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baresip: update to baresip-2.5.0
Module Name: pkgsrc-wip
Committed By: Yorick Hardy <yorickhardy%gmail.com@localhost>
Pushed By: yhardy
Date: Mon Jul 18 10:07:07 2022 +0200
Changeset: 498aca0dea7667a1c863928a8c3f73fa747cc6e2
Modified Files:
baresip/Makefile
baresip/PLIST
baresip/distinfo
baresip/options.mk
Added Files:
baresip/patches/patch-modules_portaudio_module.mk
Removed Files:
baresip/patches/patch-modules_ilbc_ilbc.c
Log Message:
baresip: update to baresip-2.5.0
Still needs more testing for all the different options and plugins.
Changelog:
== 2.5.0 - 2022-07-01
== What's Changed
* audio: add optional decoding buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1842
* audio: RX filter thread needs separate sampv buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1879
* aufile: fix possible data race warning by @cspiel1 in https://github.com/baresip/baresip/pull/1880
* audiounit,coreaudio: fix kAudioObjectPropertyElementMaster deprecation by @sreimers in https://github.com/baresip/baresip/pull/1881
* av1: explicitly check for supported OBU types by @alfredh in https://github.com/baresip/baresip/pull/1882
* audiounit/coreaudio: fix kAudioObjectPropertyElementMain by @sreimers in https://github.com/baresip/baresip/pull/1885
* ci/build: bump macos min. sdk to 10.12 by @sreimers in https://github.com/baresip/baresip/pull/1883
* ci: run only for pull requests and main branch by @sreimers in https://github.com/baresip/baresip/pull/1887
* multicast: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1892
* dtls_srtp: enable ECC by default, remove RSA by @alfredh in https://github.com/baresip/baresip/pull/1891
* ci/build: add ubuntu 22.04 by @sreimers in https://github.com/baresip/baresip/pull/1890
* test: add check for memory leaks by @sreimers in https://github.com/baresip/baresip/pull/1896
* stream,metric: RX real-time - make metric thread-safe by @cspiel1 in https://github.com/baresip/baresip/pull/1895
* Cmake findre by @alfredh in https://github.com/baresip/baresip/pull/1893
* test: wait for both audio and video to be established by @alfredh in https://github.com/baresip/baresip/pull/1903
* docs: remove old TODO file by @alfredh in https://github.com/baresip/baresip/pull/1902
* audio: fixed check for aubuf started flag by @cspiel1 in https://github.com/baresip/baresip/pull/1904
* use new mutex interface by @cspiel1 in https://github.com/baresip/baresip/pull/1905
* audio: make rx.filtl thread-safe by @cspiel1 in https://github.com/baresip/baresip/pull/1897
* audio: allocate correct buffer size for static auplay srate by @cspiel1 in https://github.com/baresip/baresip/pull/1906
* Pulseaudio Async Interface Module by @cHuberCoffee in https://github.com/baresip/baresip/pull/1907
* Do not destroy register client when it is unregistered by @juha-h in https://github.com/baresip/baresip/pull/1908
* Two spaces are required after email address by @juha-h in https://github.com/baresip/baresip/pull/1909
* cmake: add alsa module by @alfredh in https://github.com/baresip/baresip/pull/1910
* cmake: fix static openssl and thread linking by @sreimers in https://github.com/baresip/baresip/pull/1911
* In start_registering, create register clients if reg list is empty by @juha-h in https://github.com/baresip/baresip/pull/1913
* ctrl_dbus: use new thread and mtx interface by @cspiel1 in https://github.com/baresip/baresip/pull/1916
* cmake: add pulse and pulse_async module by @cHuberCoffee in https://github.com/baresip/baresip/pull/1919
* Un-subscribe mwi at un-register by @juha-h in https://github.com/baresip/baresip/pull/1918
* call: update media on session progress. by @RobertMi21 in https://github.com/baresip/baresip/pull/1922
* ctrl_dbus send event in main thread by @cspiel1 in https://github.com/baresip/baresip/pull/1921
* uag: add timestamps to SIP trace by @cspiel1 in https://github.com/baresip/baresip/pull/1914
* main: fix open timers check by @sreimers in https://github.com/baresip/baresip/pull/1925
* cmake: add account module by @alfredh in https://github.com/baresip/baresip/pull/1926
---
== 2.4.0 - 2022-06-01
== What's Changed
* mulitcast unmute bad quality by @cspiel1 in https://github.com/baresip/baresip/pull/1821
* menu ringback for parallel call by @cspiel1 in https://github.com/baresip/baresip/pull/1827
* multicast: support error code EAGAIN of jbuf_get() by @cspiel1 in https://github.com/baresip/baresip/pull/1832
* use RTP clock rate for timestamp calculation by @cspiel1 in https://github.com/baresip/baresip/pull/1834
* av1 obu by @alfredh in https://github.com/baresip/baresip/pull/1835
* av1 packetizer by @alfredh in https://github.com/baresip/baresip/pull/1836
* av1: depacketizer by @alfredh in https://github.com/baresip/baresip/pull/1837
* Disabled debug statement by @juha-h in https://github.com/baresip/baresip/pull/1838
* h264: move from rem to re by @sreimers in https://github.com/baresip/baresip/pull/1839
* ua: send new event UA_EVENT_CREATE at successful ua allocation by @cHuberCoffee in https://github.com/baresip/baresip/pull/1840
* evdev: fix wrong ioctl size by @sreimers in https://github.com/baresip/baresip/pull/1843
* aufile: ausrc_prm has to be copied when source is allocated by @cspiel1 in https://github.com/baresip/baresip/pull/1844
* conf: missing pointer initialization found by clang analyzer by @cspiel1 in https://github.com/baresip/baresip/pull/1845
* mk/modules: fix omx RPI detection by @sreimers in https://github.com/baresip/baresip/pull/1847
* auconv: add auconv_to_float (fixes #1833) by @alfredh in https://github.com/baresip/baresip/pull/1849
* avfilter: migrate to C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1850
* avformat: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1851
* selfview: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1852
