Sébastien Moutte authored
      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.
      9c8ea356
    • Stefan Kost's avatar
      configure.ac: Activate monoscope when building with --enable-experimental. Fix · e687b50f
      Stefan Kost authored
      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.
      e687b50f
  13. 09 Feb, 2007 1 commit
    • Tim-Philipp Müller's avatar
      gst/avi/gstavimux.c: Explicitly cast result of pointer arithmetic to integer... · d8f5483d
      Tim-Philipp Müller authored
      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.
      d8f5483d
  14. 08 Feb, 2007 1 commit
  15. 07 Feb, 2007 3 commits
    • Tim-Philipp Müller's avatar
      docs/plugins/inspect/plugin-rtp.xml: Update for new elements. · ba2af9fa
      Tim-Philipp Müller authored
      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.
      ba2af9fa
    • Tim-Philipp Müller's avatar
      Make progressreport element post messages with the current progress on the... · b5ee4225
      Tim-Philipp Müller authored
      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.
      b5ee4225
    • Tim-Philipp Müller's avatar
      ext/hal/hal.*: Some small cleanups; deal with errors when parsing the HAL ALSA... · 784a4689
      Tim-Philipp Müller authored
      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.
      784a4689
  16. 06 Feb, 2007 4 commits
    • Tim-Philipp Müller's avatar
      gst/smpte/gstsmpte.c: Let's try this again and use the right cast this time. · 2a873dd9
      Tim-Philipp Müller authored
      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.
      2a873dd9
    • Tim-Philipp Müller's avatar
      gst/smpte/gstsmpte.c: Add cast to avoid compiler warnings with older GLib... · 7dd530e6
      Tim-Philipp Müller authored
      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.
      7dd530e6
    • Tim-Philipp Müller's avatar
      ext/gconf/: In gconfaudiosink, get the right key as the old key in do_toggle... · 881308d5
      Tim-Philipp Müller authored
      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.
      881308d5
    • Sebastian Dröge's avatar
      gst/audiofx/: Some small cleanups and port both elements to the new... · cdba2c42
      Sebastian Dröge authored
      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
      cdba2c42
  17. 03 Feb, 2007 1 commit
    • Tim-Philipp Müller's avatar
      Fix up to use the newly ported (actually working) GstAudioFilter. · f7935f9a
      Tim-Philipp Müller authored
      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.
      f7935f9a