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|
2019-12-01 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.5
Alfred E. Heggestad (138):
mnat: add struct mnat pointer to session handler
ice: add ice-lite, move to per-account config
modules: check return value from uag_event_register()
menu: check return value of account_set_answermode
ua: move ua_print_sip_status to debug_cmd module
pcp: updated mnat api
modules: rename gst1.so to gst.so
account: make answermode code more robust
bfcp: remove code
ua: remove uag_tls()
sdl: add support for YUYV422 pixel format
modules: rename gst_video1.so to gst_video.so
sdl: add support for UYVY422 pixel format
ua: fix whitespace
test: mock_mnat_register return void
stream: debug tuning
test: enable wait_connected flag on mock mnat
dtls_srtp: dont store remote address on the state
test: enable wait_secure flag on mock mediaenc
modules: rename sdl2.so to sdl.so
menc: sort handlers in logical order
audiounit: use error ENOTSUP if AudioSessionSetActive fails
test: add webrtc test-case
audiounit: check return value of AudioUnitSetProperty()
bv32: remove module (#793)
audiounit: fix samplerate for iOS
config: add snd_path to template
account: remove check for deprecated password
audio: use dynamically allocated string for device name
Update README.md
avcodec: remove unused prototype
avcodec: fix unused parameter warning
move h265.so into avcodec.so
Update .travis.yml (#798)
Account specific audio source and playback (#796)
mpa: switch encoder to use lame (#797)
mk: remove GPROF
ua: add support for SIP trace (#804)
AAC codec (#805)
audio: set the correct variable to false if pthread_create() fails
sdl: properly close window (OSX)
ua: fix warning
rtcpsummary: use call object from event handler
test: move aucodec list one level up
video: add vidcodec accessor
refactoring; move rtp stats code to separate .c file
rtpstat cleanup
call: check magic
vidinfo: add video overlay box with decoder info
vidinfo: fix compiler warning on linux
vidinfo: fix compiler warning on Android
stream: rename to stream_set_session_handlers()
Fix osx build (#809)
vidinfo: delete old file
test: move ausrc list one level up
test: move auplay list one level up
test: move aufilt list one level up
video: use vidcodec's list in video_decoder_set()
stream: add stream-list to stream/audio/video API
vidinfo: remove pixelformat, add packetloss
mk: add detection in SYSROOT_LOCAL
dtls_srtp: add media name and component type to logline
mnat: make mnat_find() public
audio: print name in parenthesis if not set
video: move vidfilt list one level up
audio: remove hack for starting source/player first
video: set stream samplerate in alloc
video: fix potential use of free'd string
ice: fix documentation
g711: use designated initialisers
g722: use designated initialisers
g726: use designated initialisers
ilbc: use designated initialisers
mk: check for ilbc in
gsm: use designated initialisers
amr: fix detection in SYSROOT_LOCAL
amr: use designated initialisers
isac: use designated initialisers
test: add audio_codecs to account testcase
win32: sort module exports in alphabetical order
win32: add debug_cmd to static list of modules
echo: no need to use uag_current() -- ref #815
mk: detect aac in SYSROOT_LOCAL
modules: use designated initializers
modules: use designated initializers
v4l2_codec: use designated initializers
wincons: use designated initializers
gzrtp: use designated initializers
avcodec: use designated initializers
video: request keyframe during packet-loss
mk: detect mqtt.so in SYSROOT_LOCAL
menu: fix formatting
menu: clean up usage of uag_current() -- ref #815
menu: save UA aor for redialing
avcodec: use AVFrame key_frame flag to check for keyframes (#830)
call: add call_find_id() -- ref #815
menu: new command /callfind -- ref #815
ice: make username/password optional
mqtt: add ua/call selection -- fixes #815