* audio: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1853
* metric: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1854
* play: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1855
* dns: add query cache by @sreimers in https://github.com/baresip/baresip/pull/1848
* video: C11 mutex by @alfredh in https://github.com/baresip/baresip/pull/1856
* aufile: C11 threads by @alfredh in https://github.com/baresip/baresip/pull/1858
* audio: add more locking by @alfredh in https://github.com/baresip/baresip/pull/1857
* aufile/play: fix run data race by @sreimers in https://github.com/baresip/baresip/pull/1859
* mc: multicast receiver enable state fix by @cHuberCoffee in https://github.com/baresip/baresip/pull/1861
* audio: C11 thread by @alfredh in https://github.com/baresip/baresip/pull/1860
* av1: add packetize handler by @alfredh in https://github.com/baresip/baresip/pull/1865
* net/net_debug: add default route hint by @sreimers in https://github.com/baresip/baresip/pull/1864
* ice: fix local prio calculation by @sreimers in https://github.com/baresip/baresip/pull/1863
* avformat: open codec if not passthrough by @alfredh in https://github.com/baresip/baresip/pull/1866
* dtls_srtp: Minor whitespace fix by @robert-scheck in https://github.com/baresip/baresip/pull/1870
* vp8: add packetize handler by @alfredh in https://github.com/baresip/baresip/pull/1868
* vp9: add packetizer by @alfredh in https://github.com/baresip/baresip/pull/1871
* debug_cmd: support absolute path for command aufileinfo by @cspiel1 in https://github.com/baresip/baresip/pull/1875
* event: add diverter URI to UA event by @cspiel1 in https://github.com/baresip/baresip/pull/1876
* aufileinfo with synchronous response by @cspiel1 in https://github.com/baresip/baresip/pull/1877
**Full Changelog**: https://github.com/baresip/baresip/compare/v2.3.0...v2.4.0
---
== [2.3.0] - 2022-05-01
== What's Changed
* mc: multicast mute function by @cHuberCoffee in https://github.com/baresip/baresip/pull/1805
* mc: reject incoming call if high prio multicast is received by @cHuberCoffee in https://github.com/baresip/baresip/pull/1804
* mc: mcplayer stream fade-out and fade-in by @cHuberCoffee in https://github.com/baresip/baresip/pull/1802
* clean_number now will remove all non-digit chars by @mbattista in https://github.com/baresip/baresip/pull/1806
* Workflows cmakelint by @alfredh in https://github.com/baresip/baresip/pull/1808
* ccheck: check all CMakeLists.txt files by @sreimers in https://github.com/baresip/baresip/pull/1810
* mk: remove win32 MSVC project files by @alfredh in https://github.com/baresip/baresip/pull/1811
* cmake: add modules by @sreimers in https://github.com/baresip/baresip/pull/1812
* ajb,aubuf: timestamp is given in [us] by @cspiel1 in https://github.com/baresip/baresip/pull/1809
* call: allow optional leading space in SIP INFO for dtmf-relay by @thomas-karl in https://github.com/baresip/baresip/pull/1814
* conf: add fs_file_extension() by @alfredh in https://github.com/baresip/baresip/pull/1816
* Updated debian version by @juha-h in https://github.com/baresip/baresip/pull/1817
* pulse: fix timestamp integer overrun for arm by @cspiel1 in https://github.com/baresip/baresip/pull/1818
* fix audio multicast artefacts by @cspiel1 in https://github.com/baresip/baresip/pull/1819
* audio: flush aubuf if ssrc changes by @cspiel1 in https://github.com/baresip/baresip/pull/1822
* Debian control dependency update by @juha-h in https://github.com/baresip/baresip/pull/1823
* pulse: support restart of pulseaudio during stream by @cspiel1 in https://github.com/baresip/baresip/pull/1824
* version 2.3.0 by @alfredh in https://github.com/baresip/baresip/pull/1826
== New Contributors
* @thomas-karl made their first contribution in https://github.com/baresip/baresip/pull/1814
---
== [2.0.2] - 2022-04-09
== What's Changed
* Added API function call_diverteruri by @juha-h in https://github.com/baresip/baresip/pull/1780
* Avoid undeclared 'CLOCK_REALTIME' on RHEL/CentOS 7 (fixes #1781) by @robert-scheck in https://github.com/baresip/baresip/pull/1782
* audio: add lock in audio_send_digit by @GGGO in https://github.com/baresip/baresip/pull/1786
* vumeter: use new auframe_level() by @sreimers in https://github.com/baresip/baresip/pull/1788
* reg.c: use already declared acc by @GGGO in https://github.com/baresip/baresip/pull/1789
* aubuf adaptive jitter buffer by @cspiel1 in https://github.com/baresip/baresip/pull/1784
* multicast set aubuf silence by @cspiel1 in https://github.com/baresip/baresip/pull/1791
* ccheck: fix line number in error print by @cspiel1 in https://github.com/baresip/baresip/pull/1793
* test: check the correct stream in UA_EVENT_CALL_MENC by @alfredh in https://github.com/baresip/baresip/pull/1794
* audio: missing lock around stream_send by @GGGO in https://github.com/baresip/baresip/pull/1796
* docs: remove obsolete jitter_buffer_wish from config example by @cspiel1 in https://github.com/baresip/baresip/pull/1798
* Multicast jbuf and aubuf changes by @cHuberCoffee in https://github.com/baresip/baresip/pull/1797
* uag: uag_hold_resume() should not return err if there is no call to hold by @cspiel1 in https://github.com/baresip/baresip/pull/1799
* stream: remove mbuf_get_left check in rtp_handler by @GGGO in https://github.com/baresip/baresip/pull/1801
* cmake: preliminary support by @alfredh in https://github.com/baresip/baresip/pull/1800
== New Contributors
* @GGGO made their first contribution in https://github.com/baresip/baresip/pull/1786
---
== [2.0.1] - 2022-03-27
=== What's Changed
* audio: fix rx_thread (adaptive jitter buffer) by @sreimers in https://github.com/baresip/baresip/pull/1769
* test: init fixture by @alfredh in https://github.com/baresip/baresip/pull/1772
