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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
# $NetBSD$
PKG_OPTIONS_VAR= PKG_OPTIONS.baresip
baresip: update to baresip-0.6.5 HEADS UP: module h265.so is now part of avcodec.so (and has been removed) module sdl2.so has been renamed sdl.so (adjust video_display if needed) Changelog: 2019-12-01 Alfred E. Heggestad * Version 0.6.5 Alfred E. Heggestad (138): mnat: add struct mnat pointer to session handler ice: add ice-lite, move to per-account config modules: check return value from uag_event_register() menu: check return value of account_set_answermode ua: move ua_print_sip_status to debug_cmd module pcp: updated mnat api modules: rename gst1.so to gst.so account: make answermode code more robust bfcp: remove code ua: remove uag_tls() sdl: add support for YUYV422 pixel format modules: rename gst_video1.so to gst_video.so sdl: add support for UYVY422 pixel format ua: fix whitespace test: mock_mnat_register return void stream: debug tuning test: enable wait_connected flag on mock mnat dtls_srtp: dont store remote address on the state test: enable wait_secure flag on mock mediaenc modules: rename sdl2.so to sdl.so menc: sort handlers in logical order audiounit: use error ENOTSUP if AudioSessionSetActive fails test: add webrtc test-case audiounit: check return value of AudioUnitSetProperty() bv32: remove module (#793) audiounit: fix samplerate for iOS config: add snd_path to template account: remove check for deprecated password audio: use dynamically allocated string for device name Update README.md avcodec: remove unused prototype avcodec: fix unused parameter warning move h265.so into avcodec.so Update .travis.yml (#798) Account specific audio source and playback (#796) mpa: switch encoder to use lame (#797) mk: remove GPROF ua: add support for SIP trace (#804) AAC codec (#805) audio: set the correct variable to false if pthread_create() fails sdl: properly close window (OSX) ua: fix warning rtcpsummary: use call object from event handler test: move aucodec list one level up video: add vidcodec accessor refactoring; move rtp stats code to separate .c file rtpstat cleanup call: check magic vidinfo: add video overlay box with decoder info vidinfo: fix compiler warning on linux vidinfo: fix compiler warning on Android stream: rename to stream_set_session_handlers() Fix osx build (#809) vidinfo: delete old file test: move ausrc list one level up test: move auplay list one level up test: move aufilt list one level up video: use vidcodec's list in video_decoder_set() stream: add stream-list to stream/audio/video API vidinfo: remove pixelformat, add packetloss mk: add detection in SYSROOT_LOCAL dtls_srtp: add media name and component type to logline mnat: make mnat_find() public audio: print name in parenthesis if not set video: move vidfilt list one level up audio: remove hack for starting source/player first video: set stream samplerate in alloc video: fix potential use of free'd string ice: fix documentation g711: use designated initialisers g722: use designated initialisers g726: use designated initialisers ilbc: use designated initialisers mk: check for ilbc in gsm: use designated initialisers amr: fix detection in SYSROOT_LOCAL amr: use designated initialisers isac: use designated initialisers test: add audio_codecs to account testcase win32: sort module