stream: no RTCP socket for mediaenc, if muxed
mqtt: encode response with JSON -- fix #826
main: change help text for -4 and -6 (ref #834)
docs: thanks to @premultiply
net: change prefer_ipv6 to int af, fixes #834
pulse: fix log text
avformat: fix build on Debian 8
stream: log more details
stream: make stream_start_mediaenc() public for core
stream: move start_mediaenc to call.c
audio: add ptime to struct aucodec (#849)
README: add i2s module
docs: thanks to Christian Spielberger
test: remove unused setting of int err
stream: remove code not executed
message: no need to check err
net: fix potential deref of NULL pointer
vidinfo: no need to check err
natpmp: no need to check err
ilbc: no need to check err
ctrl_tcp: no need to check oe_cmd here
avformat: remove got_pict hack
avcodec: minor fixes
sdl: remove int err, not needed
pulse: check the correct pointer
v4l2_codec: remove int err, not needed
alsa: store return value in a long
avcodec: copy key_frame flag from hardware frame
stream: make some functions public
ctrl_tcp: restore mbuf pos on errors
metric: add lock for multi-threading
sdl: skip plane if wstep is zero
aubridge: clear pointers after thread has exited
fix doxygen comments
av1: set allow_lowbitdepth to get correct pixel-format
bump version to 0.6.5
stream: check return value of metric_init()
stream: update doxygen comments
update doxygen comments
Christian Spielberger (2):
aui2s: add rtos i2s audio driver module (#848)
i2s: add doxygen defgroup header (#850)
Juha Heinanen (3):
- Added 'net_set_address' and 'net_set_af' API functions.
Added safety check to net_set_address() API function.
- Added AF_UNSPEC to supported net_set_af families.
juha-h (4):
- Exposed net_dns_debug function to API. (#791)
Added possibility to include ";extra" parameter to an account and access (#803)
- Search include files also from opencore-amr source directory (#822)
Merge pull request #843 from alfredh/net_stuff
premultiply (5):
Unify response list header layout (#821)
More common list format (#823)
MPA fmtp mirroring (#837)
MPA layer 3 encoding fixes (#839)
MPA L2 and L3 encoding (#844)
trampster (2):
Pass NULL to pa_simple_new if no device specified to indicate to PulseAudio we want to use the default (#785)
Add support for aes_256_gcm (#790)
2019-09-01 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.4
Aleksei (1):
Update MSVS project (Remove mos.c) (#744)
Alfred E. Heggestad (77):
test: added testcase for RTCP
speex_pp: handle changes in frame_size
mos: remove code (unused) (#739)
speex_aec: deprecate and remove module (#740)
aufilt: remove ptime parameter
audio: add last sample count
audio: remove recv ptime and pt from debug
stream: common function for start mediaenc
stream: added set/is_secure
test: sort tests in alphabetical order
test: added mock aufilt and testcase
mnat: add media connected handler
ice: use connected handler
stream: add mnat_connected_handler
call: split call start into audio/video
stream: print mnat_connected
dtls_srtp: remove 100ms timer
stream: add stream_start()
call: split update_media into audio/video
srtp: use designated initializers to init struct menc
zrtp: use designated initializers to init struct menc
dtls_srtp: use designated initializers to init struct menc
mnat: add wait_connected flag (#752)
menc: add wait_secure flag (#754)
opus: add opus_packet_loss config
audio: fix rtp timestamps for opus mono (ref #753)
webrtc interop (#756)
avcodec: remove compile time check (LIBAVUTIL_VERSION_INT)
avcodec: minor cleanup
avcodec: define KEYFRAME_INTERVAL
travis: use ubuntu xenial (#764)
remove PIX_FMT wrapper for old ffmpeg
stream: add lost count to RTP handler
aucodec: add buffer to packet-loss handler
add support for Opus FEC (WIP) (#755)
test: explicitly set BEHAVIOUR_ANSWER
test: set default action to ACTION_RECANCEL
x11: remove support for 16-bit RGB colors
sdl2: use vidisp name 'sdl'