* test: refactoring of test_account_uri_complete by @alfredh in https://github.com/baresip/baresip/pull/1773
* mk: check also if extensions/XShm.h is present by @cspiel1 in https://github.com/baresip/baresip/pull/1774
* menu: support custom SIP headers by @cspiel1 in https://github.com/baresip/baresip/pull/1775
* menu: use new sdp_dir_decode by @cspiel1 in https://github.com/baresip/baresip/pull/1776
* menu: avoid multiple hash entries with same key by @cspiel1 in https://github.com/baresip/baresip/pull/1777
* menu: support audio file config value "none" by @cspiel1 in https://github.com/baresip/baresip/pull/1778
* intercom: add video preview call by @cspiel1 in https://github.com/baresip/baresip/pull/1779
---
== [2.0.0] - 2022-03-11
=== What's Changed
* debug_cmd: use module_event() for aufileinfo events by @cspiel1 in https://github.com/baresip/baresip/pull/1345
* multicast: use module_event() for sending events by @cspiel1 in https://github.com/baresip/baresip/pull/1346
* ctrl_dbus: use module_event() to send exported event by @cspiel1 in https://github.com/baresip/baresip/pull/1347
* ua,call: add CALL_EVENT_OUTGOING by @cspiel1 in https://github.com/baresip/baresip/pull/1348
* GTK caller history by @mbattista in https://github.com/baresip/baresip/pull/1350
* Convert FRITZ!Box XML phone book into Baresip contacts by @robert-scheck in https://github.com/baresip/baresip/pull/1382
* menu: play ringtone on audio_alert device by @cspiel1 in https://github.com/baresip/baresip/pull/1396
* menu: use str_isset() for command parameter by @cspiel1 in https://github.com/baresip/baresip/pull/1397
* dtls_srtp: use elliptic curve cryptography by @cHuberCoffee in https://github.com/baresip/baresip/pull/1385
* Support for s16 playback in jack. Needed for play tones by @srperens in https://github.com/baresip/baresip/pull/1399
* Check that account ;sipnat param has valid value by @juha-h in https://github.com/baresip/baresip/pull/1401
* Tls sipcert per acc by @cHuberCoffee in https://github.com/baresip/baresip/pull/1376
* Vidsrc add packet handler by @alfredh in https://github.com/baresip/baresip/pull/1402
* ToS for video and sip by @cspiel1 in https://github.com/baresip/baresip/pull/1393
* account: add accounts parameter to force media address family by @cspiel1 in https://github.com/baresip/baresip/pull/1395
* Selective early media by @cspiel1 in https://github.com/baresip/baresip/pull/1398
* ua,uag: split ua.c and uag.c by @cspiel1 in https://github.com/baresip/baresip/pull/1349
* Account media af template by @cspiel1 in https://github.com/baresip/baresip/pull/1406
* account: add missing client certificate parameter to template by @cHuberCoffee in https://github.com/baresip/baresip/pull/1408
* account: update answermode values in template by @cspiel1 in https://github.com/baresip/baresip/pull/1405
* menu: command uafind raises UA to head by @cspiel1 in https://github.com/baresip/baresip/pull/1407
* ctrl_dbus: fix possible memleak on failed initialization by @cspiel1 in https://github.com/baresip/baresip/pull/1410
* video passthrough by @alfredh in https://github.com/baresip/baresip/pull/1418
* menu: enable auto answer calls also for command dialdir by @cspiel1 in https://github.com/baresip/baresip/pull/1412
* menu: add command for settings media local direction by @cspiel1 in https://github.com/baresip/baresip/pull/1413
* Accounts addr params by @cspiel1 in https://github.com/baresip/baresip/pull/1414
* Accounts example cleanup by @cspiel1 in https://github.com/baresip/baresip/pull/1415
* menu,call: fix hangup for outgoing call by @cspiel1 in https://github.com/baresip/baresip/pull/1417
* multicast: add source and player API calls by @cHuberCoffee in https://github.com/baresip/baresip/pull/1403
* menu: add command /uareg by @alfredh in https://github.com/baresip/baresip/pull/1421
* menu: return complete URI for commands dial,dialdir by @cspiel1 in https://github.com/baresip/baresip/pull/1424
* menu: in command dialdir call uag_find_requri() with uri by @cspiel1 in https://github.com/baresip/baresip/pull/1425
* gst: replace variable length array (buf) with mem_zalloc by @sreimers in https://github.com/baresip/baresip/pull/1426
* menu: avoid possible memleaks for dial/dialdir commands by @cspiel1 in https://github.com/baresip/baresip/pull/1430
* uag: use local cuser for selecting user-agent (#1433) by @cspiel1 in https://github.com/baresip/baresip/pull/1434
* Work on Intercom module by @cspiel1 in https://github.com/baresip/baresip/pull/1432
* Attended Transfer on GTK by @mbattista in https://github.com/baresip/baresip/pull/1435
* Update README.md with configuration suggestion by @webstean in https://github.com/baresip/baresip/pull/1438
* README fixes by @juha-h in https://github.com/baresip/baresip/pull/1440
* Accounts examples and template by @cspiel1 in https://github.com/baresip/baresip/pull/1441
* serreg: use a timer for registration restart by @cspiel1 in https://github.com/baresip/baresip/pull/1445
* gst: audio playback not correct for some WAV files. by @RobertMi21 in https://github.com/baresip/baresip/pull/1442
* Working on intercom (ringtone override) by @cspiel1 in https://github.com/baresip/baresip/pull/1436
* Use line number 0 if user did not provide any line number by @negbie in https://github.com/baresip/baresip/pull/1451
* AMR Bandwidth Efficient mode support by @srperens in https://github.com/baresip/baresip/pull/1423
* Working on Intercom (menu: allow other modules to reject a call) by @cspiel1 in https://github.com/baresip/baresip/pull/1437
* auframe: add samplerate and channels by @sreimers in https://github.com/baresip/baresip/pull/1452
* account: comment out very basic example in template by @cspiel1 in https://github.com/baresip/baresip/pull/1458
* call answer media dir by @cspiel1 in https://github.com/baresip/baresip/pull/1449
