=== release 1.12.2 === 2017-07-14 Sebastian Dröge * configure.ac: releasing 1.12.2 2017-07-14 13:22:45 +0300 Sebastian Dröge * po/el.po: po: Update translations 2017-07-13 12:47:02 +0300 Sebastian Dröge * gst/isomp4/qtdemux.c: qtdemux: Fix parsing of RLE depth Regression introduced by 86b427dc70562f891a551ffc9f96cefe1cafcddd https://bugzilla.gnome.org/show_bug.cgi?id=784812 2017-05-20 17:09:52 +0200 Josep Torra * sys/osxaudio/gstosxcoreaudio.c: osxaudio: fixes playback of mono streams with no channel-mask field in caps Fixes a negotiation error seen when trying to playback of a .MOV file with a mono AAC audio stream decoded by avcdec_aac that doesn't set channel-mask field but sink was requiring channel-mask=0x3. 2017-07-07 21:15:57 +0900 Yasushi SHOJI * gst/rtp/gstrtpgsmpay.c: rtpgsmpay: fix accidental garbage data before actual payload Do not allocate payload size outbuf if appending payload buffer. The commit 137672ff1824948bda4b1b1967de8c24a0055b67 attached payload to the output buffer but forgot to remove payload allocation. That effectively doubled payload size and add zero'ed or random bytes. Makes the following pipeline work again: gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink https://bugzilla.gnome.org/show_bug.cgi?id=784616 2017-07-03 11:47:13 -0400 Nicolas Dufresne * gst/rtpmanager/gstrtprtxreceive.c: rtprtxreceive: Add memory and boundary checks This element was not checking if mapping the RTP buffer and the payload worked, and was not checking if the RTX payload was large enough. https://bugzilla.gnome.org/show_bug.cgi?id=784484 2017-07-03 20:27:29 +0100 Tim-Philipp Müller * gst/imagefreeze/gstimagefreeze.c: imagefreeze: fix use-after-free on seek event Get seqnum before unreffing the seek event. https://bugzilla.gnome.org/show_bug.cgi?id=784486 2017-06-29 18:59:58 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Create send/recv mutexes once, not on every connect() Also fixes a crash caused by freeing an uninitialized mutex in an error case. https://bugzilla.gnome.org//show_bug.cgi?id=784282 2017-06-22 11:38:56 +0300 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Actually use the receive lock when receiving, not the send lock