autogen.sh: Require automake 1.7

Original commit message from CVS:
* autogen.sh:
Require automake 1.7
* ext/alsaspdif/Makefile.am:
* ext/divx/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/neon/Makefile.am:
* ext/sdl/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/theora/Makefile.am:
* ext/wavpack/Makefile.am:
* ext/xvid/Makefile.am:
* gst/modplug/Makefile.am:
Fix up Makefile.am accordingly.

docs/plugins/inspect/: Add jack and update.

Original commit message from CVS:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dfbvideosink.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-glimagesink.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
Add jack and update.

configure.ac: Don't build equalizer unless we have core from CVS (it won't work with earlier versions due to GstChild...

Original commit message from CVS:
* configure.ac:
Don't build equalizer unless we have core from CVS (it won't
work with earlier versions due to GstChildProxy brokeness).
Also up requirements to last released core/base.

ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).

Original commit message from CVS:
2007-04-27  Julien MOUTTE  <julien@moutte.net>

* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.

gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.

Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
Move reconsideration code to the rtpsession object.
Simplify timout handling and add reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
(obtain_source), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_bye), (rtp_session_process_rtcp),
(calculate_rtcp_interval), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_start_rtcp),
(session_report_blocks), (session_cleanup), (session_sdes),
(session_bye), (is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Handle timeout of inactive sources and senders.
Implement BYE scheduling.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_process_sr), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add members to check for timeouts.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
(rtp_stats_calculate_bye_interval):
* gst/rtpmanager/rtpstats.h:
Use RFC algorithm for calculating the reporting interval.

docs/plugins/: Add documentation for osxvideo

Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/inspect/plugin-osxvideo.xml:
Add documentation for osxvideo

gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.

Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Implement forward and reverse reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
(rtp_session_get_num_active_sources), (rtp_session_process_sr),
(session_report_blocks):
* gst/rtpmanager/rtpsession.h:
Small cleanups.

gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.

Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.

gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.

Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.

gst/rtpmanager/gstrtpbin.c: fix for pad name change

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.

Plug some leaks; try to make build bot happy again.

Original commit message from CVS:
* gst/y4m/gsty4mencode.c: (gst_y4m_encode_init),
(gst_y4m_encode_setcaps):
* tests/check/elements/y4menc.c: (GST_START_TEST):
Plug some leaks; try to make build bot happy again.

gst/Makefile.am: Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).

Original commit message from CVS:
* gst/Makefile.am:
Fix distcheck, hopefully (rtpmanager is already in GST_PLUGINS_ALL).

gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.

Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.

gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.

Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.

tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).

Original commit message from CVS:
* tests/check/elements/audioresample.c:
Add unit test for audioresample shutdown crasher (#420106).

ext/faad/gstfaad.c: FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in quicktime because of sample rate mi...

Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_open_decoder):
FAAD fails to decode low (e.g. 8 kHz) sample rate AAC data in
quicktime because of sample rate mismatches.
Reenable overriding the implicit SBR behaviour (accidently changed?)
to allow playback of these files.

configure.ac: Change rtpmanager disabling to keep -bad releasable.

Original commit message from CVS:
* configure.ac:
Change rtpmanager disabling to keep -bad releasable.

Fix wtay's hack.  rtpmanager is disabled in configure.ac on line 268.

Original commit message from CVS:
* configure.ac:
* gst/Makefile.am:
Fix wtay's hack.  rtpmanager is disabled in configure.ac on
line 268.

gst/Makefile.am: Add rtpmanager dir to dist.

Original commit message from CVS:
* gst/Makefile.am:
Add rtpmanager dir to dist.

configure.ac: Disable rtpmanager for now because it depends on CVS -base.

Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.

gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.

Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.

gst/nsf/types.h: Rename #ifndef header guard symbol to something less generic, so types.h doesn't get skipped over wh...

Original commit message from CVS:
* gst/nsf/types.h:
Rename #ifndef header guard symbol to something less generic, so
types.h doesn't get skipped over when compiling on MingW. Include
GLib headers and use those to set the endianness and the basic
types so that this isn't entirely broken for non-x86 architectures.

gst/mve/gstmvedemux.c: Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on

Original commit message from CVS:
* gst/mve/gstmvedemux.c: (gst_mve_audio_init):
Use G_LITTLE_ENDIAN instead of LITTLE_ENDIAN, so stuff compiles on
MingW (no idea though why we add a BYTE_ORDER endianness field if
the audio is compressed).

ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied

Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.

docs/plugins/inspect/: Add xml doc files for Windows sinks

Original commit message from CVS:
* docs/plugins/inspect/plugin-directdraw.xml:
* docs/plugins/inspect/plugin-directsound.xml:
* docs/plugins/inspect/plugin-waveform.xml:
Add xml doc files for Windows sinks
* win32/vs6/libgstqtdemux.dsp:
* win32/vs6/libgstmpegvideoparse.dsp:
* win32/vs6/gst_plugins_bad.dsw:
Update projects files.

gst/rtpmanager/: Protect lists and structures with locks.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
(create_recv_rtp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_finalize),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_request_new_pad):
Protect lists and structures with locks.
Return FLOW_OK from RTCP messages for now.

gst/qtdemux/qtdemux.c: Make timescale 32 bits again so we don't screw up the pts_offset calculations.

Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
Make timescale 32 bits again so we don't screw up the pts_offset
calculations.

gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
Emit pt map requests and cache results.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps),
(gst_jitter_buffer_sink_setcaps),
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Emit request-pt-map signals.

gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
Some more custom marshallers.
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
(pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
* gst/rtpmanager/gstrtpbin.h:
Prepare for caching pt maps.
Connect to signals to collect pt maps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add request_clock_rate signal.
Use scale insteat of scale_int because the later does not deal with
negative numbers.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_chain):
* gst/rtpmanager/gstrtpptdemux.h:
Implement request-pt-map signal.

gst/qtdemux/: Handle version 1 mdhd atoms to get extended precision durations.

Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),
(qtdemux_parse_samples), (qtdemux_parse_segments),
(qtdemux_parse_trak), (qtdemux_parse_tree):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_mdhd):
Handle version 1 mdhd atoms to get extended precision durations.
Fixes #426972.

gst/rtpmanager/: Added custom marshallers for signals.

Original commit message from CVS:
* gst/rtpmanager/.cvsignore:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpbin-marshal.list:
Added custom marshallers for signals.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Prepare for emiting pt map signals.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Fix signals.

gst/rtpmanager/gstrtpbin.*: Provide a clock.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
* gst/rtpmanager/gstrtpbin.h:
Provide a clock.

gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
Fix pad template name parsing.

gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.

Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Add some debug and comments.
Fix double unref() in error cases.

gst/qtdemux/gstrtpxqtdepay.*: Try to recover from packet loss a little better.

Original commit message from CVS:
* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
(gst_rtp_xqt_depay_change_state):
* gst/qtdemux/gstrtpxqtdepay.h:
Try to recover from packet loss a little better.

gst/rtpmanager/gstrtpbin.*: Add debugging category.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.

update dutch

Original commit message from CVS:
update dutch

po/: Added Danish translation.

Original commit message from CVS:
submitted by: Mogens Jaeger <mogens@jaeger.tf>
* po/LINGUAS:
* po/da.po:
Added Danish translation.

gst/rtpmanager/: Added simple SSRC demuxer.

Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
(create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
(gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Added simple SSRC demuxer.

ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links.

Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.

sys/glsink/glimagesink.c: Fix handling of video/x-raw-yuv.  Add overlay handling.

Original commit message from CVS:
* sys/glsink/glimagesink.c:
Fix handling of video/x-raw-yuv.  Add overlay handling.

update with rtp plugin

Original commit message from CVS:
update with rtp plugin

gst/rtpmanager/: Some more ghostpad magic.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
Some more ghostpad magic.

gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.

Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
Add .h file so it can be disted properly.

Add RTP session management elements. Still in progress.

Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.

ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th...

Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
(gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps),
(gst_wavpack_dec_clip_outgoing_buffer),
(gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain):
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_chain):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.c:
Don't play audioconvert. As wavpack wants/outputs all samples with
width==32 and depth=[1,32] accept this and let audioconvert convert
to accepted formats instead of doing it in the element for n*8 depths.
This also adds support for non-n*8 depths and prevents some useless
memory allocations. Fixes #421598
Also add a workaround for bug #421542 in wavpackenc for now...
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
* tests/check/elements/wavpackenc.c: (GST_START_TEST):
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
Consider the change above in the unit tests and test if the correct
caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in
the wavpackparse unit test.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init),
(gst_wavpack_dec_sink_set_caps):
Set caps on the src pad as soon as possible.
* ext/wavpack/gstwavpackdec.h:
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/gstwavpackparse.h:
Fix indention. gst-indent is now called by cicl.

gst/qtdemux/: Process 'ctts' atoms, which are present in AVC ISO files (.mov files with h264 video).

Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_prepare_current_sample),
(gst_qtdemux_chain), (qtdemux_parse_samples):
* gst/qtdemux/qtdemux_dump.c: (qtdemux_dump_ctts):
* gst/qtdemux/qtdemux_dump.h:
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Process 'ctts' atoms, which are present in AVC ISO files (.mov files
with h264 video).
Use the offset present in 'ctts' to calculate the PTS for each packet
and set the PTS on outgoing buffers.
Fixes #423283

ext/xvid/gstxviddec.c: Add some debug log and fix a stupid output buffer duration bug.

Original commit message from CVS:
2007-03-27  Julien MOUTTE  <julien@moutte.net>

* ext/xvid/gstxviddec.c: (gst_xviddec_chain): Add some
debug log and fix a stupid output buffer duration bug.

update spec file for x264 encoder

Original commit message from CVS:
update spec file for x264 encoder

Add libx264-based h264 encoder plugin (#421110). Probably doesn't handle 'odd' widths and heights correctly yet.

Original commit message from CVS:
Patch by: Michal Benes <michal.benes at itonis tv>
Patch by: Josef Zlomek <josef.zlomek at itonis tv>
* configure.ac:
* ext/Makefile.am:
* ext/x264/Makefile.am:
* ext/x264/gstx264enc.c: (gst_x264_enc_me_get_type),
(gst_x264_enc_analyse_get_type),
(gst_x264_enc_timestamp_queue_init),
(gst_x264_enc_timestamp_queue_free),
(gst_x264_enc_timestamp_queue_put),
(gst_x264_enc_timestamp_queue_get), (gst_x264_enc_header_buf),
(gst_x264_enc_set_src_caps), (gst_x264_enc_sink_set_caps),
(gst_x264_enc_base_init), (gst_x264_enc_class_init),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_close_encoder), (gst_x264_enc_dispose),
(gst_x264_enc_sink_event), (gst_x264_enc_chain),
(gst_x264_enc_encode_frame), (gst_x264_enc_change_state),
(gst_x264_enc_set_property), (gst_x264_enc_get_property),
(plugin_init):
* ext/x264/gstx264enc.h:
Add libx264-based h264 encoder plugin (#421110). Probably doesn't
handle 'odd' widths and heights correctly yet.

gst/interleave/deinterleave.c: Remove 'channel-positions' field when munging input caps into 1-channel output caps (I...

Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps):
Remove 'channel-positions' field when munging input caps into
1-channel output caps (I guess technically we should set the
position for each channel on the output caps if it's non-NONE,
but I'll save that as a task for another day).

gst/vmnc/vmncdec.c: Redesign to include a parser for raw files (no timestamps in that mode yet, though).

Original commit message from CVS:
* gst/vmnc/vmncdec.c: (gst_vmnc_dec_class_init),
(gst_vmnc_dec_init), (vmnc_dec_finalize), (gst_vmnc_dec_reset),
(vmnc_handle_wmvi_rectangle), (render_colour_cursor),
(render_cursor), (vmnc_make_buffer), (vmnc_handle_wmvd_rectangle),
(vmnc_handle_wmve_rectangle), (vmnc_handle_wmvf_rectangle),
(vmnc_handle_wmvg_rectangle), (vmnc_handle_wmvh_rectangle),
(vmnc_handle_wmvj_rectangle), (render_raw_tile), (render_subrect),
(vmnc_handle_raw_rectangle), (vmnc_handle_copy_rectangle),
(vmnc_handle_hextile_rectangle), (vmnc_handle_packet),
(vmnc_dec_setcaps), (vmnc_dec_chain_frame), (vmnc_dec_chain),
(vmnc_dec_set_property), (vmnc_dec_get_property):
Redesign to include a parser for raw files (no timestamps in that
mode yet, though).

gst/interleave/deinterleave.c: Don't leak input buffer in chain function; maintain our own list of source pads - ther...

Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_remove_pads), (gst_deinterleave_process),
(gst_deinterleave_chain):
Don't leak input buffer in chain function; maintain our own list of
source pads - there are no guarantees about the order of the list
in the GstElement struct, and we want a very specific order; lastly,
some more debugging.

ext/neon/gstneonhttpsrc.c: Alloc user agent string only once.

Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_init):
Alloc user agent string only once.

ext/wavpack/gstwavpackparse.c: Revert last commit, preventing infinite plugging loops with ranks is no clean solution...

Original commit message from CVS:
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Revert last commit, preventing infinite plugging loops with ranks
is no clean solution and in general there's no reason why one wants
to parse framed wavpack data again.

ext/wavpack/gstwavpackenc.c: Send the new segment event in time format instead of bytes. This allows "wavpackenc ! wa...

Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_push_block):
Send the new segment event in time format instead of bytes. This
allows "wavpackenc ! wavpackdec ! someaudiosink" pipelines.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Accept framed and non-framed input, wavpackparse doesn't care. To
prevent "wavpackparse ! wavpackparse ! ..." pipelines lower the
rank of wavpackparse by one. This allows "wavpackenc ! wavpackparse !
..." pipelines.

gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...

Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.

ext/wavpack/gstwavpackdec.c: Revert to use gst_pad_alloc_buffer() here. We can and should use it.

Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Revert to use gst_pad_alloc_buffer() here. We can and should use it.
Thanks to Jan and Mike for noticing my mistake.

ext/wavpack/gstwavpackenc.*: Put the write helpers into the GstWavpackEnc struct directly and not as a pointer to sav...

Original commit message from CVS:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_chain),
(gst_wavpack_enc_rewrite_first_block):
* ext/wavpack/gstwavpackenc.h:
Put the write helpers into the GstWavpackEnc struct directly and not
as a pointer to save two small, but useless mallocs. This also makes
it possible to drop the finalize method.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_push_buffer):
For consistency reasons also set GST_BUFFER_OFFSET_END on the outgoing
buffers the same way wavpackenc does it.

ext/wavpack/gstwavpackdec.c: Don't use gst_pad_alloc_buffer() as we might clip the buffer later and

Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
Don't use gst_pad_alloc_buffer() as we might clip the buffer later and
BaseTransform-based elements will likely break because of wrong
unit-size. Also plug a possible memleak that happens when decoding
fails for some reason.

ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks.

Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.

sys/osxvideo/osxvideosink.m: Fix previous commit, we want to pass the NSView in the message.

Original commit message from CVS:
* sys/osxvideo/osxvideosink.m:
Fix previous commit, we want to pass the NSView in the message.

sys/osxvideo/osxvideosink.m: Emit 'have-ns-view' message when working in embedded mode. The message will contain a po...

Original commit message from CVS:
* sys/osxvideo/osxvideosink.m:
Emit 'have-ns-view' message when working in embedded mode. The message
will contain a pointer to the newly created NSView.

gst/mpegvideoparse/: Move the MPEG specific byte parsing into the mpegpacketiser code.

Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_util_find_start_code),
(collect_packets), (set_par_from_dar), (set_fps_from_code),
(mpeg_util_parse_extension_packet), (mpeg_util_parse_sequence_hdr),
(mpeg_util_parse_picture_hdr):
* gst/mpegvideoparse/mpegpacketiser.h:
* gst/mpegvideoparse/mpegvideoparse.c:
(mpegvideoparse_handle_sequence), (mpegvideoparse_handle_picture),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (plugin_init):
* gst/mpegvideoparse/mpegvideoparse.h:
Move the MPEG specific byte parsing into the mpegpacketiser code.
Add parsing of picture types, that just feeds into a debug message
for now.
Fix some 64-bit format strings.

Changelog surgery

Original commit message from CVS:
Changelog surgery

gst/equalizer/gstiirequalizer10bands.c: A 10 band EQ should be initialized to 1 bands and not to 3.

Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_init):
A 10 band EQ should be initialized to 1 bands and not to 3.

Port mpeg1videoparse to 0.10 and give it rank SECONDARY-1, so that it's below existing decoders.

Original commit message from CVS:
* configure.ac:
* gst/mpeg1videoparse/Makefile.am:
* gst/mpeg1videoparse/gstmp1videoparse.c:
* gst/mpeg1videoparse/gstmp1videoparse.h:
* gst/mpeg1videoparse/mp1videoparse.vcproj:
* gst/mpegvideoparse/Makefile.am:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_packetiser_init),
(mpeg_packetiser_free), (mpeg_packetiser_add_buf),
(mpeg_packetiser_flush), (mpeg_find_start_code),
(get_next_free_block), (complete_current_block),
(append_to_current_block), (start_new_block), (handle_packet),
(collect_packets), (mpeg_packetiser_handle_eos),
(mpeg_packetiser_get_block), (mpeg_packetiser_next_block):
* gst/mpegvideoparse/mpegpacketiser.h:
* gst/mpegvideoparse/mpegvideoparse.c: (mpegvideoparse_get_type),
(gst_mpegvideoparse_base_init), (gst_mpegvideoparse_class_init),
(mpv_parse_reset), (gst_mpegvideoparse_init),
(gst_mpegvideoparse_dispose), (set_par_from_dar),
(set_fps_from_code), (mpegvideoparse_parse_seq),
(gst_mpegvideoparse_time_code), (gst_mpegvideoparse_flush),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (gst_mpegvideoparse_change_state),
(plugin_init):
* gst/mpegvideoparse/mpegvideoparse.h:
* gst/mpegvideoparse/mpegvideoparse.vcproj:
Port mpeg1videoparse to 0.10 and give it rank SECONDARY-1, so
that it's below existing decoders.
Rename it to mpegvideoparse to reflect that it handles MPEG-1 and
MPEG-2 now.
Re-write the parsing code so that it collects packets differently
and timestamps Picture packets correctly.
Add a list of FIXME's at the top.

gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...

Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.

gst/audioresample/gstaudioresample.c: Handle discontinuous streams.

Original commit message from CVS:
2007-03-14  Julien MOUTTE  <julien@moutte.net>

* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.

tests/icles/equalizer-test.c: Port the example to new equalizer api.

Original commit message from CVS:
* tests/icles/equalizer-test.c: (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):
Port the example to new equalizer api.

sys/osxvideo/: Fix leaks when running a NSApp.

Original commit message from CVS:
* sys/osxvideo/cocoawindow.h:
* sys/osxvideo/cocoawindow.m:
* sys/osxvideo/osxvideosink.h:
* sys/osxvideo/osxvideosink.m:
Fix leaks when running a NSApp.
Accept any kind of resolutions.
Works in fullscreen. Can maximize.
Only thing left before being able to move this to -good is documentation
and embedded window support.

po/: Updated translations.

Original commit message from CVS:
* po/hu.po:
* po/it.po:
* po/sv.po:
Updated translations.

gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...

Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample

gst/equalizer/: Add 3 and 10 band version and add missing gst_object_sync_values.

Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_class_init),
(gst_iir_equalizer_band_get_type),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_transform_ip), (plugin_init):
* gst/equalizer/gstiirequalizer10bands.c:
(gst_iir_equalizer_10bands_base_init),
(gst_iir_equalizer_10bands_class_init),
(gst_iir_equalizer_10bands_init),
(gst_iir_equalizer_10bands_set_property),
(gst_iir_equalizer_10bands_get_property):
* gst/equalizer/gstiirequalizer10bands.h:
* gst/equalizer/gstiirequalizer3bands.c:
(gst_iir_equalizer_3bands_base_init),
(gst_iir_equalizer_3bands_class_init),
(gst_iir_equalizer_3bands_init),
(gst_iir_equalizer_3bands_set_property),
(gst_iir_equalizer_3bands_get_property):
* gst/equalizer/gstiirequalizer3bands.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_init):
Add 3 and 10 band version and add missing gst_object_sync_values.
* gst/spectrum/gstspectrum.c: (gst_spectrum_event),
(gst_spectrum_transform_ip):
Add some comments about float support.

add debugging and reformat docs

Original commit message from CVS:
add debugging and reformat docs

gst/mpegaudioparse/: Remove bogus 2nd copy of mp3parse - it's actually in -ugly.

