baresip - Debian dgit repo for package baresip

	Commit message (Collapse)	Author	Age
*	audio: set the exact framesize for plc-handler	Alfred E. Heggestad	2016-05-17
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*	Merged with master and added proper RTP time stamp handling	Christian Hoene	2016-05-14
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\| *	minor nicify	Christian Hoene	2016-05-14
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\| *	Fixed bug of sending empty packets	Christian Hoene	2016-05-12
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* \|	aucodec: split srate into srate and crate (Clock Rate)	Alfred E. Heggestad	2016-05-01
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* \|	fix some ccheck warnings	Alfred E. Heggestad	2016-04-24
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*	Added resampler to decode	Christian Hoene	2016-04-22
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*	First version with MPA coding	Christian Hoene	2016-04-22
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*	fixes alfredh/baresip#93 glibc >=2.20 _BSD_SOURCE warning	Sebastian Reimers	2015-12-13
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*	audio: add milliseconds to log message	Alfred E. Heggestad	2015-12-13
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*	audio: flush tx-buffer for all tx-modes	Alfred E. Heggestad	2015-11-06
\| \| \| \| \|	thanks to Thibault Gueslin for reporting the bug and suggestion a nice fix
*	audio: add API to set source/player while in call	Alfred E. Heggestad	2015-10-11
\| \| \| \| \| \| \| \| \| \| \|	int audio_set_source(struct audio au, const char mod, const char device); int audio_set_player(struct audio au, const char mod, const char device); example usage: audio_set_player(call_audio(ua_call(uag_cur())), "alsa", "default"); fixes #70
*	audio: lookup pt for telephone-event (fixes #72)	Alfred E. Heggestad	2015-09-19
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*	audio: print module/device-name if ausrc_alloc fails	Alfred E. Heggestad	2015-08-02
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*	fix warnings on Mac OSX	Alfred E. Heggestad	2014-10-18
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*	audio: fix audio_srate range for G.722	Alfred E. Heggestad	2014-08-09
\| \| \| \|	thanks to Remigiusz Borek for reporting the issue
*	fix warning	Alfred E. Heggestad	2014-05-20
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*	rtpstat: fixup some formatting	Alfred E. Heggestad	2014-05-20
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*	Cleanup of XRTP stats print function	Lorenzo Mangani	2014-05-20
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*	Fixed JI value to timestamp-units based on codec audiorate	Lorenzo Mangani	2014-05-19
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*	Minor cleanup	Lorenzo Mangani	2014-05-18
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*	Move XRTP stats to audio and implement EN/DE parameters	Lorenzo Mangani	2014-05-18
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*	fix call mute-state for multiple calls	Alfred E. Heggestad	2014-05-18
\| \| \| \| \| \| \|	added audio_ismuted() and remove static state from menu.c thanks to Remik who reported the issue and suggested a nice solution :)
*	audio: fix timestamp calculation for sending	Alfred E. Heggestad	2014-05-17
\| \| \| \| \| \| \| \| \| \|	according to RFC 3551 the RTP timestamp is independent of number of channels and encoding. fix the timestamp calculation for outgoing RTP packets, by dividing the total number of samples on number of channels. the bug was not affecting calls with mono audio, but was very likely to affect calls with stereo audio (e.g. OPUS)
*	added some missing comments	Alfred E. Heggestad	2014-04-12
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*	ausrc/auplay api: change to 16-bit samples	Alfred E. Heggestad	2014-02-13
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*	remove fmt parameter from ausrc/auplay api	Alfred E. Heggestad	2014-02-13
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*	baresip 0.4.10	Alfred E. Heggestad	2014-02-09