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Import baresip-0.4.11 as wip/baresip. Baresip is a portable and modular SIP User-Agent with audio and video support. Features * Audio codecs: AMR, BV32, Celt, G.711, G.722, G.722.1, G.726, GSM, iLBC, iSAC, L16, OPUS, Silk, Speex * Video codecs: H.263, H.264, MPEG4, VP8 * Audio drivers: Alsa, Coreaudio, Gstreamer, Symbian MDA, OpenSLES, OSS, Portaudio, Windows wave * Video sources: FFmpeg avformat, MacOSX qtcapture, MacOSX quicktime, Video4Linux and Video4Linux2, X11 Grabber * Video output modules: OpenGL, SDL/SDL2, X11, DirectFB * NAT Traversal modules: STUN, TURN, ICE, NAT-PMP * Media encryption modules: SRTP, DTLS-SRTP, ZRTP TODO: Builds and packages, but doesn't run: baresip v0.4.11 Copyright (C) 2010 - 2014 Alfred E. Heggestad et al. config: creating config template /home/wiz/.baresip/config /usr/pkg/lib/baresip/modules: detected 21 modules _res is not supported for multi-threaded programs. zsh: abort (core dumped) baresip
2014-08-10 00:26:01 +00:00
$NetBSD: distinfo,v 1.1 2014/08/10 00:26:01 thomasklausner Exp $
baresip: update to baresip-1.0.0 (Presumably the changelog [Unreleased] section is for 1.0.0) = Baresip Changelog == [Unreleased] === Added - aac: add AAC_STREAMTYPE_AUDIO enum value - aac: add AAC_ prefix - Video mode param to call_answer(), ua_answer() and ua_hold_answer [#966] - video_stop_display() API function [#977] - module: add path to module_load() function - conf: add conf_configure_buf - test: add usage of g711.so module [#978] - JSON initial codec state command and response [#973] - account_set_video_codecs() API function [#981] - net: add fallback dns nameserver [#996] - gtk: show call_peername in notify title [#1006] - call: Added call_state() API function that returns enum state of the call [#1013] - account_set_stun_user() and account_set_stun_pass() API functions [#1015] - API functions account_stun_uri and account_set_stun_uri. [#1018] - ausine: Audio sine wave input module [#1021] - gtk/menu: replace spaces from uri [#1007] - jack: allowing jack client name to be specified in the config file [#1025] [#1020] - snapshot: Add snapshot_send and snapshot_recv commands [#1029] - webrtc_aec: 'extended_filter' config option [#1030] - avfilter: FFmpeg filter graphs integration [#1038] - reg: view proxy expiry value in reg_status [#1068] - account: add parameter rwait for re-register interval [#1069] - call, stream, menu: add cmd to set the direction of video stream [#1073] === Changed - **Using [baresip/re](https://github.com/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 [#1009] - jack: modifying info message on jack client creation [#1019] - call: when video stream is disabled, stop also video display [#1023] - dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048 [#1062] [#1056] - rst: use a min ptime of 20ms - aac: change ptime to 4ms === Fixed - avcodec: fix H.264 interop with Firefox - winwave: waveInGetPosition is no longer supported for use as of Windows Vista [#960] - 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 [#986] - mk: Test also for /usr/lib64/libspeexdsp.so to cover Fedora/RHEL/CentOS [#992] - config: Allow distribution specific CA trust bundle locations (fixes [#993] - config: Allow distribution specific default audio device (fixes [#994] - 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 [#1031] - 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 === 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) === Contributors (many thanks) - Alfred E. Heggestad - Alexander Gramner - Andrew Webster - Christian Spielberger - Christoph Huber - Davide Alberani - Ethan Funk - Juha Heinanen - mbattista - Michael Malone - Mikl Kurkov - ndilieto - Robert