stream: reorder functions
ice: check argument
avcodec: remove decoder framerate
stream: add media name to debug
test: remove testcase for C++
avcodec: remove log-line
avcodec: remove old ffmpeg wrapper for AVCodecID
test: add media-line to mnat mock
vidsrc: add wanted pixel format to parameters
stream: add handler for mnat connected
vp9: decode V and P fields
opus: update docs (ref #768)
test: use pixel format from api
test: use pixel format YUV420P
fix warning on linux
video: fix warning
test: fix warnings
dtmfio: fix warning
opus_multistream: fix warnings
Avcodec hwaccel (#770)
fakevideo: use pixel format from parameters
sdl2: add support for pixel format NV21
bump version to 0.6.4
mk: update Doxyfile
gst1: remove hard-coded uri
config: update default config
config: use cloudflare as sample DNS servers
config: print default hwaccel for avcodec.so module
dtls_srtp: make it more robust
test: use LD to link selftest
avformat: remove check for avformat >= 53.4.0
video: remove video_view
avformat: better logging
config: add mqtt template (ref #780)
Update README.md
docs: refresh example config
config: remove openl.so from template
README: fix typo
Timmo Verlaan (1):
menu: sndcode should signal release of key (#749)
seamus (1):
menu: add answermode command (#779)
trampster (1):
Allow CALL_CLOSED to be raised when call_id is not set. (#748)
2019-06-22 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.3
Alfred E. Heggestad (99):
baresip: remove prefer_ipv6 from api, use config instead
ua: remove prefer_ipv6 from api, use config instead
audio: allocate mbuf for encoded telephony events
net: remove af from api, use config instead
gst: remove old module, use gst1 instead
gst_video: remove old module, use gst_video1 instead
gst1: update comment
httpd: update comment
call: remove unused constant
reg: print address family of registration
ua: clean up prefer_ipv6 code
test: disable test for AUDIO_MODE_THREAD
config: remove old check for rtcp_enable
config: remove unused macro SA_INIT
config: remove unused MOD_PRE
test: mock aucodec support all sample formats
audio: check that ptime is within the range of 1-60ms
audio: dont check sample format for packetloss handler
audio: use audio codec srate directly, remove get_srate wrapper
audio: remove get_framesize
audio: handle rtcp sample-rate for asymmetric codecs
audio: remove get_ch()
mpa: return posix error code instead of -1
mk: sort list of files in alphabetical order
menu: sort and align incall commands table
ua: check input argument to ua_print_supported
test: check error from test fixture
ua: use a print handler to print allowed methods
ua: use a single tick instead of backtick for logging
audio: mirror ptime attribute if changed by peer (ref #688) (#700)
audio: receive ptime is always set
plc: count samples from audio input
use sizeof(x) instead of sizeof x
account: fix typo
timestamp: add timestamp_calc_seconds()
call: remove const from menc_event_handler
call: swap order of menc event and error handler
mnat: make struct mnat public
mnat: change to a simpler register api
menc: protocol is always UDP
mnat: change api to always use UDP protocol
stream: add remote RTP/RTCP address to object
menc: add remote RTP/RTCP address to API
dtls_srtp: use remote address from mediaenc API
dtls_srtp: remove dtls_print_sha1_fingerprint
remove audio/video codec cycle
net: add network_if_getname()
sdp: remove sdp_media_format_cycle (unused)
sdp: remove sdp_rattr() -- unused
pcp: updated MNAT api
gzrtp: updated menc api (ref #713)
dtls_srtp: fix warning
zrtp: fix warnings
net: use network_if_getname to get interface name
stream: use enum media_type instead of a string
call: only include aucodec codecs in remote sdp (ref #718)
call: simplify audio encoder/decodet set
stream: add pointer to medianat module
webrtc_aec: add warning
ua: use KEYCODE_REL in dtmf handler (ref #719)
call: add prefix to logline