* Account auto answer beep by @cspiel1 in https://github.com/baresip/baresip/pull/1461
* serreg: unregister correct User-Agents on registration failure by @cspiel1 in https://github.com/baresip/baresip/pull/1462
* mk: enable auto-detect of av1 module by @alfredh in https://github.com/baresip/baresip/pull/1463
* ctrl dbus makefile depends by @cspiel1 in https://github.com/baresip/baresip/pull/1457
* stream: check if media is present before enabling the RTP timeout by @cspiel1 in https://github.com/baresip/baresip/pull/1465
* ctrl_dbus: generate dbus code and documentation in makefile by @cspiel1 in https://github.com/baresip/baresip/pull/1456
* auframe: always set srate and ch by @janh in https://github.com/baresip/baresip/pull/1468
* auto answer beep per alert info URI by @cspiel1 in https://github.com/baresip/baresip/pull/1466
* auframe: move to rem by @sreimers in https://github.com/baresip/baresip/pull/1470
* mixminus: add conference feature by @sreimers in https://github.com/baresip/baresip/pull/1411
* vidbridge: check vidbridge_disp_display args fixes segfault by @sreimers in https://github.com/baresip/baresip/pull/1471
* gst: fixed some memory leaks by @RobertMi21 in https://github.com/baresip/baresip/pull/1476
* ua, menu: move auto answer delay handling to menu (#1474) by @cspiel1 in https://github.com/baresip/baresip/pull/1475
* ua,menu: move handling of ANSWERMODE_AUTO to menu (#1474) by @cspiel1 in https://github.com/baresip/baresip/pull/1478
* ausine: support for multiple samplerates by @alfredh in https://github.com/baresip/baresip/pull/1479
* account: fix IPv6 only URI for account_uri_complete() by @cspiel1 in https://github.com/baresip/baresip/pull/1472
* ilbc: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1483
* aubridge/device: remove unused sampv_out (old resample code) by @sreimers in https://github.com/baresip/baresip/pull/1484
* pkg-config version check by @sreimers in https://github.com/baresip/baresip/pull/1481
* mk: support more locations for libre.pc and librem.pc by @cspiel1 in https://github.com/baresip/baresip/pull/1486
* net: remove unused domain by @alfredh in https://github.com/baresip/baresip/pull/1489
* audio: fix aufilt_setup update handling by @sreimers in https://github.com/baresip/baresip/pull/1498
* SIP redirect callbackfunction by @cHuberCoffee in https://github.com/baresip/baresip/pull/1495
* add secure websocket tls context by @sreimers in https://github.com/baresip/baresip/pull/1499
* test: add stunuri by @alfredh in https://github.com/baresip/baresip/pull/1503
* turn: refactoring, add compv by @alfredh in https://github.com/baresip/baresip/pull/1505
* fmt: add string to bool function by @cspiel1 in https://github.com/baresip/baresip/pull/1501
* mk: check glib-2.0 at least like in ubuntu 18.04 by @cspiel1 in https://github.com/baresip/baresip/pull/1507
* registration fixes by @cspiel1 in https://github.com/baresip/baresip/pull/1510
* uag,menu: add commands to enable/disable UDP/TCP/TLS by @cspiel1 in https://github.com/baresip/baresip/pull/1502
* config,audio: add setting audio.telev_pt by @cspiel1 in https://github.com/baresip/baresip/pull/1509
* stream: fix telephone event (#1494) by @cspiel1 in https://github.com/baresip/baresip/pull/1506
* Fix I2S compile error, use auframe by @andreaswatch in https://github.com/baresip/baresip/pull/1512
* ci/tools: fix pylint by @sreimers in https://github.com/baresip/baresip/pull/1515
* config: not all audio config was printed by @cspiel1 in https://github.com/baresip/baresip/pull/1516
* net: replace network_if_getname with net_if_getname by @sreimers in https://github.com/baresip/baresip/pull/1518
* account: add setting audio payload type for telephone-event by @cspiel1 in https://github.com/baresip/baresip/pull/1517
* uag,menu: simplify transport enable/disable and support also ws/wss by @cspiel1 in https://github.com/baresip/baresip/pull/1514
* rst: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1519
* turn: add TCP and TLS transports by @alfredh in https://github.com/baresip/baresip/pull/1520
* speex_pp: remove deprecated module by @alfredh in https://github.com/baresip/baresip/pull/1521
* call: allow video calls by only rejecting a call without any common codecs by @cHuberCoffee in https://github.com/baresip/baresip/pull/1523
* multicast: add missing join for multicast addresses by @cHuberCoffee in https://github.com/baresip/baresip/pull/1524
* confg,uag: rework on sip_transports setting by @cspiel1 in https://github.com/baresip/baresip/pull/1525
* ua: check if peer is capable of video for early video by @cHuberCoffee in https://github.com/baresip/baresip/pull/1526
* mqtt/subscribe: replace fixed command buf and increase response size by @sreimers in https://github.com/baresip/baresip/pull/1527
* mqtt: add reconnect handling (lost broker connection) by @sreimers in https://github.com/baresip/baresip/pull/1528
* event: increase module_event buffer size by @sreimers in https://github.com/baresip/baresip/pull/1532
* mqtt/subscribe: use safe odict_string to prevent crashes by @sreimers in https://github.com/baresip/baresip/pull/1534
* stream: add stream_set_label by @alfredh in https://github.com/baresip/baresip/pull/1537
* Makefile dependency check improvements by @sreimers in https://github.com/baresip/baresip/pull/1531
* account: add enable/disable flag for video by @cspiel1 in https://github.com/baresip/baresip/pull/1536
* audio: use account specific audio telev pt correctly by @cspiel1 in https://github.com/baresip/baresip/pull/1542
* net: add missing HAVE_INET6 by @cspiel1 in https://github.com/baresip/baresip/pull/1543
* account: remove unused API function for video enable by @cspiel1 in https://github.com/baresip/baresip/pull/1544
* gst: changed log level for end of file message by @RobertMi21 in https://github.com/baresip/baresip/pull/1548
