| Commit message (Collapse) | Author | Age |
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config:
videnc_format yuv420p # yuv420p, yuv444p, ...
The default pixel format is still YUV420P
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Thank you, this is a very nice contribution.
Alfred
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config:
auenc_format s16 # s16, float
audec_format s16 # s16, float
modules: only opus module supports this for now.
default is s16
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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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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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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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* vidisp: add fullscreen flag
* Add video_fullscreen option
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* add RTP Header extension for Client-to-Mixer Audio Level Indication
https://tools.ietf.org/html/rfc6464
requires libre from this commit or later:
https://github.com/creytiv/re/commit/1544a1e375c76a80084b411d21b0431f95e9cdfb
* fix warnings
* fix warnings
* minor cleanup
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* config: list several modules that were missing
* config: hide daala, comment out echo and hide dtmfio for non-linux systems
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1. Added support for RTP timeout. The feature is disabled by default
and can be enabled with config "rtp_timeout N" where N is the number
of seconds of RTP inactivity. If this is detected, the call is
closed with a "special" SIP reason code of 701.
2. Added support for automatic re-connect in the menu module.
This can be enabled by setting the 2 config items:
redial_attempts 3
redial_delay 5
This work was contributed by Sveriges Radio. Thanks
goes to Ola Palm and Jim Eld.
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- The network instance is now in struct network and does not
use any local/static data
- A new top-level struct in baresip.c owns the single instance
of struct network
it is a long-term goal to remove all local/static data
from libbaresip and make it fully re-entrant.
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the config option is used for incoming calls, if the call
is not answered after X seconds. The default value is
120 seconds. If the value is set to 0 the timeout timer
is not enabled.
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While at it, also make spacing of this configuration item more consistent.
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- Slightly modified patch provided by Dmitrij D. Czarkoff
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- speex_quality/speex_complexity value is not mapped to sdp mode value
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http://en.wikipedia.org/wiki/Kilobit
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- remove FFmpeg keyword, as both FFmpeg and libav is supported
- avcodec.so need libavcodec
- avformat.so needs libavformat and others
- also made the "quicktime.so" module deprecated
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