configure.ac: releasing 0.10.15, "No need to argue"RELEASE-0_10_15

Original commit message from CVS:
=== release 0.10.15 ===

2007-11-15  Jan Schmidt <jan.schmidt@sun.com>

* configure.ac:
releasing 0.10.15, "No need to argue"

1 files changed, 84 insertions, 1 deletions

diff --git a/NEWS b/NEWS
index d2217a19d..ffac3bacc 100644
--- a/NEWS
+++ b/NEWS
@@ -1,10 +1,93 @@
-This is GStreamer Base Plug-ins 0.10.14, "Light Years Ahead"
+This is GStreamer Base Plug-ins 0.10.15, "No need to argue"
 
 Please note that decodebin2 API included in this release is still
 considered unstable and WILL change in future releases. At this stage, only
 developers or early adopters should consider using the decodebin2 API embodied
 in its signals and properties.
 
+Changes since 0.10.14:
+
+      * RTP/RTSP/RTCP/SDP support improved
+      * New FFT support library libgstfft, based on Kiss FFT
+      * New formats supported in volume and audiotestsrc
+      * Fixes in audiorate and videorate
+      * Audio capture fixes
+      * Playbin and decodebin fixes
+      * New tagdemux base class for ID3/APE style tag readers
+      * Fix a nasty crash in the X sinks on shutdown
+      * New tags supported
+      * Add support for multichannel WAV files.
+      * Preserve channel layout information when up/down-mixing.
+      * Many bug-fixes and improvements
+
+Bugs fixed since 0.10.14:
+
+      * 475395 : decodebin2 leaks request-pads
+      * 475451 : [decodebin2] leaks ghostpad
+      * 378770 : [xvimagesink] race condition in event thread?
+      * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
+      * 430677 : [audioconvert] does not preserve channel positions when f...
+      * 442654 : [volume] controller bypassed by default
+      * 445529 : [volume] support for 24/32-bit audio/x-raw-int
+      * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
+      * 451970 : Subparse requires HTML parser
+      * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
+      * 459334 : [textoverlay] expose pango line alignment property
+      * 459585 : [basertpdepayload] api without namespace
+      * 460422 : [audiotestsrc] Add support for float and double output
+      * 462805 : [alsa] compilation fails with gcc 4.2
+      * 462979 : Add 'silent' property to GstTimeOverlay
+      * 463215 : [audioconvert] compile errors
+      * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
+      * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
+      * 464690 : Add connection-speed property to uridecodebin element
+      * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
+      * 465028 : some warnings with mingw
+      * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
+      * 468129 : [basertpaudiopayload] event handler returns the wrong value
+      * 468619 : New library gstfft: FFT library for integer and float typ...
+      * 470456 : [API] add gst_missing_*_installer_detail_new()
+      * 470766 : [ssaparse] line breaks in SSA subtitle parser
+      * 471067 : Make the SDP code useable for generating SDP descriptions
+      * 471194 : [rtpbuffer] RTP headers are wrong for win32
+      * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
+      * 474384 : gstrtsp-enumtypes.c and .h needed for win32
+      * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
+      * 475731 : rtspconnection is able to read incomplete messages
+      * 483620 : All Rtp buffers are discarded --  gst_rtp_buffer_get_payl...
+      * 484989 : memleak, not unrefed caps for gstbasertppayload.c
+      * 489010 : Please change default channel order for WAVE_EXT-less .wa...
+      * 491722 : [playbin] regression: crash with external subtitles
+      * 492098 : [GstFFT] Broken scaling
+      * 492114 : Build issues on Windows/MSVC
+      * 492306 : compilation errors with MinGW
+      * 492813 : Missing symbols in libgstrtp.def
+      * 493986 : Build issues on Windows (missing symbols)
+      * 494346 : pre-release vs6 patch
+      * 496548 : Including malloc.h breaks macos build
+      * 496724 : DSW file references non-existent DSP files
+      * 464079 : audiotestsrc doesn't respond to conversion queries properly
+      * 442065 : floatcast.h includes config.h and might break other apps
+      * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
+      * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
+      * 464028 : Move connection-speed from playbin to playbasebin
+
+API added since 0.10.14:
+
+      * GstTagDemux base class for simple tag demuxers
+      * GstBaseAudioSrc::provide-clock property
+      * gst_rtcp_ntp_to_unix()
+      * gst_rtcp_unix_to_ntp()
+      * gst_rtp_buffer_get_header_len()
+      * gst_rtp_buffer_get_extension_data()
+      * gst_rtp_buffer_compare_seqnum()
+      * gst_rtp_buffer_ext_timestamp()
+      * gst_rtcp_packet_sdes_copy_entry()
+      * gst_install_plugins_supported()
+      * gst_missing_*_installer_detail_new() convenience API
+      * gst_rtsp_connection_poll()
+      * GstTextOverlay::line-alignment property
+
 Changes since 0.10.13:
 
       * Audio dither and noise-shaping when reducing bit-depth