Original commit message from CVS:
* gst/mpegaudioparse/Makefile.am:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/mpegaudioparse/gstmpegaudioparse.h:
* gst/mpegaudioparse/mpegaudioparse.vcproj:
Remove bogus 2nd copy of mp3parse - it's actually
in -ugly.

examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.

Original commit message from CVS:
* examples/app/.cvsignore:
The buildbot demands .cvsignore files, and I comply.

sys/directdraw/gstdirectdrawsink.*: Handle display mode changes during playback.

Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directdraw/gstdirectdrawsink.h:
Handle display mode changes during playback.

Add appsrc/appsink example.

Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/app/Makefile.am:
* examples/app/appsrc_ex.c:
Add appsrc/appsink example.
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst/app/gstapp.c:
Add appsink.

ext/: Printf format string fixes.

Original commit message from CVS:
* ext/nas/nassink.c: (NAS_createFlow):
* ext/sndfile/gstsfsrc.c: (gst_sf_src_create):
Printf format string fixes.

gst/equalizer/: Refactor plugin into a base class and a first subclass (nband eq). The nband eq uses GstChildProxy an...

Original commit message from CVS:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (_do_init),
(gst_iir_equalizer_band_set_property),
(gst_iir_equalizer_band_get_property),
(gst_iir_equalizer_band_class_init),
(gst_iir_equalizer_band_get_type),
(gst_iir_equalizer_child_proxy_get_child_by_index),
(gst_iir_equalizer_child_proxy_get_children_count),
(gst_iir_equalizer_child_proxy_interface_init),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_finalize), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_setup), (plugin_init):
* gst/equalizer/gstiirequalizer.h:
* gst/equalizer/gstiirequalizernbands.c:
(gst_iir_equalizer_nbands_base_init),
(gst_iir_equalizer_nbands_class_init),
(gst_iir_equalizer_nbands_init),
(gst_iir_equalizer_nbands_set_property),
(gst_iir_equalizer_nbands_get_property):
* gst/equalizer/gstiirequalizernbands.h:
Refactor plugin into a base class and a first subclass (nband eq). The
nband eq uses GstChildProxy and is controlable. More subclasses will
follow.

ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele...

Original commit message from CVS:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.

ext/wavpack/: Use a general wavpack debug category for common code.

Original commit message from CVS:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.c:
Use a general wavpack debug category for common code.
* ext/wavpack/gstwavpackstreamreader.c:
(gst_wavpack_stream_reader_set_pos_abs),
(gst_wavpack_stream_reader_set_pos_rel),
(gst_wavpack_stream_reader_write_bytes):
Use the general wavpack debug category here too and add debug
output to the functions that should not be called at all by
the wavpack library.
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_plugin_init):
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_plugin_init):
Change debugging category names to conform to the conventions.

gst/qtdemux/qtdemux.*: Share qtdemux debug category across all files, otherwise all debugging in files other than qtd...

Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
* gst/qtdemux/qtdemux.h:
Share qtdemux debug category across all files, otherwise all debugging
in files other than qtdemux.c would end up in the default category.

gst/spectrum/gstspectrum.*: One FIXME less, by resolving message timestamps against the playback segment.

Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_event), (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
One FIXME less, by resolving message timestamps against the playback
segment.

gst/spectrum/gstspectrum.c: Fix and cleanup default property values.

Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init),
(gst_spectrum_init), (gst_spectrum_set_property),
(gst_spectrum_transform_ip):
Fix and cleanup default property values.
Add FIXMEs for stuff that looks rather wrong.

gst/spectrum/: Remove two obsolete and confusing comments.

Original commit message from CVS:
* gst/spectrum/demo-audiotest.c: (message_handler):
* gst/spectrum/demo-osssrc.c: (message_handler):
Remove two obsolete and confusing comments.

ext/nas/nassink.c: Some more cleanups/changes; use boilerplate macro.

Original commit message from CVS:
* ext/nas/nassink.c: (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_getcaps),
(gst_nas_sink_unprepare):
Some more cleanups/changes; use boilerplate macro.

ext/nas/: Bunch of nassink clean-ups: make build by adding the right CFLAGS and LIBS to Makefile.am; rename structure...

