re/re.changes

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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

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Wed Jun 24 07:32:59 UTC 2020 - Martin Hauke <mardnh@gmx.de>

- Specfile cleanup

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

- Initial build