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File re.changes of Package re
------------------------------------------------------------------- Wed Dec 27 18:56:06 UTC 2023 - Martin Hauke <mardnh@gmx.de> - Update to version 3.8.0 Changes: * https://github.com/baresip/re/releases/tag/v3.8.0 * https://github.com/baresip/re/releases/tag/v3.7.0 * https://github.com/baresip/re/releases/tag/v3.6.2 * https://github.com/baresip/re/releases/tag/v3.6.1 * https://github.com/baresip/re/releases/tag/v3.6.0 * https://github.com/baresip/re/releases/tag/v3.5.1 * https://github.com/baresip/re/releases/tag/v3.5.0 ------------------------------------------------------------------- Fri Sep 8 18:11:26 UTC 2023 - Andreas Stieger <andreas.stieger@gmx.de> - Update to 3.4.0: * rtpext: uniform parameter name fixes doxygen warning * mk: add rem to doxygen inputs * vidmix: allow different pixel format * ajb doxygen * aes: correct parameters for stub * ci/build: fail on cmake and compile warnings * fmt: fix format string in fmt_timestamp() * hmac,md5,sha: add mbedtls backend * test: no need to rewind freshly allocated mbuf * httpauth: basic challenge creation and verification functions * Fix include of re_thread.h in re_tmr.h * btrace: fix WIN32_LEAN_AND_MEAN macro redefine * aumix: add record sum handler * ci/win: disable x86 testing * sipsess: allow UPDATE and INFO in early dialog * prefix macro VERSION * main: use HAVE_SIGNAL in init.c * test: change to ASSERT_XXX macros, remove EXPECT_XXX macros * fmt: handy functions for pointer-length objects * test: add TWCC test from Chrome 114 packet * sipsess/listen: Fix target_refresh_handler * ci/mingw: downgrade cmake * cmake: fix target include path for subdir projects ------------------------------------------------------------------- Fri Jul 28 07:52:39 UTC 2023 - Paolo Stivanin <info@paolostivanin.com> - Update to 3.3.0: Breaking changes: * librem is now merged with libre Changes: * https://github.com/baresip/re/releases/tag/v3.3.0 * https://github.com/baresip/re/releases/tag/v3.2.0 * https://github.com/baresip/re/releases/tag/v3.1.0 * https://github.com/baresip/re/releases/tag/v3.0.0 ------------------------------------------------------------------- Sun Dec 11 17:35:34 UTC 2022 - Andreas Stieger <andreas.stieger@gmx.de> - re 2.10.0: * h264: add STAP-A * h265: add missing NAL types * rtpext: move from baresip to re * dns: fix dnsc_conf_set memory leak * developer visible fixes ------------------------------------------------------------------- Sun Dec 4 19:57:12 UTC 2022 - Andreas Stieger <andreas.stieger@gmx.de> - re 2.9.0: * general maintenance and bugfix release ------------------------------------------------------------------- Sat Oct 1 08:43:14 UTC 2022 - Martin Hauke <mardnh@gmx.de> - Update to release 2.8.0 * No high level changelog provided, see packaged CHANGELOG.md for details. - Use CMake for the build ------------------------------------------------------------------- Thu Aug 25 22:26:54 UTC 2022 - Jan Engelhardt <jengelh@inai.de> - Update to release 2.6.0 * sip: add RFC 3262, 3311 support * bfcp: Add support for TCP transport ------------------------------------------------------------------- Tue Jun 28 21:23:08 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. ------------------------------------------------------------------- Sat May 21 10:55:44 UTC 2022 - Andreas Stieger <andreas.stieger@gmx.de> - update to 2.3.0: * network improvements * static code analysis fixes * aubuf adaptive jitter buffer * Support adding CRLs * shim: new module * new Trice module * error corrections and developer visible fixes * ToS for video and sip ------------------------------------------------------------------- Sat Apr 24 10:08:48 UTC 2021 - Martin Hauke <mardnh@gmx.de> - Update to version 2.0.1 Added * aac: add AAC_STREAMTYPE_AUDIO enum value * aac: add AAC_ prefix * Video mode param to call_answer(), ua_answer() and ua_hold_answer * video_stop_display() API function * module: add path to module_load() function * conf: add conf_configure_buf * test: add usage of g711.so module * JSON initial codec state command and response * account_set_video_codecs() API function * net: add fallback dns nameserver * gtk: show call_peername in notify title * call: Added call_state() API function that returns enum state of the call * account_set_stun_user() and account_set_stun_pass() API functions. * API functions account_stun_uri and account_set_stun_uri. * ausine: Audio sine wave input module * gtk/menu: replace spaces from uri * jack: allowing jack client name to be specified in the config file * snapshot: Add snapshot_send and snapshot_recv commands * webrtc_aec: 'extended_filter' config option * avfilter: FFmpeg filter graphs integration * reg: view proxy expiry value in reg_status * account: add parameter rwait for re-register interval * call, stream, menu: add cmd to set the direction of video stream * Added AMRWBENC_PATH env var to amr module module.mk Changed * Using baresip/re fork now * audio: move calculation to audio_jb_current_value * avformat: clean up docs * gzrtp: update docs * account: increased size of audio codec list to 16 * video: make video_sdp_attr_decode public * config: Derive default audio driver from default audio device * jack: modifying info message on jack client creation * call: when video stream is disabled, stop also video display * dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048 * rst: use a min ptime of 20ms * aac: change ptime to 4ms Fixed * avcodec: fix H.264 interop with Firefox * avcodec: call av_hwdevice_ctx_create before if-statement * account: use single quote instead of backtick * ice: fix segfault in connh #980 * call: Update call->got_offer when re-INVITE or answer to re-INVITE is received * config: Allow distribution specific CA trust bundle locations * config: Allow distribution specific default audio device * mqtt: fix err is never read (found by clang static analyzer) * avcodec: fix err is never read (found by clang static analyzer) * gtk: notification buttons do not work on Systems #1012 * gtk: fix dtmf_tone and add tones as feedback #1010 * pulse: drain pulse buffers before freeing #1016 * jack: jack_play connect all physical ports #1028 * Makefile: do not try to install modules if build is static * gzrtp: media_alloc function is missing #1034 #1022 * call: when updating video, check if video stream has been disabled #1037 * amr: fix length check, fixes #1011 * modules: fix search path for avdevice.h #1043 * gtk: declare variables C89 style * config: init newly added member * menu: fix segfault in ua_event_handler #1059 #1061 * debug_cmd: fix OpenSSL no-deprecated #1065 * aac: handle missing bitrate parameter in SDP format * av1: properly configure encoder * call: When terminating outgoing call, terminate also possible refer subscription #1082 * menu: fix segfault in /aubitrate command * amr: should check if file (instead of directory) exists Removed * ice: remove support for ICE-lite * ice: remove ice_debug, use log level DEBUG instead * ice: make stun server optional * config: remove ice_debug option (unused) * opengles: remove module (not working) #1079 ------------------------------------------------------------------- Wed Jun 24 07:32:59 UTC 2020 - Martin Hauke <mardnh@gmx.de> - Specfile cleanup ------------------------------------------------------------------- Fri Nov 5 00:00:00 UTC 2010 - Alfred E. Heggestad <aeh@db.org> - Initial build
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