webrtc_aec: add sample format converter to decoder (ref #712)
webrtc_aec: add sample format converter to encoder (ref #712)
webrtc_aec: add enc/dec to log line
webrtc_aec: fix enum warning
webrtc_aec: echo_cancellation.h is included in aec.h (ref #712)
config: add sip_cafile to template
net: add a function to print IP-addr and interface
net: dont init local address to 127.0.0.1
audio: handle marker bit in stream.c (#724)
avcodec: make sure ffmpeg input buffer has AV_INPUT_BUFFER_PADDING_SIZE space at the end
stream: update doxygen comments
stream: only flush jitter-buffer if it was started
avcodec: fallback define for AV_INPUT_BUFFER_PADDING_SIZE
stream: add pseq_set flag
stream: dont calculate loss if no jitter buffer
webrtc_aec: add support for 32000Hz samplerate
net: multiple nameservers in net_use_nameserver()
webrtc_aec: add reference to webrtc native
dtmfio: use UA_EVENT_CALL_DTMF_START to handle dtmf events
test: use event handler to receive DTMF events
webrtc_aec: remove samplerate check
prepare for 0.6.3 release
gst_video1: cleanup
stream: print mediaenc id
config: add net prefix to prefer_ipv6
codec2: print mode
codec2: modern init of struct aucodec
codec2: add config param codec2_mode
codec2: round up bytes per frame
win32: add httpd module to static.c
codec2: update description
mk: add detection of codec2.so module
video: check if frame pointer is valid
contact: set err properly
audio: no need to clear err, it is not used
config: add opus_samplerate to template
travis: add building of codec2 on OSX (#736)
config: add webrtc_aec to template
Christian Spielberger (2):
call: reset streams on call hold (#707)
Bugfix/flush buffers on call hold (#716)
Dmitry (2):
opus: fixed opus_inbandfec param name in config and examples (#704)
menu: set default values for optional config params (#705)
Juha Heinanen (1):
webrtc_aec: enable delay-agnostic echo cancellation
Nicolas Tizon (1):
audio: increase buffer size for audio device string (#710)
juha-h (5):
- added prefer_ipv6 config variable (#692)
webrtc_aec module: added pthread.h include to .cpp files (#714)
- Updated ilbc module encode/decode/pkloss function arguments (#723)
- Use opus in mono mode (opus/48000/1) if opus_stereo or (#730)
- Added opus_samplerate config parameter. (#733)
premultiply (1):
Interop: Parameters reordering and whitespace removal (#698)
weili-jiang (1):
Count ua references prior to destroy in case UA_EVENT_SHUTDOWN causes references to be removed (#702)
2019-04-19 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.2
Alfred E. Heggestad (124):
daala: remove module
remove USE_VIDEO compile flag (#658)
config: remove sip_trans_bsize option
contact: fix bug in contact prev/next
cmd: remove unused complete flag
log: add command to toggle loglevel ('v')
debug_cmd: fix warning
Remove natbd module (#659)
message: make listen/unlisten more robust (ref #650)
update doxygen comments
srtp: fix warnings
update README
stream: define port 9 as PORT_DISCARD
add offerer flag to video and stream
stream: check for multiplexed RTCP packets on RTP port
stream: change logic for rtcp-mux attribute
omx: update doxygen comment
bv32: add doxygen header
mk: modules in alphabetical order
mk: modules in alphabetical order
mk: modules in alphabetical order
h265: use avcodec API for the encoder
h265: fixes for Debian 8
h265: make it work with old encoder api
h265: init time_base manually
fix av_packet_free
allocate avpacket
h265: cleanup
cleanup
cleanup
h265: add configurable decoder
cleanup
use pkg-config for libs
h265: update documentation
mk: enable h265.so if avcodec installed
h265: include avutil mem.h
travis: use ubuntu 16.04
h265: add wrapper for av_frame_alloc
h265: add wrapper for avcodec_free_context
h265: fix config
fix crash with ffmpeg 2.8
add wrapper for av_packet_free
fix warning
use av_free_packet
cleanup
deprecate v4l.so -- use v4l2.so instead