Scheck - Roger Sandholm - Sebastian Reimers [#966]: https://github.com/baresip/baresip/pull/966 [#977]: https://github.com/baresip/baresip/pull/977 [#978]: https://github.com/baresip/baresip/pull/978 [#973]: https://github.com/baresip/baresip/pull/973 [#981]: https://github.com/baresip/baresip/pull/981 [#996]: https://github.com/baresip/baresip/pull/996 [#1006]: https://github.com/baresip/baresip/pull/1006 [#1013]: https://github.com/baresip/baresip/pull/1013 [#1015]: https://github.com/baresip/baresip/pull/1015 [#1018]: https://github.com/baresip/baresip/pull/1018 [#1021]: https://github.com/baresip/baresip/pull/1021 [#1007]: https://github.com/baresip/baresip/pull/1007 [#1025]: https://github.com/baresip/baresip/pull/1025 [#1020]: https://github.com/baresip/baresip/pull/1020 [#1029]: https://github.com/baresip/baresip/pull/1029 [#1030]: https://github.com/baresip/baresip/pull/1030 [#1038]: https://github.com/baresip/baresip/pull/1038 [#1009]: https://github.com/baresip/baresip/pull/1009 [#1019]: https://github.com/baresip/baresip/pull/1019 [#1023]: https://github.com/baresip/baresip/pull/1023 [#1062]: https://github.com/baresip/baresip/pull/1062 [#1056]: https://github.com/baresip/baresip/pull/1056 [#960]: https://github.com/baresip/baresip/pull/960 [#980]: https://github.com/baresip/baresip/pull/980 [#986]: https://github.com/baresip/baresip/pull/986 [#992]: https://github.com/baresip/baresip/pull/992 [#993]: https://github.com/baresip/baresip/pull/993 [#994]: https://github.com/baresip/baresip/pull/994 [#1012]: https://github.com/baresip/baresip/pull/1012 [#1010]: https://github.com/baresip/baresip/pull/1010 [#1016]: https://github.com/baresip/baresip/pull/1016 [#1028]: https://github.com/baresip/baresip/pull/1028 [#1031]: https://github.com/baresip/baresip/pull/1031 [#1034]: https://github.com/baresip/baresip/pull/1034 [#1022]: https://github.com/baresip/baresip/pull/1022 [#1037]: https://github.com/baresip/baresip/pull/1037 [#1011]: https://github.com/baresip/baresip/pull/1011 [#1043]: https://github.com/baresip/baresip/pull/1043 [#1059]: https://github.com/baresip/baresip/pull/1059 [#1061]: https://github.com/baresip/baresip/pull/1061 [#1065]: https://github.com/baresip/baresip/pull/1065 [#1068]: https://github.com/baresip/baresip/pull/1068 [#1069]: https://github.com/baresip/baresip/pull/1069 [#1073]: https://github.com/baresip/baresip/pull/1073 [Unreleased]: https://github.com/baresip/baresip/compare/v0.6.6...HEAD
2020-11-28 16:16:20 +00:00
SHA1 (baresip-1.0.0.tar.gz) = 575024228abf9527da8e21dbf36142043935379d
RMD160 (baresip-1.0.0.tar.gz) = fe40d85cf978e9c17e04dde0dfd23e78f0e63018
SHA512 (baresip-1.0.0.tar.gz) = 798ea9cd9892fe72e20d50c9a3b45b6b4c195da6d2f4052977276a7ae5ef90ac4f0c2bd6c5e48f53a4370a23f0e005dd73d1f46571965fc1bde5014b97bdc258
Size (baresip-1.0.0.tar.gz) = 989461 bytes
baresip: update to baresip-0.6.3 2019-06-22 Alfred E. Heggestad * Version 0.6.3 Alfred E. Heggestad (99): baresip: remove prefer_ipv6 from api, use config instead ua: remove prefer_ipv6 from api, use config instead audio: allocate mbuf for encoded telephony events net: remove af from api, use config instead gst: remove old module, use gst1 instead gst_video: remove old module, use gst_video1 instead gst1: update comment httpd: update comment call: remove unused constant reg: print address family of registration ua: clean up prefer_ipv6 code test: disable test for AUDIO_MODE_THREAD config: remove old check for rtcp_enable config: remove unused macro SA_INIT config: remove unused MOD_PRE test: mock aucodec support all sample formats audio: check that ptime is within the range of 1-60ms audio: dont check sample format for packetloss handler audio: use audio codec srate directly, remove