Release 1.7.1

1 files changed, 91 insertions, 15 deletions

diff --git a/RELEASE b/RELEASE
index d133af621..4cccb28f7 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,18 +1,17 @@
 
-Release notes for GStreamer Base Plugins 1.6.0
+Release notes for GStreamer Base Plugins 1.7.1
 
 
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
+The GStreamer team is pleased to announce the first release of the unstable
+1.7 release series. The 1.7 release series is adding new features on top of
+the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework. The unstable 1.7 release series
+will lead to the stable 1.8 release series in the next weeks. Any newly added
+API can still change until that point.
 
 
-This release has been in the works for more than a year and is packed with new
-features, bug fixes and other improvements.
-
-
-See
-http://gstreamer.freedesktop.org/releases/1.6/
-for the full list of changes.
+Binaries for Android, iOS, Mac OS X and Windows will be provided separately
+during the unstable 1.7 release series.
 
 
 
@@ -62,10 +61,43 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
 
 Bugs fixed in this release
      
-      * 752148 : Drawing from paths passed to cairo does not work with PANGOCAIRO_BACKEND=coretext
-      * 754344 : libs: build rtp after audio
-      * 754833 : dmabuf & fdmemory: fix allocator_alloc documentation
-      * 755392 : video: bugs with gst_video_frame_copy and videoconvert (with test scripts)
+      * 681447 : video overlay composition: fix video blending over transparent frame
+      * 705579 : Playbin prevents plugins requesting a GstContext to work properly
+      * 726117 : typefinding: issue in MPEG-TS detection logic for streams with Null Pids
+      * 726472 : rtpbasepayload: Implement video SDP attributes
+      * 727970 : videorate: remove dead code
+      * 730926 : tags: add GST_TAG_PRIVATE_DATA and expose ID3 private frame ( " PRIV " ) data
+      * 731791 : videometa: add GstVideoAffineTransformationMeta
+      * 738687 : midi: add alsamidisrc, an ALSA MIDI sequencer source
+      * 749596 : rtsp-over-http authentication failure
+      * 751470 : encodebin: Fix special case.
+      * 752651 : decodebin: segfault on setting to NULL
+      * 753852 : gstreamer: base: Fix memory leaks when context parse fails.
+      * 754054 : videorate: remove unnecessary break statement
+      * 754196 : audiodecoder-test: port to using GstHarness
+      * 754223 : audioencoder-tests: port to use GstHarness
+      * 754450 : audiotestsrc: remove frequency and channel number limit
+      * 755260 : decodebin: Fix a race condition accessing the decode_chain field.
+      * 755301 : audioconvert: Integer- > Float conversion creates values slightly smaller than -1.0
+      * 755440 : gst-play: Add keyboard shortcut '0' to seek to beginning
+      * 755482 : videotestsrc: Force alpha downstream if foreground color contains alpha
+      * 756804 : playsink: text_sink dynamic reconnection is not working
+      * 757008 : tests: typefindfunctions: Fix error leak
+      * 757068 : audio{filter,convert,resample}: Clip input buffers to the segment before handling them
+      * 757351 : audioconvert: Latest audioconvert outputs noise
+      * 757480 : Use GST_STIME_FORMAT and GST_STIME_ARGS with GstClockTimeDiff
+      * 757926 : pbutils:encoding-target: Fix string memory leak
+      * 757927 : tests:video: Fix overlay rectangle and buffer leak
+      * 757928 : audio-quantize: Fix dither_buffer memory leak
+      * 758235 : rtspconnection: add support for parsing custom headers
+      * 758744 : allocators: Add logging category for GstFdMemory
+      * 758911 : audiobasesink/src: send latency message on setcaps
+      * 758922 : rtspconnection should optionally make HTTP requests with abs_path instead of absoluteURI
+      * 759126 : appsrc: issues with duration query handling
+      * 759329 : convertframe: Support video crop when convert frame
+      * 759356 : encodebin: Implement an encoding profile serialization format
+      * 742875 : [API] new audiovisualizer base class
+      * 758754 : oggdemux: failing to play an Opus sample file
 
 ==== Download ====
 
@@ -102,6 +134,50 @@ subscribe to the gstreamer-devel list.
         
 Contributors to this release
     
-      * Aurélien Zanelli
+      * Andreas Frisch
+      * Antonio Ospite
+      * Arnaud Vrac
+      * Csaba Toth
+      * Edward Hervey
+      * Eunhae Choi
+      * Evan Callaway
+      * Guillaume Desmottes
+      * Havard Graff
+      * Jan Schmidt
+      * Joan Pau Beltran
+      * Julien Isorce
+      * Kazunori Kobayashi
+      * Koop Mast
+      * Luis de Bethencourt
+      * Mathieu Duponchelle
+      * Matthew Waters
+      * Michael Olbrich
+      * Miguel París Díaz
+      * Nicolas Dufresne
+      * Nirbheek Chauhan
+      * Ognyan Tonchev
+      * Pankaj Darak
+      * Pavel Bludov
+      * Perry Hung
+      * Philippe Normand
+      * Rajat Verma
+      * Ravi Kiran K N
+      * Reynaldo H. Verdejo Pinochet
       * Sebastian Dröge
+      * Sebastian Rasmussen
+      * Song Bing
+      * Stefan Sauer
+      * Stian Selnes
+      * Thiago Santos
+      * Thibault Saunier
+      * Thomas Bluemel
+      * Tim-Philipp Müller
+      * Vincent Penquerc'h
+      * Vineeth T M
+      * Vineeth TM
+      * Vivia Nikolaidou
+      * William Manley
+      * Wim Taymans
+      * Xavier Claessens
+      * eunhae choi
  
\ No newline at end of file