* multicast: add new configurable multicast TTL config parameter by @cHuberCoffee in https://github.com/baresip/baresip/pull/1545
* call: fix early video capability check (wrong SDP direction checked) by @cHuberCoffee in https://github.com/baresip/baresip/pull/1549
* audio: catch end of file message in ausrc error handler (#1539) by @RobertMi21 in https://github.com/baresip/baresip/pull/1550
* menu: added stopringing command by @RobertMi21 in https://github.com/baresip/baresip/pull/1551
* stream: remove obsolete rx.jbuf_started by @cspiel1 in https://github.com/baresip/baresip/pull/1552
* ua: downgrade level of message "ua: using best effort AF" by @viordash in https://github.com/baresip/baresip/pull/1553
* outgoing calls early callid by @cspiel1 in https://github.com/baresip/baresip/pull/1547
* audio: changed log level for ausrc error handler messages by @RobertMi21 in https://github.com/baresip/baresip/pull/1554
* SIP default protocol by @cspiel1 in https://github.com/baresip/baresip/pull/1538
* serreg: fix server selection in case all server were unavailable by @cHuberCoffee in https://github.com/baresip/baresip/pull/1557
* multicast: fix missing unlock by @alfredh in https://github.com/baresip/baresip/pull/1559
* config: replace strcpy by saver re_snprintf (#1558) by @cspiel1 in https://github.com/baresip/baresip/pull/1560
* multicast: fix coverity scan by @alfredh in https://github.com/baresip/baresip/pull/1561
* odict: hide struct odict_entry by @sreimers in https://github.com/baresip/baresip/pull/1562
* ctrl_dbus: use mqueue to trigger processing of command in remain thread by @cspiel1 in https://github.com/baresip/baresip/pull/1565
* multicast,config: add separate jitter buffer configuration by @cspiel1 in https://github.com/baresip/baresip/pull/1566
* ua: emit CALL_CLOSED event when user agent is deleted by @cspiel1 in https://github.com/baresip/baresip/pull/1564
* core: move stream_enable_rtp_timeout to api by @sreimers in https://github.com/baresip/baresip/pull/1569
* stream: add mid sdp attribute by @alfredh in https://github.com/baresip/baresip/pull/1570
* rtpext: change length type to size_t by @alfredh in https://github.com/baresip/baresip/pull/1573
* avcodec: remove old backwards compat wrapper by @alfredh in https://github.com/baresip/baresip/pull/1575
* main: Added option (-a) to set the ua agent string. by @RobertMi21 in https://github.com/baresip/baresip/pull/1576
* menu fix tones for parallel outgoing calls by @cspiel1 in https://github.com/baresip/baresip/pull/1577
* Fix win32 by @viordash in https://github.com/baresip/baresip/pull/1579
* Fix static analyzer warnings by @viordash in https://github.com/baresip/baresip/pull/1580
* call: added auto dtmf mode by @RobertMi21 in https://github.com/baresip/baresip/pull/1583
* RTP inbound telephone events should not lead to packet loss by @cspiel1 in https://github.com/baresip/baresip/pull/1581
* Running tests in a win32 project by @viordash in https://github.com/baresip/baresip/pull/1585
* stream: wrong media direction after setting stream to hold by @RobertMi21 in https://github.com/baresip/baresip/pull/1587
* move network check to module by @cspiel1 in https://github.com/baresip/baresip/pull/1584
* serreg: do not ignore returned errors of ua_register() by @cspiel1 in https://github.com/baresip/baresip/pull/1589
* Bundle media mux by @alfredh in https://github.com/baresip/baresip/pull/1588
* mixausrc: no warnings flood when sampc changes by @cspiel1 in https://github.com/baresip/baresip/pull/1595
* ua: select laddr with route to SDP offer address by @cspiel1 in https://github.com/baresip/baresip/pull/1590
* net,uag: allow incoming peer-to-peer calls with user@domain by @cspiel1 in https://github.com/baresip/baresip/pull/1591
* uag: in uag_reset_transp() select laddr with route to SDP raddr by @cspiel1 in https://github.com/baresip/baresip/pull/1592
* uag: exit if transport could not be added by @cspiel1 in https://github.com/baresip/baresip/pull/1593
* avcodec: use const AVCodec by @alfredh in https://github.com/baresip/baresip/pull/1602
* module: deprecate module_tmp by @alfredh in https://github.com/baresip/baresip/pull/1600
* test: use ausine as audio source by @alfredh in https://github.com/baresip/baresip/pull/1601
* Selftest fakevideo by @alfredh in https://github.com/baresip/baresip/pull/1604
* When adding local address, check that it has not been added already by @juha-h in https://github.com/baresip/baresip/pull/1606
* start without network by @cspiel1 in https://github.com/baresip/baresip/pull/1607
* config: add netroam module by @sreimers in https://github.com/baresip/baresip/pull/1608
* multicast: allow any port number for sender and receiver by @cHuberCoffee in https://github.com/baresip/baresip/pull/1609
* netroam: add netlink immediate network change detection by @cspiel1 in https://github.com/baresip/baresip/pull/1612
* remove uag transp rm (#1611) by @cspiel1 in https://github.com/baresip/baresip/pull/1616
* net dns srv get by @cspiel1 in https://github.com/baresip/baresip/pull/1615
* move calls to stream_start_rtcp to call.c by @alfredh in https://github.com/baresip/baresip/pull/1617
* video: null pointer check for the display handler by @cspiel1 in https://github.com/baresip/baresip/pull/1621
* audio: add lock by @alfredh in https://github.com/baresip/baresip/pull/1619
* ua: select proper af and laddr for outgoing IP calls by @cspiel1 in https://github.com/baresip/baresip/pull/1618
* audio: lock stream by @alfredh in https://github.com/baresip/baresip/pull/1622
* test: replace mock ausrc with ausine by @alfredh in https://github.com/baresip/baresip/pull/1623
* menu ringback session progress by @cspiel1 in https://github.com/baresip/baresip/pull/1625
* New module providing webrtc aec mobile mode filter by @juha-h in https://github.com/baresip/baresip/pull/1626