get_srate wrapper audio: remove get_framesize audio: handle rtcp sample-rate for asymmetric codecs audio: remove get_ch() mpa: return posix error code instead of -1 mk: sort list of files in alphabetical order menu: sort and align incall commands table ua: check input argument to ua_print_supported test: check error from test fixture ua: use a print handler to print allowed methods ua: use a single tick instead of backtick for logging audio: mirror ptime attribute if changed by peer (ref #688) (#700) audio: receive ptime is always set plc: count samples from audio input use sizeof(x) instead of sizeof x account: fix typo timestamp: add timestamp_calc_seconds() call: remove const from menc_event_handler call: swap order of menc event and error handler mnat: make struct mnat public mnat: change to a simpler register api menc: protocol is always UDP mnat: change api to always use UDP protocol stream: add remote RTP/RTCP address to object menc: add remote RTP/RTCP address to API dtls_srtp: use remote address from mediaenc API dtls_srtp: remove dtls_print_sha1_fingerprint remove audio/video codec cycle net: add network_if_getname() sdp: remove sdp_media_format_cycle (unused) sdp: remove sdp_rattr() -- unused pcp: updated MNAT api gzrtp: updated menc api (ref #713) dtls_srtp: fix warning zrtp: fix warnings net: use network_if_getname to get interface name stream: use enum media_type instead of a string call: only include aucodec codecs in remote sdp (ref #718) call: simplify audio encoder/decodet set stream: add pointer to medianat module webrtc_aec: add warning ua: use KEYCODE_REL in dtmf handler (ref #719) call: add prefix to logline webrtc_aec: add sample format converter to decoder (ref #712) webrtc_aec: add sample format converter to encoder (ref #712) webrtc_aec: add enc/dec to log line webrtc_aec: fix enum warning webrtc_aec: echo_cancellation.h is included in aec.h (ref #712) config: add sip_cafile to template net: add a function to print IP-addr and interface net: dont init local address to 127.0.0.1 audio: handle marker bit in stream.c (#724) avcodec: make sure ffmpeg input buffer has AV_INPUT_BUFFER_PADDING_SIZE space at the end stream: update doxygen comments stream: only flush jitter-buffer if it was started avcodec: fallback define for AV_INPUT_BUFFER_PADDING_SIZE stream: add pseq_set flag stream: dont calculate loss if no jitter buffer webrtc_aec: add support for 32000Hz samplerate net: multiple nameservers in net_use_nameserver() webrtc_aec: add reference to webrtc native dtmfio: use UA_EVENT_CALL_DTMF_START to handle dtmf events test: use event handler to receive DTMF events webrtc_aec: remove samplerate check prepare for 0.6.3 release gst_video1: cleanup stream: print mediaenc id config: add net prefix to prefer_ipv6 codec2: print mode codec2: modern init of struct aucodec codec2: add config param codec2_mode codec2: round up bytes per frame win32: add httpd module to static.c codec2: update description mk: add detection of codec2.so module video: check if frame pointer is valid contact: set err properly audio: no need to clear err, it is not used config: add opus_samplerate to template travis: add building of codec2 on OSX (#736) config: add webrtc_aec to template Christian Spielberger (2): call: reset streams on call hold (#707) Bugfix/flush buffers on call hold (#716) Dmitry (2): opus: fixed opus_inbandfec param name in config and examples (#704) menu: set default values for optional config params (#705) Juha Heinanen (1): webrtc_aec: enable delay-agnostic echo cancellation Nicolas Tizon (1): audio: increase buffer size for audio device string (#710) juha-h (5): - added prefer_ipv6 config variable (#692) webrtc_aec module: added pthread.h include to .cpp files (#714) - Updated ilbc module encode/decode/pkloss function arguments (#723) - Use opus in mono mode (opus/48000/1) if opus_stereo or (#730) - Added opus_samplerate config parameter. (#733) premultiply (1): Interop: Parameters reordering and whitespace removal (#698) weili-jiang (1): Count ua references prior to destroy in case UA_EVENT_SHUTDOWN causes references to be removed (#702)