exports in alphabetical order win32: add debug_cmd to static list of modules echo: no need to use uag_current() -- ref #815 mk: detect aac in SYSROOT_LOCAL modules: use designated initializers modules: use designated initializers v4l2_codec: use designated initializers wincons: use designated initializers gzrtp: use designated initializers avcodec: use designated initializers video: request keyframe during packet-loss mk: detect mqtt.so in SYSROOT_LOCAL menu: fix formatting menu: clean up usage of uag_current() -- ref #815 menu: save UA aor for redialing avcodec: use AVFrame key_frame flag to check for keyframes (#830) call: add call_find_id() -- ref #815 menu: new command /callfind -- ref #815 ice: make username/password optional mqtt: add ua/call selection -- fixes #815 stream: no RTCP socket for mediaenc, if muxed mqtt: encode response with JSON -- fix #826 main: change help text for -4 and -6 (ref #834) docs: thanks to @premultiply net: change prefer_ipv6 to int af, fixes #834 pulse: fix log text avformat: fix build on Debian 8 stream: log more details stream: make stream_start_mediaenc() public for core stream: move start_mediaenc to call.c audio: add ptime to struct aucodec (#849) README: add i2s module docs: thanks to Christian Spielberger test: remove unused setting of int err stream: remove code not executed message: no need to check err net: fix potential deref of NULL pointer vidinfo: no need to check err natpmp: no need to check err ilbc: no need to check err ctrl_tcp: no need to check oe_cmd here avformat: remove got_pict hack avcodec: minor fixes sdl: remove int err, not needed pulse: check the correct pointer v4l2_codec: remove int err, not needed alsa: store return value in a long avcodec: copy key_frame flag from hardware frame stream: make some functions public ctrl_tcp: restore mbuf pos on errors metric: add lock for multi-threading sdl: skip plane if wstep is zero aubridge: clear pointers after thread has exited fix doxygen comments av1: set allow_lowbitdepth to get correct pixel-format bump version to 0.6.5 stream: check return value of metric_init() stream: update doxygen comments update doxygen comments Christian Spielberger (2): aui2s: add rtos i2s audio driver module (#848) i2s: add doxygen defgroup header (#850) Juha Heinanen (3): - Added 'net_set_address' and 'net_set_af' API functions. Added safety check to net_set_address() API function. - Added AF_UNSPEC to supported net_set_af families. juha-h (4): - Exposed net_dns_debug function to API. (#791) Added possibility to include ";extra" parameter to an account and access (#803) - Search include files also from opencore-amr source directory (#822) Merge pull request #843 from alfredh/net_stuff premultiply (5): Unify response list header layout (#821) More common list format (#823) MPA fmtp mirroring (#837) MPA layer 3 encoding fixes (#839) MPA L2 and L3 encoding (#844) trampster (2): Pass NULL to pa_simple_new if no device specified to indicate to PulseAudio we want to use the default (#785) Add support for aes_256_gcm (#790)
2019-12-18 07:08:30 +00:00
PKG_SUPPORTED_OPTIONS= alsa cairo ffmpeg gstreamer gtk ilbc jack
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
PKG_SUPPORTED_OPTIONS+= libvpx oss opencore-amr opus portaudio