* uag: respect setting sip_listen (#1627) by @cspiel1 in https://github.com/baresip/baresip/pull/1628
* select laddr for SDP with respect to net_interface by @cspiel1 in https://github.com/baresip/baresip/pull/1630
* stream: do not start audio during early-video by @cspiel1 in https://github.com/baresip/baresip/pull/1629
* remove struct media_ctx by @alfredh in https://github.com/baresip/baresip/pull/1632
* ci: add libwebrtc-audio-processing-dev (module webrtc_aec) by @sreimers in https://github.com/baresip/baresip/pull/1635
* auconv: new module for audio format conversion by @alfredh in https://github.com/baresip/baresip/pull/1634
* Support for IPv6 link local address for streams by @cspiel1 in https://github.com/baresip/baresip/pull/1624
* call: check if address family is valid also for video stream by @cspiel1 in https://github.com/baresip/baresip/pull/1636
* audio: pass pointer to tx->ausrc_prm instead of local variable by @cspiel1 in https://github.com/baresip/baresip/pull/1637
* menu: add an event for call transfer by @cspiel1 in https://github.com/baresip/baresip/pull/1641
* netroam: error handling for reset transport by @cspiel1 in https://github.com/baresip/baresip/pull/1642
* mk: use CC_TEST for auto detect modules by @sreimers in https://github.com/baresip/baresip/pull/1647
* test: use dtls_srtp.so module instead of mock by @alfredh in https://github.com/baresip/baresip/pull/1646
* stream: create jbuf only if use_rtp is set by @cspiel1 in https://github.com/baresip/baresip/pull/1648
* multicast: fix memleak in player destructor by @cspiel1 in https://github.com/baresip/baresip/pull/1653
* stream: split up sender/receiver by @alfredh in https://github.com/baresip/baresip/pull/1654
* set sdp laddr to SIP src address by @cspiel1 in https://github.com/baresip/baresip/pull/1645
* serreg fix fallback accounts by @cspiel1 in https://github.com/baresip/baresip/pull/1660
* ctrl_dbus: print command with the warning by @cspiel1 in https://github.com/baresip/baresip/pull/1662
* call: new transfer call state to handle transfered calls correctly by @cHuberCoffee in https://github.com/baresip/baresip/pull/1658
* serreg: prevent fast register retries if offline by @cspiel1 in https://github.com/baresip/baresip/pull/1663
* av1: update packetization code by @alfredh in https://github.com/baresip/baresip/pull/1657
* call: magic check in sipsess_desc_handler() by @cspiel1 in https://github.com/baresip/baresip/pull/1664
* alsa: use snd_pcm_drop instead of snd_pcm_drain by @sreimers in https://github.com/baresip/baresip/pull/1669
* Increased debian compat level to 10 by @juha-h in https://github.com/baresip/baresip/pull/1667
* conf: fix conf_configure_buf() config parse by @sreimers in https://github.com/baresip/baresip/pull/1666
* stream flush rtp socket by @cspiel1 in https://github.com/baresip/baresip/pull/1671
* Transfer like rfc5589 by @cHuberCoffee in https://github.com/baresip/baresip/pull/1678
* GTK: mem_derefer call earlier by @mbattista in https://github.com/baresip/baresip/pull/1682
* netroam: add fail counter and event by @cspiel1 in https://github.com/baresip/baresip/pull/1685
* Added API functions stream_metric_get_(tx|rx)_bitrate by @juha-h in https://github.com/baresip/baresip/pull/1686
* Multicast new functions by @cHuberCoffee in https://github.com/baresip/baresip/pull/1687
* avcodec: Enable pass-through for more codecs by @abrodkin in https://github.com/baresip/baresip/pull/1692
* menu: filter for the correct call state in menu_selcall by @cHuberCoffee in https://github.com/baresip/baresip/pull/1693
* test: fix warning on mingw32 by @alfredh in https://github.com/baresip/baresip/pull/1696
* menu: Play ringback in play device by @myrkr in https://github.com/baresip/baresip/pull/1698
* sip: add optional TCP source port by @cspiel1 in https://github.com/baresip/baresip/pull/1695
* rtpext: change id unsigned -> uint8_t by @alfredh in https://github.com/baresip/baresip/pull/1701
* ci: add mingw build test by @sreimers in https://github.com/baresip/baresip/pull/1700
* test: use mediaenc srtp instead of mock by @alfredh in https://github.com/baresip/baresip/pull/1702
* test: remove mock mediaenc by @alfredh in https://github.com/baresip/baresip/pull/1704
* descr: add session_description by @alfredh in https://github.com/baresip/baresip/pull/1706
* use fs_isfile() by @alfredh in https://github.com/baresip/baresip/pull/1709
* stream: only call rtp_clear for audio by @alfredh in https://github.com/baresip/baresip/pull/1710
* checks if call is available before calling call, closes #1708 by @mbattista in https://github.com/baresip/baresip/pull/1712
* conf: add conf_loadfile by @alfredh in https://github.com/baresip/baresip/pull/1713
* ice: remove ice_mode by @sreimers in https://github.com/baresip/baresip/pull/1714
* audio: use auframe in encode_rtp_send, ref #1699 by @alfredh in https://github.com/baresip/baresip/pull/1715
* Increased account's max video codec count from four to eight by @juha-h in https://github.com/baresip/baresip/pull/1717
* gtk: Avoid duplicate call_timer registration by @myrkr in https://github.com/baresip/baresip/pull/1719
* Attended call transfer by @cHuberCoffee in https://github.com/baresip/baresip/pull/1718
* menu: exclude given call when searching for active call by @cspiel1 in https://github.com/baresip/baresip/pull/1721
* menu: play call waiting tone on audio_player device by @cspiel1 in https://github.com/baresip/baresip/pull/1722
* ci/build/macos: link ffmpeg@4 by @sreimers in https://github.com/baresip/baresip/pull/1725
* module auresamp by @cspiel1 in https://github.com/baresip/baresip/pull/1705
* test: remove h264 testcode, already in retest by @alfredh in https://github.com/baresip/baresip/pull/1726
* h265: move from avcodec to rem by @alfredh in https://github.com/baresip/baresip/pull/1728