2019-06-22 15:51:41 +00:00
SHA1 (patch-modules_ilbc_ilbc.c) = a2a7d685c4989bf910a9d5b8582d1261fce32e1c
baresip: update to baresip-0.4.20 Pkgsrc changes: * change the build procedure a bit to accommodate modules * use the options framework top optionally build a number of modules * install example configuration files Notes: On my NetBSD machine the "oss" and "portaudio" modules do not work correctly. This is probably due to the fact that the module opens and initializes the device twice (once for reading and once for writing) but I was unable to diagnose the exact problem. A workaround is to use the jack plugin (or pulseaudio) by enabling the "jack.so" module in ~/.baresip/config. The jack server can start automatically: echo "/usr/pkg/bin/jackd -T -r -d sun" > ~/.jackdrc export JACK_START_SERVER=yes The sndiod module currently does not build. Changelog: 2016-07-22 Alfred E. Heggestad <aeh@db.org> * Version 0.4.20 * GIT URL: https://github.com/alfredh/baresip.git * GIT tag: v0.4.20 * NOTE: Requires libre v0.4.17 or later Requires librem v0.4.7 or later * new modules: - pulse Pulseaudio driver - vp9 VP9 video codec * config: - audio_path Path to audio files - call_local_timeout Timeout for incoming calls - redial_attempts Number of redial attempts - redial_delay Redial delay in seconds * baresip-core: - baresip: added a global baresip instance (WIP) - call: add RTP timeout (thanks to Sveriges Radio) - config: added call_local_timeout for incoming call timeout - config: added compile-time configureable CONFIG_PATH - config: added 'audio_path' config variable (thanks Juha Heinanen) - net: made it re-entrant with struct network - ua: added uag_set_exit_handler - ua: fix bug with reg_uri limited to 64-chars - video: vidfilters should not modify decoded image * selftest: - add test for network - add test for sending SIP OPTIONS - add test for RTP timeout * Modules: * avcodec: fix usage of deprecated API * avformat: remove support for scaling fix usage of deprecated API * cons: relay log-messages to active UDP/TCP connections https://github.com/alfredh/baresip/issues/144 * h265: fix usage of deprecated API * menu: added support for re-dial on failure (thanks to Sveriges Radio) * mpa: Bug with reinit of codec structs (thanks Christian Hoene) * natpmp: added support for RTCP * presence: use correct struct in deref handler * pulse: new module for Pulseaudio driver (thanks to Matthias Apitz for testing) * vidloop: vidfilters should not modify decoded image * vp8: module renamed from vpx.so to vp8.so * vp9: new module implementing VP9 video codec * wincons: use ReadConsoleInput, thanks to GGGO (fixes #139) https://github.com/alfredh/baresip/issues/139 2016-05-20 Alfred E. Heggestad <aeh@db.org> * Version 0.4.19 * GIT URL: https://github.com/alfredh/baresip.git * GIT tag: v0.4.19 * NOTE: Requires libre v0.4.14 or later Requires librem v0.4.7 or later * new modules: - mpa MPA Speech and Audio Codec (thanks Christian Hoene) * baresip-core: - audio: remove is_g722 exception use aucodec's rtp clockrate for calculating RTP timestamp plc: make sure sampc is exactly one ptime frame - aucodec: split srate into DSP srate and RTP clockrate (these are different for e.g. G.722 and MDA) - mos: add mos_calculate() (thanks Lorenzo Mangani) - net: use configured dns servers only, if specified - ua: fix