| Commit message (Collapse) | Author | Age |
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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 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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1. for a given UA, each call has a unique line number starting
from 1
2. the list of calls is sorted in an arbitrary order, but the
last list element indicates "current call"
3. the '@' command takes a numeric argument which is the
line-number of the wanted call
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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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thanks to Gary Metalle for testing
ref. issue #111
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this fixes #79
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