baresip: update to baresip-0.6.2 2019-04-19 Alfred E. Heggestad * Version 0.6.2 Alfred E. Heggestad (124): daala: remove module remove USE_VIDEO compile flag (#658) config: remove sip_trans_bsize option contact: fix bug in contact prev/next cmd: remove unused complete flag log: add command to toggle loglevel ('v') debug_cmd: fix warning Remove natbd module (#659) message: make listen/unlisten more robust (ref #650) update doxygen comments srtp: fix warnings update README stream: define port 9 as PORT_DISCARD add offerer flag to video and stream stream: check for multiplexed RTCP packets on RTP port stream: change logic for rtcp-mux attribute omx: update doxygen comment bv32: add doxygen header mk: modules in alphabetical order mk: modules in alphabetical order mk: modules in alphabetical order h265: use avcodec API for the encoder h265: fixes for Debian 8 h265: make it work with old encoder api h265: init time_base manually fix av_packet_free allocate avpacket h265: cleanup cleanup cleanup h265: add configurable decoder cleanup use pkg-config for libs h265: update documentation mk: enable h265.so if avcodec installed h265: include avutil mem.h travis: use ubuntu 16.04 h265: add wrapper for av_frame_alloc h265: add wrapper for avcodec_free_context h265: fix config fix crash with ffmpeg 2.8 add wrapper for av_packet_free fix warning use av_free_packet cleanup deprecate v4l.so -- use v4l2.so instead h265: tested with YUV444P pixel format h265: check pixel format on changes Merge remote-tracking branch 'origin/master' into h265_use_avcodec_encoder h265: fix avcodec_free_context wrapper H265 use avcodec encoder (#668) debian: add source format 1.0 aufile: add sample config (ref #663) v4l: remove module, use v4l2.so instead video: remote orient parameter video: remove video_set_orient aubridge: remove audio resampler aubridge: fix warning aubridge: add support for multiple sample formats aubridge: fix warnings auloop: remove usage of audio codec jack: add support for FLOAT sample format Avcodec remove libx264 (#671) config: add avcodec.so sample config stream: set rtcp-mux attribute if enabled sdl: remove module stream: send a dummy RTCP packet to open NAT pinhole avcodec: use pkg-config for linker flags avformat: use pkg-config for linker flags config: remove usage of USE_AVCODEC coreaudio: remove ios specific code h265: one file per line avformat: move AVCodec from struct to stack avformat: remove codec_id check avformat: minor cleanup debian: remove usage of shlibs:Depends from dev package debug_cmd: fix warning avformat: minor cleanup avformat: minor cleanup vidloop: rename intra to keyframe vidloop: print keyframes only if codec is enabled avcodec: move destructor to the top of the file plc: check input arguments auloop: rename ab to aubuf Opus multistream (#678) mqtt: minor updates mqtt: use re_snprintf mqtt: update documentation (fixes #669) h264: fix h264_is_keyframe, IDR_SLICE is keyframe avcodec: check input arguments vidloop: show video display pixel-format in summary avcodec: clean up decoding code avcodec: add color range MPEG h265: add color range and GOP size ffmpeg: check avutil version for color range avcodec: set slice-max-size in H264 packetization-mode 0 avcodec: add sdp.c avcodec: move h264_fmtp_cmp to sdp.c avcodec: add support for H264 packetization mode 1 av1: update comment avcodec: handle H264 STAP-A packets test: fix ua register test-cases (ref #680) ua: add delayed_close flag (ref #680) menu: call