h265: tested with YUV444P pixel format
h265: check pixel format on changes
Merge remote-tracking branch 'origin/master' into h265_use_avcodec_encoder
h265: fix avcodec_free_context wrapper
H265 use avcodec encoder (#668)
debian: add source format 1.0
aufile: add sample config (ref #663)
v4l: remove module, use v4l2.so instead
video: remote orient parameter
video: remove video_set_orient
aubridge: remove audio resampler
aubridge: fix warning
aubridge: add support for multiple sample formats
aubridge: fix warnings
auloop: remove usage of audio codec
jack: add support for FLOAT sample format
Avcodec remove libx264 (#671)
config: add avcodec.so sample config
stream: set rtcp-mux attribute if enabled
sdl: remove module
stream: send a dummy RTCP packet to open NAT pinhole
avcodec: use pkg-config for linker flags
avformat: use pkg-config for linker flags
config: remove usage of USE_AVCODEC
coreaudio: remove ios specific code
h265: one file per line
avformat: move AVCodec from struct to stack
avformat: remove codec_id check
avformat: minor cleanup
debian: remove usage of shlibs:Depends from dev package
debug_cmd: fix warning
avformat: minor cleanup
avformat: minor cleanup
vidloop: rename intra to keyframe
vidloop: print keyframes only if codec is enabled
avcodec: move destructor to the top of the file
plc: check input arguments
auloop: rename ab to aubuf
Opus multistream (#678)
mqtt: minor updates
mqtt: use re_snprintf
mqtt: update documentation (fixes #669)
h264: fix h264_is_keyframe, IDR_SLICE is keyframe
avcodec: check input arguments
vidloop: show video display pixel-format in summary
avcodec: clean up decoding code
avcodec: add color range MPEG
h265: add color range and GOP size
ffmpeg: check avutil version for color range
avcodec: set slice-max-size in H264 packetization-mode 0
avcodec: add sdp.c
avcodec: move h264_fmtp_cmp to sdp.c
avcodec: add support for H264 packetization mode 1
av1: update comment
avcodec: handle H264 STAP-A packets
test: fix ua register test-cases (ref #680)
ua: add delayed_close flag (ref #680)
menu: call uag_current() directly
video: save pixel format of outgoing stream
aulevel: add support for sample format FLOAT
Vidfilt add param (#682)
test: copy uri_cmp source from libre
sdl2: handle window closed event (SDL_QUIT)
vidloop: stop loop if window was closed
sdl2: add support for quit key
avcodec: fixes for packetization_mode 1
bump version to 0.6.2
vidfilt: update doxygen comments
rtcpsummary: fix warnings about unused variables
mqtt: fix warnings about unused variables
gst_video1: use GST_BUFFER_PTS
mk: add echo module to list of basic modules
core: remove some unused values
menu: call parameter is used
mwi: minor formatting changes
audio: align debug text
Update README.md
ua: add ua_destroy() -- ref #686
Andreas Hansson (1):
Added debug cmd to print UUID (#674)
Juha Heinanen (2):
do not add basic modules if BASIC_MODULES has value 'no'
exclude more modules if BASIC_MODULES=no
Nicolas Tizon (1):
vidloop: update vidsrc (#662)
Roger Sandholm (1):
Readme and config example correction, httpd comments (#666)
Timmo Verlaan (1):
menu: add uadel to delete a uac by aor (#680)
juha-h (1):
srtp: added sending of MENC_EVENT_SECURE event (#660)
premultiply (1):
Add 48 khz sampling rate support (#685)
weili-jiang (1):
Command not found returns error (#664)
2019-02-17 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.1
Aleksei (2):
Update MSVS project (#655)
Fix warnings at windows compilation (#656)
Alfred E. Heggestad (111):
call: prm parameter is mandatory
core: add address-family to stream_param (ref #583)
core: add cname to stream_param (ref #583)
realtime: remove code
contact: add support for current contact (#573)
play: update doxygen comments
mk: add detection of OpenGL framework (ref #575)
stream: make call optional (closes #583)
net: update doxygen comments
contact: update doxygen comments