potential NULL-pointer crash for uag.cfg * selftest: - add test for SIP registration with DNS - add test for SIP registration with authentication - add test for MOS calculations - added a mock DNS Server - added a mock SIP Server * Modules: * aucodec: add support for NV12 and YUVJ420P pixel formats * daala: update to libdaala version 0.0-1564-g79787c7 * gtk: fix autodetection of libgtk+ 2.0 (thanks Charles Lehner) * h265: remove call to x265_cleanup, caused crash on OpenBSD * mpa: new module that implements MPA Speech and Audio Codec (this module was contributed by Christian Hoene) * opus: added new configuration parameters: opus_cbr {yes,no} # Constant Bitrate (inverse of VBR) opus_inbandfec {yes,no} # Enable inband FEC opus_dtx {yes,no} # Enable DTX * presence: improved interoperability, allow white space before xml element closing tags (thanks Juha Heinanen) * x11: added borderless window (thanks Doug Blewett) 2016-03-12 Alfred E. Heggestad <aeh@db.org> * Version 0.4.18 * GIT URL: https://github.com/alfredh/baresip.git * GIT tag: v0.4.18 * NOTE: Requires libre v0.4.14 or later Requires librem v0.4.7 or later * baresip-core: - call: fix SIP INFO with dtmf-relay (thanks Gary Metalle) - ua: add event UA_EVENT_CALL_CLOSED for ua_hangup() * selftest: - add tests for answer a call and hangup * Modules: * alsa: fix potential crash (thanks Gary Metalle) * audiounit: fix compilation for iOS (issue #91) * avcodec: fix compilation for FFmpeg 3.0 * avformat: fix compilation for FFmpeg 3.0 * gtk: always handle incoming calls (thanks Charles Lehner) * h265: fix compilation for FFmpeg 3.0 * menu: add config 'menu_bell off/on' to enable Bell alert add command 'A' for switch audio device (thanks AlexMarlo) * v4l2_codec: add list of encoders (fixes #99) 2016-01-17 Alfred E. Heggestad <aeh@db.org> * Version 0.4.17 * GIT URL: https://github.com/alfredh/baresip.git * GIT tag: v0.4.17 * NOTE: Requires libre v0.4.14 or later Requires librem v0.4.7 or later * new modules: - echo Echo server module - jack JACK Audio Connection Kit audio-driver * baresip-core: - config: keep config object in memory - ua: moved playing of ringtones out of core, to "menu" module (let's keep the core nice and slim..) - ui: added ui_password_prompt() * selftest: - silence debug/info log by default, only print warnings (use -v to see verbose logging) * Modules: * alsa: added config option to specify the sample format "alsa_sample_format {s16,float,s24_3le}" thanks to Ola Palm for valuable feedback * audiounit: fix recording on OSX (thanks Sebastian Reimers) print hardware samplerate in debug mode * auloop: add support for 44100 Hz samplerate * daala: update to latest libdaala API (thanks Dmitrij D. Czarkoff) * echo: new module which implements a simple Echo-server, to be used in combination with the aubridge.so module. contributed by Sebastian Reimers * gtk: fixes to support C89 compiler (thanks Dmitrij D. Czarkoff) * jack: new module which implements audio-driver for JACK * menu: playing of ringtones moved here, from ua.c * sndio: fix crash when device open fails (thanks Dmitrij D. Czarkoff) 2015-12-01 Alfred E. Heggestad <aeh@db.org> * Version 0.4.16 * GIT URL: https://github.com/alfredh/baresip.git * GIT commit bed2241da3261e472f09b21958f0cc1324a94f27 * GIT tag: v0.4.16 * NOTE: Requires libre v0.4.14 or later * new modules: - v4l2_codec Video4Linux2 video codec (H264 hardware encoding) - vidinfo Video info overlay module * baresip-core: - audio: add audio_set_source() and audio_set_player() - audio: flush tx-buffer for all modes (thanks Thibault Gueslin) - call: add call_is_outgoing() - call: check address-family of incoming SDP offer (thanks Olle) - h264: move H.264 packetization code to core - main: add -u