Original commit message from CVS:
* ext/nas/Makefile.am:
* ext/nas/README:
* ext/nas/nassink.c: (gst_nas_sink_get_type),
(gst_nas_sink_base_init), (gst_nas_sink_class_init),
(gst_nas_sink_init), (gst_nas_sink_finalize),
(gst_nas_sink_getcaps), (gst_nas_sink_prepare),
(gst_nas_sink_unprepare), (gst_nas_sink_delay),
(gst_nas_sink_reset), (gst_nas_sink_write),
(gst_nas_sink_set_property), (gst_nas_sink_get_property),
(gst_nas_sink_open), (gst_nas_sink_close), (NAS_flush),
(NAS_sendData), (NAS_EventHandler), (gst_nas_sink_sink_get_format),
(NAS_createFlow), (plugin_init):
* ext/nas/nassink.h:
Bunch of nassink clean-ups: make build by adding the right CFLAGS
and LIBS to Makefile.am; rename structure, macros and functions
according to canonical naming scheme; move some things around a bit;
use GST_CAT_DEFAULT instead of GST_CAT_* everywhere; remove README
file that didn't really contain any useful information anyway (the
useful bits have been moved into the 'host' property description).

ext/directfb/dfbvideosink.c: Chain up in finalize.

Original commit message from CVS:
* ext/directfb/dfbvideosink.c: (gst_dfbvideosink_finalize):
Chain up in finalize.

Fix up dist.

Original commit message from CVS:
* configure.ac:
* gst/vmnc/Makefile.am:
Fix up dist.

Add VMnc decoder.

Original commit message from CVS:
* configure.ac:
* gst/vmnc/Makefile.am:
* gst/vmnc/vmncdec.c: (gst_vmnc_dec_base_init),
(gst_vmnc_dec_class_init), (gst_vmnc_dec_init),
(gst_vmnc_dec_reset), (vmnc_handle_wmvi_rectangle),
(render_colour_cursor), (render_cursor), (vmnc_make_buffer),
(vmnc_handle_wmvd_rectangle), (vmnc_handle_wmve_rectangle),
(vmnc_handle_wmvf_rectangle), (vmnc_handle_wmvg_rectangle),
(vmnc_handle_wmvh_rectangle), (vmnc_handle_wmvj_rectangle),
(render_raw_tile), (render_subrect), (vmnc_handle_raw_rectangle),
(vmnc_handle_hextile_rectangle), (vmnc_handle_packet),
(vmnc_dec_setcaps), (vmnc_dec_chain), (vmnc_dec_change_state),
(vmnc_dec_set_property), (vmnc_dec_get_property), (plugin_init):
Add VMnc decoder.
Still missing support for:
- rectangle types I didn't find in my samples (e.g. copy, RRE,
ZRLE)
- alpha-composited cursors

gst-libs/gst/app/Makefile.am: Install the headers.

Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Install the headers.

gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.

Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.

gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.

Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Hacking to address issues in 413418.

Move the app library to gst-libs/gst/app (duh!)

Original commit message from CVS:
* Makefile.am:
* configure.ac:
* ext/Makefile.am:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Move the app library to gst-libs/gst/app (duh!)

Commit NAS Sink, closed bugzilla 345633

Original commit message from CVS:
Commit NAS Sink, closed bugzilla 345633

ext/dts/gstdtsdec.c: A few small clean-ups.

Original commit message from CVS:
* ext/dts/gstdtsdec.c: (gst_dtsdec_init), (gst_dtsdec_sink_event):
A few small clean-ups.
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
More debug output for failure cases.

ext/dts/gstdtsdec.c: Don't do forced downmixing to stereo, but check what downstream can do and let libdts do the dow...

Original commit message from CVS:
Patch by: Young-Ho Cha  <ganadist at chollian net>
* ext/dts/gstdtsdec.c: (gst_dtsdec_handle_frame),
(gst_dtsdec_change_state):
Don't do forced downmixing to stereo, but check what downstream
can do and let libdts do the downmixing based on that (#400555).

Remove spurious conflict marker

Original commit message from CVS:

Remove spurious conflict marker