jack_play: connect all physical ports #1028
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agramner
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Jul 1, 2020
* jack_play: connect all physical ports * Fix ccheck errors * Fix more ccheck errors
wip-sync
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Nov 28, 2020
(Presumably the changelog [Unreleased] section is for 1.0.0) = Baresip Changelog == [Unreleased] === Added - aac: add AAC_STREAMTYPE_AUDIO enum value - aac: add AAC_ prefix - Video mode param to call_answer(), ua_answer() and ua_hold_answer [#966] - video_stop_display() API function [#977] - module: add path to module_load() function - conf: add conf_configure_buf - test: add usage of g711.so module [#978] - JSON initial codec state command and response [#973] - account_set_video_codecs() API function [#981] - net: add fallback dns nameserver [#996] - gtk: show call_peername in notify title [#1006] - call: Added call_state() API function that returns enum state of the call [#1013] - account_set_stun_user() and account_set_stun_pass() API functions [#1015] - API functions account_stun_uri and account_set_stun_uri. [#1018] - ausine: Audio sine wave input module [#1021] - gtk/menu: replace spaces from uri [#1007] - jack: allowing jack client name to be specified in the config file [#1025] [#1020] - snapshot: Add snapshot_send and snapshot_recv commands [#1029] - webrtc_aec: 'extended_filter' config option [#1030] - avfilter: FFmpeg filter graphs integration [#1038] - reg: view proxy expiry value in reg_status [#1068] - account: add parameter rwait for re-register interval [#1069] - call, stream, menu: add cmd to set the direction of video stream [#1073] === Changed - **Using [baresip/re](https://github.com/baresip/re) fork now** - audio: move calculation to audio_jb_current_value - avformat: clean up docs - gzrtp: update docs - account: increased size of audio codec list to 16 - video: make video_sdp_attr_decode public - config: Derive default audio driver from default audio device [#1009] - jack: modifying info message on jack client creation [#1019] - call: when video stream is disabled, stop also video display [#1023] - dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048 [#1062] [#1056] - rst: use a min ptime of 20ms - aac: change ptime to 4ms === Fixed - avcodec: fix H.264 interop with Firefox - winwave: waveInGetPosition is no longer supported for use as of Windows Vista [#960] - avcodec: call av_hwdevice_ctx_create before if-statement - account: use single quote instead of backtick - ice: fix segfault in connh [#980] - call: Update call->got_offer when re-INVITE or answer to re-INVITE is received [#986] - mk: Test also for /usr/lib64/libspeexdsp.so to cover Fedora/RHEL/CentOS [#992] - config: Allow distribution specific CA trust bundle locations (fixes [#993] - config: Allow distribution specific default audio device (fixes [#994] - mqtt: fix err is never read (found by clang static analyzer) - avcodec: fix err is never read (found by clang static analyzer) - gtk: notification buttons do not work on Systems [#1012] - gtk: fix dtmf_tone and add tones as feedback [#1010] - pulse: drain pulse buffers before freeing [#1016] - jack: jack_play connect all physical ports [#1028] - Makefile: do not try to install modules if build is static [#1031] - gzrtp: media_alloc function is missing [#1034] [#1022] - call: when updating video, check if video stream has been disabled [#1037] - amr: fix length check, fixes [#1011] - modules: fix search path for avdevice.h [#1043] - gtk: declare variables C89 style - config: init newly added member - menu: fix segfault in ua_event_handler [#1059] [#1061] - debug_cmd: fix OpenSSL no-deprecated [#1065] - aac: handle missing bitrate parameter in SDP format - av1: properly configure encoder === Removed - ice: remove support for ICE-lite - ice: remove ice_debug, use log level DEBUG instead - ice: make stun server optional - config: remove ice_debug option (unused) === Contributors (many thanks) - Alfred E. Heggestad - Alexander Gramner - Andrew Webster - Christian Spielberger - Christoph Huber - Davide Alberani - Ethan Funk - Juha Heinanen - mbattista - Michael Malone - Mikl Kurkov - ndilieto - Robert Scheck - Roger Sandholm - Sebastian Reimers [#966]: baresip/baresip#966 [#977]: baresip/baresip#977 [#978]: baresip/baresip#978 [#973]: baresip/baresip#973 [#981]: baresip/baresip#981 [#996]: baresip/baresip#996 [#1006]: baresip/baresip#1006 [#1013]: baresip/baresip#1013 [#1015]: baresip/baresip#1015 [#1018]: baresip/baresip#1018 [#1021]: baresip/baresip#1021 [#1007]: baresip/baresip#1007 [#1025]: baresip/baresip#1025 [#1020]: baresip/baresip#1020 [#1029]: baresip/baresip#1029 [#1030]: baresip/baresip#1030 [#1038]: baresip/baresip#1038 [#1009]: baresip/baresip#1009 [#1019]: baresip/baresip#1019 [#1023]: baresip/baresip#1023 [#1062]: baresip/baresip#1062 [#1056]: baresip/baresip#1056 [#960]: baresip/baresip#960 [#980]: baresip/baresip#980 [#986]: baresip/baresip#986 [#992]: baresip/baresip#992 [#993]: baresip/baresip#993 [#994]: baresip/baresip#994 [#1012]: baresip/baresip#1012 [#1010]: baresip/baresip#1010 [#1016]: baresip/baresip#1016 [#1028]: baresip/baresip#1028 [#1031]: baresip/baresip#1031 [#1034]: baresip/baresip#1034 [#1022]: baresip/baresip#1022 [#1037]: baresip/baresip#1037 [#1011]: baresip/baresip#1011 [#1043]: baresip/baresip#1043 [#1059]: baresip/baresip#1059 [#1061]: baresip/baresip#1061 [#1065]: baresip/baresip#1065 [#1068]: baresip/baresip#1068 [#1069]: baresip/baresip#1069 [#1073]: baresip/baresip#1073 [Unreleased]: baresip/baresip@v0.6.6...HEAD
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When for example making a call with mono audio with 2 physical jack playback ports the audio was only routed to the left channel. This fixes the problem by connecting all physical playback ports to at least one of the baresip registered ports.
Also added JackPortIsPhysical flag when looking up jack ports.