File baresip.changes of Package baresip

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File baresip.changes of Package baresip (Revision c44df238937d2c3a2e518c4ba6d5e43b)

Currently displaying revision c44df238937d2c3a2e518c4ba6d5e43b , Show latest

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Thu Jan  5 22:30:53 UTC 2023 - Matthias Eliasson <elimat@opensuse.org>

- Update to release 2.10.0
  * sdl: small improvements
  * vidinfo: allow all pixel formats
  * vid: add support for YUV422P pixel format
  * avformat: fix hwaccel vaapi
  * mk: add deprecate notice
  * mingw: upgrade to OpenSSL 3.0.7
  * dshow: fix some warnings
  * dshow: fix pragma warning
  * ci: install libsdl2 development package
  * sdl: work in progress fixes for multi-threading
  * Stop segfaulting when no URI is passed to dial command
  * ice: local candidate policy config
  * auresamp: check handler arguments
  * fixes 2315 and GTK errors on quit
  * auresamp: avoid division
  * cmake: check for XShm.h
  * pulse_async: avoid integer overrun for timestamps in recorder
  * ua: use sdp connection data instead origin
  * rtpext: move from baresip to re
  * acc,stream: add rtcp_mux account param
  * video: video_update cleanup
  * aufile/src: add auframe support
  * ice/tmr_async_handler: fix possible segfault
  * webrtc: fix browser offer handling
  * Space at the beginning of sip: creates errors
  * opus_multistream: update mimetype to ad-hoc standard
  * webrtc: add offerer and recvonly options
  * test: replace RSA cert with EC cert
  * Add OPTIONS handling for webrtc demo
  * mk: remove rtpext.c from srcs.mk
  * ua: change refer log to info()
- switch to Cmake builds
- add devel and libbaresip2 sub-packages
- add pcp module
- remove sndio and v4l2 modules

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Thu Aug 25 23:07:39 UTC 2022 - Jan Engelhardt <jengelh@inai.de>

- Update to release 2.6.0
  * Removed deprecated modules: gsm, gst, gst_video
  * Added modules: pulse_async

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Tue Jun 28 21:15:18 UTC 2022 - Antoine Belvire <antoine.belvire@opensuse.org>

- Update to version 2.4.0:
  * No high level changelog provided, see packaged CHANGELOG.md for
    details.
- Changes from version 2.3.0, 2.0.2, 2.0.1, 2.0.0:
  * No high level changelog provided, see packaged CHANGELOG.md for
    details.
- Add build dependencies on:
  * aom: Enables module av1.
  * fdk-aac: Enables module aac.
  * gcc-c++, webrtc-audio-processing: Enables module webrtc_aec.
  * gstreamer-app-1.0: Enables module gst_video.
  * libmp3lame, libmpg123, twolame: Enables module mpa.
  * libmosquitto: Enables module mqtt.
  * portaudio: Enables module portaudio.
  * sndio: Enables module sndio.
- Drop build dependency on cairo: Module cairo has been removed.
- Enable extra module avfilter.

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Wed May 12 22:23:01 UTC 2021 - Ferdinand Thiessen <rpm@fthiessen.de>

- Update to version 1.1.0
  * Added DTMF tones for A B C D
  * Fix interaction between CLI menu and GTK menu
  * gtk: show call_peername in notify title
  * call: when video stream is disabled, stop also video display
  * dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048
  * avcodec: fix H.264 interop with Firefox
  * ice: fix segfault in connh
  * aac: handle missing bitrate parameter in SDP format
  * menu: fix segfault in /aubitrate command
  * ice: remove support for ICE-lite
  * opengles: remove module
  * A lot other bug fixes and enhancements

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Mon Jun 29 10:58:50 UTC 2020 - Jan Engelhardt <jengelh@inai.de>

- Avoid sentences in title lines.

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Wed Jun 24 19:53:09 UTC 2020 - Martin Hauke <mardnh@gmx.de>

- Specfile overhaul

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Fri Nov  5 00:00:00 UTC 2010 - Alfred E. Heggestad <aeh@db.org>

- Initial build.

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File baresip.changes of Package baresip (Revision c44df238937d2c3a2e518c4ba6d5e43b)

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