option to append extra global UA parameters - main: pre-load modules after all arguments are parsed - ua: add events UA_EVENT_SHUTDOWN,UA_EXIT - ua: add ua_hold_answer() - ua: add ua_set_media_af() - ua: delay mod-unloading if mods has a ref to struct ua * build: - add verbose build with V=1 (thanks Dmitrij D. Czarkoff) - add pkg-config file (thanks William King) - add travis.yml file for Github build-system * Modules: * alsa: fix memory leaks * avcodec: move common H.264 packetization code to core * cairo: use pkg-config in makefile * daala: update to latest libdaala (thanks Dmitrij D. Czarkoff) * gst_video: use H.264 packetization API from core * gst_video1: use H.264 packetization API from core * gtk: fix segmentation fault on window close * mwi: add 500ms delay after closing subscription * oss: use pthread for ausrc instead of fd_listen (fixes FreeBSD) * presence: use sipevent_sock instance from UA core add 500ms delay after closing subscription * v4l2_codec: new module * vidinfo: new module * zrtp: fix ZRTP over TURN by moving helper to layer 10 fix ZID verification (thanks Ingo Feinerer) 2015-09-26 Alfred E. Heggestad <aeh@db.org> * Version 0.4.15 * GIT URL: https://github.com/alfredh/baresip.git * GIT commit 86262a6fc17e19e2be82eb8a2a05ec0f884d3d38 * GIT tag: v0.4.15 * NOTE: Requires libre v0.4.13 or later * added selftest binary * baresip-core: - audio: fix televent when pt != 101 (reported by AndyJRobinson) - magic: use __func__ for C99 or later - sip: make sip_req_send() public - ua: add UA_EVENT_CALL_DTMF_START/END, thanks Gary Metalle * Modules: * alsa: added extra logging * gtk: add support for libnotify (thanks Charles Lehner) * video: fix potential null deref (thanks Tomasz Ostrowski) * zrtp: added 36-bytes preamble for TURN-header 2015-08-08 Alfred E. Heggestad <aeh@db.org> * Version 0.4.14 * GIT URL: https://github.com/alfredh/baresip.git * GIT commit ebac23b0692de71ee4c3a436f0372013150c937f * GIT tag: v0.4.14 * NOTE: Requires libre v0.4.13 or later * new modules: - gtk GTK+ 2.0 UI (thanks Charles E. Lehner) - gst1 Gstreamer 1.0 audio module - gst_video1 Gstreamer 1.0 video module (thanks Thomas Strobel) - daala Experimental video-codec using Daala * baresip-core: - baresip: added -m argument to pre-load modules - config: add kqueue to sample config (thanks Dmitrij D. Czarkoff) - log: make code C89 compliant (thanks Victor Sergienko) - module: added module_preload() - ua: add CALL_EVENT_TRANSFER_FAILED - ua: skip initial white space from uri (thanks Juha Heinanen) - ua: ua_prev_call() - videnc: move videnc_packet_h to update-handler * build: - added optional $(MOD)_CFLAGS for local module CFLAGS - added project file for Visual C++ Express 2010 - freebsd: add include path to $(SYSROOT)/local/include (thanks Hellmuth Michaelis) * Modules: * avcodec: make code C89 compliant (thanks Victor Sergienko) * cons: make code C89 compliant (thanks Victor Sergienko) * daala: new module * dshow: updates for VC2010 (thanks Victor Sergienko) * gst1: new module * gst_video1: new module * gtk: new module * menu: fix crash when 0 UAs (thanks Hans Petter Selasky) added command 'H' to hold previous call (thanks xanm) * wincons: make code C89 compliant (thanks ggcoding) 2015-06-20 Alfred E. Heggestad <aeh@db.org> * Version 0.4.13 * GIT commit 2e3e825ef5532dfde5a8b52de9ebaac51aa20a9c * NOTE: Requires libre v0.4.12 or later * new modules: - aufile Audio module for using a WAV-file as audio input - b2bua Back-to-Back User-Agent (B2BUA) module - codec2 CODEC2 audio codec - gst_video Gstreamer video codec - h265 H.265 (HEVC) video codec * baresip-core: - contact: add support for access-control (thanks Doug Blewett) - ausrc: change base-class to a const pointer - auplay: change