* mc: send more details at receiver - timeout event by @cHuberCoffee in https://github.com/baresip/baresip/pull/1731
* h265: move packetizer from avcodec to rem by @alfredh in https://github.com/baresip/baresip/pull/1732
* FFmpeg 5 by @sreimers in https://github.com/baresip/baresip/pull/1734
* Fixing clang ThreadSanitizer warnings by @sreimers in https://github.com/baresip/baresip/pull/1730
* auresamp: replace anonymous union for pre C11 compilers by @cspiel1 in https://github.com/baresip/baresip/pull/1738
* aufile: align naming of alloc handlers by @sreimers in https://github.com/baresip/baresip/pull/1739
* auresamp fixes by @cspiel1 in https://github.com/baresip/baresip/pull/1741
* mc: new priority handling with multicast state by @cHuberCoffee in https://github.com/baresip/baresip/pull/1740
* remove support for Solaris platform by @alfredh in https://github.com/baresip/baresip/pull/1745
* Allow hanging up call that has not been ACKed yet by @juha-h in https://github.com/baresip/baresip/pull/1747
* Multicast identical condition and fmt string fix by @cHuberCoffee in https://github.com/baresip/baresip/pull/1751
* audio: allocate aubuf before ausrc_alloc (fixes data race) by @sreimers in https://github.com/baresip/baresip/pull/1748
* call: send supported header for 200 answering/ok by @cHuberCoffee in https://github.com/baresip/baresip/pull/1752
* event: check if media line is present for encoding audio/video dir by @cspiel1 in https://github.com/baresip/baresip/pull/1754
* Removed unused variable in modules/webrtc_aec/aec.cpp by @juha-h in https://github.com/baresip/baresip/pull/1756
* audio use module auconv by @cspiel1 in https://github.com/baresip/baresip/pull/1742
* test: use aufile module by @alfredh in https://github.com/baresip/baresip/pull/1757
* x11grab: remove module, use avformat.so instead by @alfredh in https://github.com/baresip/baresip/pull/1758
* audio: declare iterator inside for-loop (C99) by @alfredh in https://github.com/baresip/baresip/pull/1759
* aufile: set run=true before write thread starts (#1727) by @cspiel1 in https://github.com/baresip/baresip/pull/1762
* Added new API function call_supported() and used it in menu module by @juha-h in https://github.com/baresip/baresip/pull/1761
* aufile: separate aufile_src.c from aufile.c by @cspiel1 in https://github.com/baresip/baresip/pull/1765
* ctrl_dbus: fix possible data race (#1727) by @cspiel1 in https://github.com/baresip/baresip/pull/1764
* menu select other call on hangup by @cspiel1 in https://github.com/baresip/baresip/pull/1763
* event: encode also combined media direction by @cspiel1 in https://github.com/baresip/baresip/pull/1766
== New Contributors
* @srperens made their first contribution in https://github.com/baresip/baresip/pull/1399
* @negbie made their first contribution in https://github.com/baresip/baresip/pull/1451
* @andreaswatch made their first contribution in https://github.com/baresip/baresip/pull/1512
* @viordash made their first contribution in https://github.com/baresip/baresip/pull/1553
* @abrodkin made their first contribution in https://github.com/baresip/baresip/pull/1692
* @myrkr made their first contribution in https://github.com/baresip/baresip/pull/1698
---
To see a diff of this commit:
https://wip.pkgsrc.org/cgi-bin/gitweb.cgi?p=pkgsrc-wip.git;a=commitdiff;h=498aca0dea7667a1c863928a8c3f73fa747cc6e2
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
diffstat:
baresip/Makefile | 2 +-
baresip/PLIST | 17 ++----
baresip/distinfo | 8 +--
baresip/options.mk | 68 +++--------------------
baresip/patches/patch-modules_ilbc_ilbc.c | 23 --------
baresip/patches/patch-modules_portaudio_module.mk | 15 +++++
6 files changed, 33 insertions(+), 100 deletions(-)
diffs:
diff --git a/baresip/Makefile b/baresip/Makefile
index 8f390b2f7f..79c94afb83 100644
--- a/baresip/Makefile
+++ b/baresip/Makefile
@@ -1,6 +1,6 @@
# $NetBSD: Makefile,v 1.2 2014/09/05 08:06:00 thomasklausner Exp $
-DISTNAME= baresip-1.1.0
+DISTNAME= baresip-2.5.1
CATEGORIES= net audio
MASTER_SITES= ${MASTER_SITE_GITHUB:=baresip/}
GITHUB_TAG= v${PKGVERSION_NOREV}
diff --git a/baresip/PLIST b/baresip/PLIST
index c77956afb0..7be1b7ad1e 100644
--- a/baresip/PLIST
+++ b/baresip/PLIST
@@ -4,17 +4,17 @@ lib/baresip/modules/account.so
${PLIST.alsa}lib/baresip/modules/alsa.so
${PLIST.opencore-amr}lib/baresip/modules/amr.so
lib/baresip/modules/aubridge.so
+lib/baresip/modules/auconv.so
lib/baresip/modules/aufile.so
-lib/baresip/modules/auloop.so
+lib/baresip/modules/auresamp.so
lib/baresip/modules/ausine.so
${PLIST.ffmpeg}lib/baresip/modules/avcodec.so
${PLIST.ffmpeg}lib/baresip/modules/avformat.so
-lib/baresip/modules/b2bua.so
-${PLIST.cairo}lib/baresip/modules/cairo.so
lib/baresip/modules/cons.so
lib/baresip/modules/contact.so
lib/baresip/modules/ctrl_tcp.so
lib/baresip/modules/debug_cmd.so
+lib/baresip/modules/dtls_srtp.so
lib/baresip/modules/ebuacip.so
lib/baresip/modules/echo.so
lib/baresip/modules/fakevideo.so
@@ -22,21 +22,18 @@ lib/baresip/modules/g711.so
lib/baresip/modules/g722.so
lib/baresip/modules/g726.so
lib/baresip/modules/gsm.so
-${PLIST.gstreamer}lib/baresip/modules/gst1.so
-${PLIST.gstreamer}lib/baresip/modules/gst_video1.so
${PLIST.gtk}lib/baresip/modules/gtk.so
lib/baresip/modules/httpd.so
lib/baresip/modules/ice.so
-${PLIST.ilbc}lib/baresip/modules/ilbc.so
${PLIST.jack}lib/baresip/modules/jack.so
lib/baresip/modules/l16.so
lib/baresip/modules/menu.so
lib/baresip/modules/mixausrc.so
+lib/baresip/modules/mixminus.so
lib/baresip/modules/multicast.so
lib/baresip/modules/mwi.so
lib/baresip/modules/natpmp.so
${PLIST.opus}lib/baresip/modules/opus.so
-${PLIST.oss}lib/baresip/modules/oss.so
lib/baresip/modules/plc.so
${PLIST.portaudio}lib/baresip/modules/portaudio.so
lib/baresip/modules/presence.so
