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path: root/include/baresip.h
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* version 0.5.9Alfred E. Heggestad2018-04-15
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* update doxygen commentsAlfred E. Heggestad2018-04-12
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* vidsrc: add timestamp to the frame APIAlfred E. Heggestad2018-04-07
| | | | | | | | | | | | | | | | timestamps for v4l and v4l2 test: add timestamp to vidsrc mock v4l2: print framerate vidloop: save timestamps and print duration test with timebase of 65536 v4l2: scale timestamp to VIDEO_TIMEBASE minor fixes
* make struct vidsrc and vidisp publicAlfred E. Heggestad2018-04-02
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* test: add testcase for media encryptionAlfred E. Heggestad2018-03-29
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* timer: make tmr_jiffies_usec() publicAlfred E. Heggestad2018-03-21
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* b2bua: add handling of all inbound SIP requestsAlfred E. Heggestad2018-03-11
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* h264: add function to convert NAL unit type to a stringAlfred E. Heggestad2018-03-04
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* video: change video_fps from int to double floatAlfred E. Heggestad2018-02-25
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* vidcodec: change rtp_ts from 32-bit to 64-bitAlfred E. Heggestad2018-02-24
| | | | | | the extended RTP timestamp from the video encoder should now contain a full extended timestamp, that must never wrap.
* video: change RTP send timestamp to 64-bitAlfred E. Heggestad2018-02-23
| | | | | | | | | - change RTP timestamp for packet_handler from 32-bit to 64-bit (extended RTP timestamp). This timestamp must be monotonically increasing, and now the application does not have to handle wrapping of send timestamp. - save first/last RTP timestamp sent
* version 0.5.8Alfred E. Heggestad2018-02-11
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* log: rename log_enable_stderr to log_enable_stdoutAlfred E. Heggestad2018-02-11
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* config: add configurable video pixel formatAlfred E. Heggestad2018-02-10
| | | | | | | | config: videnc_format yuv420p # yuv420p, yuv444p, ... The default pixel format is still YUV420P
* Aucodec sample format (#352)Alfred E. Heggestad2018-02-03
| | | | | | | | | | | | | | | | * aucodec: merge s16 and fmt encode/decode into one having 2 different encode functions for s16 and other formats turned out to be a bit impractical. better to have 1 function where the sample format (fmt) can be specified. * update some aucodec modules * update test * update codec2 * update g7221 module
* call: add call_id accessorAlfred E. Heggestad2018-01-17
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* audio: sample format for audio encoder/decoderAlfred E. Heggestad2018-01-06
| | | | | | | | | | | config: auenc_format s16 # s16, float audec_format s16 # s16, float modules: only opus module supports this for now. default is s16
* audio: add function to set encoder bitrateAlfred E. Heggestad2018-01-05
| | | | | | | - auenc API updated with bitrate in bits/s - new command to set encoder bitrate: /aubitrate 128000
* baresip version 0.5.7Alfred E. Heggestad2017-12-25
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* event: move encoding of event from mqtt to event.c (ref #331)Alfred E. Heggestad2017-12-16
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* ua: add UA_EVENT_CALL_RTCP eventAlfred E. Heggestad2017-12-09
| | | | | | this event is called for every received RTCP report. mqtt: add JSON encoding of RTCP stats
* audio: remove support for AUDIO_MODE_TMRAlfred E. Heggestad2017-11-26
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* audio: remove support for tx realtime threadAlfred E. Heggestad2017-11-16
| | | | | | | | | | - 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.
* add support for specifying sample format (#317)Alfred E. Heggestad2017-11-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
* add support for EBU-ACIP parametersAlfred E. Heggestad2017-10-29
| | | | | | | | | | | 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
* version 0.5.6Alfred E. Heggestad2017-10-14
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* rename error() to error_msg() fixes #295Alfred E. Heggestad2017-09-15
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* ua: remove accessor to SIP file descriptor (fd)Alfred E. Heggestad2017-09-11
| | | | | | | - The fd is platform specific, and should not be exposed in the public API. (this function was used a long time ago for iOS applications)
* baresip version 0.5.5Alfred E. Heggestad2017-09-07
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* Video timestamp (#286)Alfred E. Heggestad2017-07-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * add timestamp to video encoder/decoder * update vp8 vp9 * random offset * revert decoder timestamp * cleanup * gst_video1: add timestamp * change timestamp to RTP-timestamp * update modules * timestamp things * save rtp timestamp * packet_handler: align params same as RTP header * align parameters same way like RTP-header * avcodec: calc ts for old FFmpeg versions * cleanup * fix gst avcodec: use correct AVPacket * minor cleanup
* video: add video_calc_rtp_timestamp()Alfred E. Heggestad2017-07-22
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* ui: make API re-entrantAlfred E. Heggestad2017-07-14
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* ui: remove ui_input(), use ui_input_key() insteadAlfred E. Heggestad2017-07-10
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* Make sdl2 Fullscreen Mode configurable (#279)Jonathan Sieber2017-07-10
| | | | | | * vidisp: add fullscreen flag * Add video_fullscreen option
* vidisp: add fullscreen flagAlfred E. Heggestad2017-07-09
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* account: move password prompt to moduleAlfred E. Heggestad2017-07-09
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* account: added accessor function to AORAlfred E. Heggestad2017-07-09
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* contact: rename 'remove' to avoid clash with REMOVE(3)Alfred E. Heggestad2017-07-09
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* Presence: Handle contacts added at run-time (#276)Jonathan Sieber2017-07-06
| | | | | | | | * Add contact_presence getter * Add contact_remove() function * presence: Handle contacts added at runtime
* add commands to load/unload module (#275)Alfred E. Heggestad2017-07-06
| | | | | | | | * add commands to load/unload module * fix warning * module: update doxygen
* release version 0.5.4Alfred E. Heggestad2017-06-23
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* ua: add ua_progress() and testcaseAlfred E. Heggestad2017-06-03
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* add RTP Header extension for Client-to-Mixer Audio Level Indication (#264)Alfred E. Heggestad2017-06-02
| | | | | | | | | | | | | | | | * 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
* vidfilt: make the API re-entrantAlfred E. Heggestad2017-05-26
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* Vidisp api reentrant (#258)Alfred E. Heggestad2017-05-26
| | | | | | * vidisp: make the API re-entrant * vidisp: update all modules to re-entrant API
* Vidsrc api reentrant (#256)Alfred E. Heggestad2017-05-23
| | | | | | * vidsrc: make the API re-entrant * vidsrc: update all modules to new API
* vidcodec: make the API re-entrantAlfred E. Heggestad2017-05-20
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* release version 0.5.3Alfred E. Heggestad2017-05-14
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* aulevel: calculate audio levels in dBovAlfred E. Heggestad2017-04-27
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* release baresip 0.5.2Alfred E. Heggestad2017-04-07
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