base-class to a const pointer - vidsrc: change base-class to a const pointer - vidisp: change base-class to a const pointer - video: smooth sending of video packets * Modules: * amr: added support for octet-align mode (thanks to Stefan Sayer) * aubridge: copy audio-samples if resampler not needed * aufile: new module for using a WAV-file as audio source * avcapture: only register 1 video source * avformat: fix segfault on recent versions of libav * b2bua: new experimental module * codec2: new module for CODEC2 audio codec * dtls_srtp: uppercase fingerprint, interop (thanks Juha Heinanen) alternative SDP protocols for interop * dtmfio: unregister event handler on close (thanks Hellmuth Michaelis) * gst_video: new module using Gstreamer as a video codec (Thanks to Victor Sergienko and Fadeev Alexander) * h265: new module for H.265 video codec * httpd: added raw mode (thanks Lorenzo Mangani) * menu: create user-agent with a command 'R' (thanks Lorenzo Mangani) * opus: add configuration of "opus_bitrate" (thanks to Juha Heinanen) * speex: add configuration of "speex_mode_nb" and "speex_mode_wb" (thanks to Dmitrij D. Czarkoff and Juha Heinanen) * vidloop: add VIDLOOP_INTERNAL_FMT and split encoder/decoder * x11: catch Window delete (thanks to Doug Blewett) * zrtp: initialize remote_zid (thanks to Ingo Feinerer) 2014-12-24 Alfred E. Heggestad <aeh@db.org> * Version 0.4.12 * GIT commit 67993e35d980375458348b264c4a35a944bb5180 * NOTE: Requires libre v0.4.11 or later * baresip: - account: add regint and pubint - audio: fix checking of sample-rate range - config: remove the "input" block - config: added support for quoted device parameters - config: fix conversion of bandwidth to kbit/s - config: generate more relevant config for FreeBSD and OpenBSD (thanks Dmitrij D. Czarkoff) - reg: add support for extracting GRUU parameter - main: add -p option to set path to audio files - sipreq: make response-handler optional - ua: add support for GRUU (RFC 5627) (many thanks to Juha Heinanen for starting this work and helping out with the testing) - ua: moved presence-status to each struct ua instance - ua: add presence status to each User-Agent instance - ua: use public-GRUU if set, otherwise local cuser - ui: make UI single instance - video: add VIDENC_INTERNAL_FMT (suggested by Victor Sergienko) * docs: added sample configuration files * account: added pubint for Publishing Interval * avcodec: upgrade to recent ffmpeg/libav APIs either FFmpeg or libav can be used * celt: deleted module (replaced by opus) * cons: update usage of struct ui, added output handler added config: cons_listen 0.0.0.0:5555 * evdev: update usage of struct ui, added output handler added config: evdev_device /dev/input/event0 * httpd: added ui output handler * menu: added command 'o' for sending OPTION request (thanks to Juha Heinanen) added command 'D' for accepting incoming calls * mwi: subscribe to MWI after Registration succeeded (thanks to Juha Heinanen) * opensles: add double-buffering and some tuning (thanks to Francesco Bradascio) * opus: added config "opus_bitrate" (thanks to Sebastian Reimers) * presence: added support for PUBLISH (thanks to Juha Heinanen) interop fixes and tuning * stdio: update usage of struct ui, added output handler * uuid: use internal version of generating UUID * v4l2: use memory mapped mode only * vumeter: dont call tmr_start from non-RE thread * wincons: update usage of struct ui, added output handler * winwave: fix bug when closing player device (thanks to Tomasz Ostrowski) add support for mapping device name to index * zrtp: add support for verify SAS (thanks to Ingo Feinerer)
2016-09-06 08:13:31 +00:00
SHA1 (patch-modules_v4l2_v4l2.c) = 71ba2d1e5c8ba61eb011bd2b6b9e0d9cdaec5797