uag_current() directly video: save pixel format of outgoing stream aulevel: add support for sample format FLOAT Vidfilt add param (#682) test: copy uri_cmp source from libre sdl2: handle window closed event (SDL_QUIT) vidloop: stop loop if window was closed sdl2: add support for quit key avcodec: fixes for packetization_mode 1 bump version to 0.6.2 vidfilt: update doxygen comments rtcpsummary: fix warnings about unused variables mqtt: fix warnings about unused variables gst_video1: use GST_BUFFER_PTS mk: add echo module to list of basic modules core: remove some unused values menu: call parameter is used mwi: minor formatting changes audio: align debug text Update README.md ua: add ua_destroy() -- ref #686 Andreas Hansson (1): Added debug cmd to print UUID (#674) Juha Heinanen (2): do not add basic modules if BASIC_MODULES has value 'no' exclude more modules if BASIC_MODULES=no Nicolas Tizon (1): vidloop: update vidsrc (#662) Roger Sandholm (1): Readme and config example correction, httpd comments (#666) Timmo Verlaan (1): menu: add uadel to delete a uac by aor (#680) juha-h (1): srtp: added sending of MENC_EVENT_SECURE event (#660) premultiply (1): Add 48 khz sampling rate support (#685) weili-jiang (1): Command not found returns error (#664)
2019-05-16 20:39:12 +00:00
PKG_SUPPORTED_OPTIONS+= pulseaudio sdl2 sndfile sndio speex v4l2
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
PKG_SUPPORTED_OPTIONS+= x11
PKG_SUGGESTED_OPTIONS= oss ilbc speex
.include "../../mk/bsd.prefs.mk"
.include "../../mk/bsd.options.mk"
PLIST_VARS+= ${PKG_SUPPORTED_OPTIONS}
###
### ALSA support (audio output)
###
.if !empty(PKG_OPTIONS:Malsa)
PLIST.alsa= yes
MAKE_FLAGS+= USE_ALSA=yes
.include "../../audio/alsa-lib/buildlink3.mk"
.else
MAKE_FLAGS+= USE_ALSA=
.endif
###
### cairo support (video input)
###
.if !empty(PKG_OPTIONS:Mcairo)
PLIST.cairo= yes
MAKE_FLAGS+= USE_CAIRO=yes
.include "../../graphics/cairo/buildlink3.mk"
.else
MAKE_FLAGS+= USE_CAIRO=
.endif
###
### Gstreamer1 support (video codecs)
###
.if !empty(PKG_OPTIONS:Mgstreamer)
PLIST.gstreamer= yes
MAKE_FLAGS+= USE_GST1=yes
MAKE_FLAGS+= USE_GST_VIDEO1=yes
.include "../../multimedia/gstreamer1/buildlink3.mk"
.include "../../multimedia/gst-plugins1-base/buildlink3.mk"
.else
MAKE_FLAGS+= USE_GST1=
MAKE_FLAGS+= USE_GST_VIDEO1=
.endif
###
### GTK gui support
###
.if !empty(PKG_OPTIONS:Mgtk)
PLIST.gtk= yes
MAKE_FLAGS+= USE_GTK=yes
.include "../../x11/gtk2/buildlink3.mk"
.else
MAKE_FLAGS+= USE_GTK=
.endif
###
### ffmpeg support (video input)
###
.if !empty(PKG_OPTIONS:Mffmpeg)
PLIST.ffmpeg= yes
MAKE_FLAGS+= USE_AVCODEC=yes
MAKE_FLAGS+= USE_AVFORMAT=yes
baresip: depend on ffmpeg4 instead of ffmpeg3
2018-12-11 14:00:41 +00:00
LFLAGS+= ${COMPILER_RPATH_FLAG}${BUILDLINK_PREFIX.ffmpeg4}/${BUILDLINK_LIBDIRS.ffmpeg4}
LFLAGS+= -L${BUILDLINK_PREFIX.ffmpeg4}/${BUILDLINK_LIBDIRS.ffmpeg4}
.include "../../multimedia/ffmpeg4/buildlink3.mk"
baresip: avformat/avcodec plugin needs multimedia/x264-devel as well
2016-11-15 12:04:55 +00:00
.include "../../multimedia/x264-devel/buildlink3.mk"
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
.else
MAKE_FLAGS+= USE_AVCODEC=
MAKE_FLAGS+= USE_AVFORMAT=
.endif
###
### ILBC support (audio codec)
###
.if !empty(PKG_OPTIONS:Milbc)
PLIST.ilbc= yes
MAKE_FLAGS+= USE_ILBC=yes
#.include "../../wip/ilbc-rfc3951/buildlink3.mk"
.include "../../wip/libilbc/buildlink3.mk"