@@ -46,22 +43,18 @@ ${PLIST.sdl2}lib/baresip/modules/sdl.so
lib/baresip/modules/selfview.so
lib/baresip/modules/serreg.so
${PLIST.sndfile}lib/baresip/modules/sndfile.so
-${PLIST.speex}lib/baresip/modules/speex_pp.so
lib/baresip/modules/srtp.so
lib/baresip/modules/stdio.so
lib/baresip/modules/stun.so
lib/baresip/modules/turn.so
lib/baresip/modules/uuid.so
${PLIST.v4l2}lib/baresip/modules/v4l2.so
-${PLIST.v4l2}lib/baresip/modules/v4l2_codec.so
lib/baresip/modules/vidbridge.so
-${PLIST.cairo}lib/baresip/modules/vidinfo.so
-lib/baresip/modules/vidloop.so
+lib/baresip/modules/vidinfo.so
${PLIST.libvpx}lib/baresip/modules/vp8.so
${PLIST.libvpx}lib/baresip/modules/vp9.so
lib/baresip/modules/vumeter.so
${PLIST.x11}lib/baresip/modules/x11.so
-${PLIST.x11}lib/baresip/modules/x11grab.so
share/baresip/autoanswer.wav
share/baresip/busy.wav
share/baresip/callwaiting.wav
diff --git a/baresip/distinfo b/baresip/distinfo
index 59d3dad3d7..94fab27cc1 100644
--- a/baresip/distinfo
+++ b/baresip/distinfo
@@ -1,7 +1,7 @@
$NetBSD: distinfo,v 1.1 2014/08/10 00:26:01 thomasklausner Exp $
-RMD160 (baresip-1.1.0.tar.gz) = b53256dfa00c874e2789da7f5c5b489d722a5192
-SHA512 (baresip-1.1.0.tar.gz) = 82616ddfb344c4a48f742a92e9fcdc1fdd3b281950fadee0f3c3c6401d6f31e2232e9a64e5aa0bd8fc54dec02ad4c4573ff6c5a71c0929d89f83e136d35f2a3a
-Size (baresip-1.1.0.tar.gz) = 1105338 bytes
-SHA1 (patch-modules_ilbc_ilbc.c) = a2a7d685c4989bf910a9d5b8582d1261fce32e1c
+BLAKE2s (baresip-2.5.1.tar.gz) = 530e58d1d39c9a8ef072bb844afa43e896cf3432eda6a517142a3ed37d989952
+SHA512 (baresip-2.5.1.tar.gz) = 35f035aed738efdb8b7e362c1ec269c9315a63198dc5b5feba03a5a16d1ca154b098a01c8ad482ee2998d40e871cfa2cab6c2fa913f1a7889009eb93fc2538e4
+Size (baresip-2.5.1.tar.gz) = 1122422 bytes
+SHA1 (patch-modules_portaudio_module.mk) = 13e2c0cf2765ea4a1e95f3b3e3bea930b89d29b8
SHA1 (patch-modules_v4l2_v4l2.c) = 71ba2d1e5c8ba61eb011bd2b6b9e0d9cdaec5797
diff --git a/baresip/options.mk b/baresip/options.mk
index e285ed1019..85c04eea1f 100644
--- a/baresip/options.mk
+++ b/baresip/options.mk
@@ -1,11 +1,10 @@
# $NetBSD$
PKG_OPTIONS_VAR= PKG_OPTIONS.baresip
-PKG_SUPPORTED_OPTIONS= alsa cairo ffmpeg gstreamer gtk ilbc jack
-PKG_SUPPORTED_OPTIONS+= libvpx oss opencore-amr opus portaudio
-PKG_SUPPORTED_OPTIONS+= pulseaudio sdl2 sndfile sndio speex v4l2
-PKG_SUPPORTED_OPTIONS+= x11
-PKG_SUGGESTED_OPTIONS= oss ilbc speex
+PKG_SUPPORTED_OPTIONS= alsa ffmpeg gtk jack libvpx
+PKG_SUPPORTED_OPTIONS+= opencore-amr opus portaudio pulseaudio
+PKG_SUPPORTED_OPTIONS+= sdl2 sndfile sndio speex v4l2 x11
+PKG_SUGGESTED_OPTIONS= portaudio
.include "../../mk/bsd.prefs.mk"
.include "../../mk/bsd.options.mk"
@@ -23,38 +22,13 @@ MAKE_FLAGS+= USE_ALSA=yes
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"
+.include "../../x11/gtk3/buildlink3.mk"
.else
MAKE_FLAGS+= USE_GTK=
.endif
@@ -66,27 +40,15 @@ MAKE_FLAGS+= USE_GTK=
PLIST.ffmpeg= yes
MAKE_FLAGS+= USE_AVCODEC=yes
MAKE_FLAGS+= USE_AVFORMAT=yes
-LFLAGS+= ${COMPILER_RPATH_FLAG}${BUILDLINK_PREFIX.ffmpeg4}/${BUILDLINK_LIBDIRS.ffmpeg4}
-LFLAGS+= -L${BUILDLINK_PREFIX.ffmpeg4}/${BUILDLINK_LIBDIRS.ffmpeg4}
-.include "../../multimedia/ffmpeg4/buildlink3.mk"
+LFLAGS+= ${COMPILER_RPATH_FLAG}${BUILDLINK_PREFIX.ffmpeg5}/${BUILDLINK_LIBDIRS.ffmpeg5}
+LFLAGS+= -L${BUILDLINK_PREFIX.ffmpeg5}/${BUILDLINK_LIBDIRS.ffmpeg5}
+.include "../../multimedia/ffmpeg5/buildlink3.mk"
.include "../../multimedia/x264-devel/buildlink3.mk"
.else
MAKE_FLAGS+= USE_AVCODEC=
MAKE_FLAGS+= USE_AVFORMAT=
.endif
-###
-### ILBC support (audio codec)
-###
-.if !empty(PKG_OPTIONS:Milbc)
-PLIST.ilbc= yes
-MAKE_FLAGS+= USE_ILBC=yes
-#.include "../../wip/ilbc-rfc3951/buildlink3.mk"
-.include "../../wip/libilbc/buildlink3.mk"
-.else
-MAKE_FLAGS+= USE_ILBC=
-.endif
-
###
### Jack Audio Connection Kit support (audio output)
###
@@ -109,16 +71,6 @@ MAKE_FLAGS+= USE_VPX=yes
MAKE_FLAGS+= USE_VPX=
.endif
-###
-### OSS support (audio output)
-###
-.if !empty(PKG_OPTIONS:Moss)
-PLIST.oss= yes
-MAKE_FLAGS+= USE_OSS=yes # full-duplex issues
-.else
-MAKE_FLAGS+= USE_OSS=
-.endif
-
###
### opencore-amr support (audio codec)
###
@@ -203,13 +155,9 @@ MAKE_FLAGS+= USE_SNDIO=
.if !empty(PKG_OPTIONS:Mspeex)
PLIST.speex= yes
MAKE_FLAGS+= HAVE_SPEEXDSP=yes
-MAKE_FLAGS+= USE_SPEEX_AEC=yes
-MAKE_FLAGS+= USE_SPEEX_PP=yes
.include "../../audio/speexdsp/buildlink3.mk"
.else
MAKE_FLAGS+= HAVE_SPEEXDSP=
-MAKE_FLAGS+= USE_SPEEX_AEC=
-MAKE_FLAGS+= USE_SPEEX_PP=
.endif
###
diff --git a/baresip/patches/patch-modules_ilbc_ilbc.c b/baresip/patches/patch-modules_ilbc_ilbc.c
deleted file mode 100644
index 2f2181fe5f..0000000000
--- a/baresip/patches/patch-modules_ilbc_ilbc.c
+++ /dev/null
@@ -1,23 +0,0 @@
-$NetBSD$
-
-Use libilbc. Almost no changes.
-
---- modules/ilbc/ilbc.c.orig 2019-06-22 07:51:28.000000000 +0000
-+++ modules/ilbc/ilbc.c
-@@ -6,9 +6,13 @@
- #include <re.h>
- #include <rem.h>
- #include <baresip.h>
--#include <iLBC_define.h>
--#include <iLBC_decode.h>
--#include <iLBC_encode.h>
-+#include <ilbc.h>
-+#include <string.h>
-+
-+#define NO_OF_BYTES_20MS 38
-+#define NO_OF_BYTES_30MS 50
-+#define NO_OF_WORDS_20MS 19
-+#define NO_OF_WORDS_30MS 25
-
-
- /**
diff --git a/baresip/patches/patch-modules_portaudio_module.mk b/baresip/patches/patch-modules_portaudio_module.mk
new file mode 100644
index 0000000000..e65369a4bb
--- /dev/null
+++ b/baresip/patches/patch-modules_portaudio_module.mk
@@ -0,0 +1,15 @@
+$NetBSD$
+
+Use pkg-config to find the portaudio library.
+
+--- modules/portaudio/module.mk.orig 2022-07-17 06:50:49.000000000 +0000
++++ modules/portaudio/module.mk
+@@ -6,6 +6,7 @@
+
+ MOD := portaudio
+ $(MOD)_SRCS += portaudio.c
+-$(MOD)_LFLAGS += -lportaudio
++$(MOD)_CFLAGS += $(shell pkg-config --cflags portaudio-2.0)
++$(MOD)_LFLAGS += $(shell pkg-config --libs portaudio-2.0)
+
+ include mk/mod.mk
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