| Commit message (Collapse) | Author | Age |
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mqtt: relay ua_event with publish
relay incoming message to long commands (with '/' prefix)
mqtt: add file for publish
mqtt: add command response
mqtt: add subscribe.c
mqtt: use fd_listen for READ/WRITE
Make + Config entry for MQTT module (#329)
* include mqtt in modules and config
* Check for mosquitto-dev
* fix mqtt module checks
* Moving MQTT config to module_app block
mqtt: add JSON payload for events
mqtt: add JSON decoding of command
mqtt: add config for broker host/port
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audio_txmode poll|thread
1. "poll" -- uses the audio driver thread to encode and
send RTP packets (default)
2. "thread" -- uses a dedicated thread to encode and send
RTP packets
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- add config "ebuacip_jb_type auto|fixed"
- add ebuacip qosrec
thanks to Ola Palm for the patch
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- there is no need to run this thread in "realtime" mode,
as the network will add much more jitter compared to
an internal thread.
(Setting a high thread priority may be a different option)
the only modules that needs Real-time is the drivers
close to I/O such as audio/video input/output drivers.
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It can be called from another thread (audio/video read).
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from Apple website:
"Deprecated in OS X v10.9. Use AVFoundation.framework and AVKit.framework
instead."
Ref: https://developer.apple.com/library/content/documentation/MacOSX/Conceptual/OSX_Technology_Overview/SystemFrameworks/SystemFrameworks.html
please use qtcapture or avcapture modules instead
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Having min_sz > tx->psize in aubuf_alloc() results in insertion of 16
packets of silence at the beginning of the RTP stream. This adds an
unnecessary delay.
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API: ausrc and auplay
- add config items for ausrc/auplay format:
ausrc_format s16|float
auplay_format s16|float
- audio.c: convert audio samples to/from signed 16-bit
Modules:
alsa
add test for sample format FLOAT
rst: add support for FLOAT sample format
audiounit: add support for FLOAT sample format
coreaudio: check for signed 16-bit audio format
oss: check for signed 16-bit sample format
winwave: check for S16LE
pulse: add support for FLOAT sample format
sndio: check for S16 format
gst1: check sample format
aufile: check sample format
aubridge: check sample format
gst: check sample format
opensles: check for S16 sample format
jack: check sample format
alsa: remove usage of local config
test: change samples to void pointer
test: change sample type to void pointer
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opus_stereo yes -- Enable stereo (default)
opus_stereo no -- Disable stereo
thanks to Ola Palm for the original patch
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Add an option to use libre SRTP processing stuff instead of the
one provided by ZRTPCPP (-DGZRTP_USE_RE_SRTP).
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add support for EBU-ACIP parameters in the SDP. This can be
enabled with the following config item:
sdp_ebuacip yes
Reference: https://tech.ebu.ch/docs/tech/tech3368.pdf
Thanks to Ola Palm from Swedish Radio for the patch
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Use SDP attribute a=zrtp-hash to protect the Hello message (RFC 6189,
section 8.1).
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Incoming RTCP packet's encrypt-flag is checked in libzrtp.
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the logo was designed by E.J.C. Lindner in collaboration
with Baresip community. thanks!
ref #232
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* modules/opensles: fix recorder speaker setup
- if there is more than one channel, use SL_SPEAKER_FRONT_LEFT |
SL_SPEAKER_FRONT_RIGHT speakers
x# Please enter the commit message for your changes. Lines starting
* modules/zrtp: make travis happy
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... with previously verified remote peer
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