.else
MAKE_FLAGS+= USE_ILBC=
.endif
###
### Jack Audio Connection Kit support (audio output)
###
.if !empty(PKG_OPTIONS:Mjack)
PLIST.jack= yes
MAKE_FLAGS+= USE_JACK=yes
.include "../../audio/jack/buildlink3.mk"
.else
MAKE_FLAGS+= USE_JACK=
.endif
###
### vp8 and vp9 support (video codec)
###
.if !empty(PKG_OPTIONS:Mlibvpx)
PLIST.libvpx= yes
MAKE_FLAGS+= USE_VPX=yes
.include "../../multimedia/libvpx/buildlink3.mk"
.else
MAKE_FLAGS+= USE_VPX=
.endif
###
### OSS support (audio output)
###
.if !empty(PKG_OPTIONS:Moss)
PLIST.oss= yes
MAKE_FLAGS+= USE_OSS=yes # full-duplex issues
.else
MAKE_FLAGS+= USE_OSS=
.endif
###
### opencore-amr support (audio codec)
###
.if !empty(PKG_OPTIONS:Mopencore-amr)
PLIST.opencore-amr= yes
MAKE_FLAGS+= USE_AMR=yes
.include "../../audio/opencore-amr/buildlink3.mk"
.else
MAKE_FLAGS+= USE_AMR=
.endif
###
### opus support (audio codec)
###
.if !empty(PKG_OPTIONS:Mopus)
PLIST.opus= yes
MAKE_FLAGS+= USE_OPUS=yes
.include "../../audio/libopus/buildlink3.mk"
.else
MAKE_FLAGS+= USE_OPUS=
.endif
###
### Portaudio support (audio output)
###
.if !empty(PKG_OPTIONS:Mportaudio)
PLIST.portaudio= yes
MAKE_FLAGS+= USE_PORTAUDIO=yes
.include "../../audio/portaudio/buildlink3.mk"
.else
MAKE_FLAGS+= USE_PORTAUDIO=
.endif
###
### Pulseaudio support (audio output)
###
.if !empty(PKG_OPTIONS:Mpulseaudio)
PLIST.pulseaudio= yes
MAKE_FLAGS+= USE_PULSE=yes
.include "../../audio/pulseaudio/buildlink3.mk"
.else
MAKE_FLAGS+= USE_PULSE=
.endif
###
### SDL2 support (video output)
###
.if !empty(PKG_OPTIONS:Msdl2)
PLIST.sdl2= yes
baresip: update to baresip-0.6.6 No changelog provided for this (github) release. Relevant commits: https://github.com/alfredh/baresip/compare/v0.6.5...v0.6.6
2020-05-09 22:31:32 +00:00
MAKE_FLAGS+= USE_SDL=yes
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
.include "../../devel/SDL2/buildlink3.mk"
.else
baresip: update to baresip-0.6.6 No changelog provided for this (github) release. Relevant commits: https://github.com/alfredh/baresip/compare/v0.6.5...v0.6.6
2020-05-09 22:31:32 +00:00
MAKE_FLAGS+= USE_SDL=
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
.endif
###
### sndfile support (audio input)
###
.if !empty(PKG_OPTIONS:Msndfile)
PLIST.sndfile= yes
MAKE_FLAGS+= USE_SNDFILE=yes
.include "../../audio/libsndfile/buildlink3.mk"
.else
MAKE_FLAGS+= USE_SNDFILE=
.endif
###
### sndio support (audio output)
###
.if !empty(PKG_OPTIONS:Msndio)
PLIST.sndio= yes
MAKE_FLAGS+= USE_SNDIO=yes
# XXX this needs a buildlink3.mk ?
DEPENDS+= sndio-[0-9]*:../../wip/sndio
.else
MAKE_FLAGS+= USE_SNDIO=
.endif
###
### speex support (audio codec)
###
.if !empty(PKG_OPTIONS:Mspeex)
PLIST.speex= yes
MAKE_FLAGS+= HAVE_SPEEXDSP=yes
MAKE_FLAGS+= USE_SPEEX_AEC=yes
MAKE_FLAGS+= USE_SPEEX_PP=yes
.include "../../audio/speexdsp/buildlink3.mk"
.else
MAKE_FLAGS+= HAVE_SPEEXDSP=
MAKE_FLAGS+= USE_SPEEX_AEC=
MAKE_FLAGS+= USE_SPEEX_PP=
.endif
###
### Video4Linux2 support (video input)
###
.if !empty(PKG_OPTIONS:Mv4l2)
PLIST.v4l2= yes
MAKE_FLAGS+= HAVE_LIBV4L2=yes
MAKE_FLAGS+= USE_V4L2=yes
.include "../../graphics/libv4l/buildlink3.mk"
.else
MAKE_FLAGS+= USE_V4L2=
.endif
###
### x11 support (video output)
###
.if !empty(PKG_OPTIONS:Mx11)
PLIST.x11= yes
LFLAGS+= ${X11_LDFLAGS}
MAKE_FLAGS+= USE_X11=yes
.include "../../x11/libX11/buildlink3.mk"
.include "../../x11/libXext/buildlink3.mk"
.else
MAKE_FLAGS+= USE_X11=
.endif