Module app unload (#589)
mk: disable opengl module in default build
opengles: updated vidisp api
avcapture: fix build for ios (ref #593)
module: update doxygen comments
ua: add missing doxygen comments
ua_set_custom_hdrs: add error checking
ua: add doxygen comments
ua: fix formatting and update doxygen
ua: update doxygen comments
call: update doxygen comments
message: update doxygen comments
custom_hdrs: add doxygen comments
stream: update doxygen comments
video: rename intra to key-frames
sip: add doxygen comments
event: add doxygen comments
mediadev: minor code formatting
mediadev: add doxygen comments
audiounit: rename comp to 'audiounit_comp'
audiounit: check input parameters
coreaudio: remove blocking sleep (#605)
audio: change logline to debug (ref #609)
auloop: show read/write counters in stream duration (seconds)
audiounit: print name of audio component used
audiounit: move aufmt_to_formatflags to audiounit.c
rpm: remove support for in-tree RPM building
account: init mnat, fix warning on mingw32
mk: make sure omx.so is only added once (fixes #612)
silk: remove codec (#611)
aubridge: use sizeof sample format instead of 2
coreaudio: clarify that coreaudio module is for macOS
audio: cleanup comment
pcp: clean up comment
audiounit: check for valid sample size
audiounit: clarify usage of inputBus and outputBus
add accessor to ausrc/auplay base-class
auloop: add summary
vidloop: rename variable
mk: detect CoreAudio framework
webrtc aec (#617)
audiounit: clean up enable/disable
auloop: print sample format
audiounit: clarify usage of inputBus and outputBus
dtls_srtp: remove support for SHA-1 fingerprint
dtls_srtp: remove unused DTLS-SRTP methods
menc: move and document event type
menc: added menc_event_name()
remove obsolete compile flag MODULE_CONF
stream: move handlers to end of struct
test: check magic in audio sample handler
test: added testcase for call with mock medianat
dshow: remove a comment
call: using str_isset() is faster than strlen()
config: remove 'rtcp_enable', always enabled (#623)
stream: remove rtcp flag
update comments
Update README.md
README: added rtcpsummary module
aubridge: add module prefix to global symbols
update copyright year to 2019
audiounit: fix warnings on ios
stream: make stream_sdpmedia public
stream: make stream_update public
audio: make audio_{encoder,decoder}_set public
stream: make struct stream_param public
audio: use samplesize to calculate packet size
audio: use sizeof int16_t instead of 2
stream: update doxygen comments
audio: update doxygen comments
video: update comment
jack: allocate array of ports from channels (ref #625)
jack: update comment
audiounit: check memory allocation
main: fix bug with reading of -u parameter value
audio: split up definition of AUDIO_SAMPSZ
gtk: minor formatting improvements
gtk: check duration_timer_tag (ref #630)
audio: make audio_alloc() public
avformat: print decoder name (ref #639)
menu: change some commands from CMD_IPRM to CMD_PRM
avformat: add pixel format mapping function
avcodec: add pixel format mapping function
avcodec: add pixel format mapping function for encoder
menu: use CMD_PRM for call transfer
contact: add flag to enable presence (fixes #645)
cmd: remove support for progress/interactive commands
menu: fix warning
config: update comment
coreaudio: add support for multiple sample formats, including FLOAT
menu: no need to use command's complete flag
jack: allocate buffer before start (ref #647)
auloop: print newline at the end
Update .travis.yml (#652)
Update .travis.yml (#653)
fix some warnings reported by @Encamy
dshow: fix warning on win32
update doxygen comments
bump version to 0.6.1
config: refresh config template
audiounit: fix warning on ios
José Luis Millán (2):
ctrl_tcp: increase command buffer size (#585)
vumeter: configuration option to disable vumeter output to stderr (#608)
Juha Heinanen (4):
new account functions
fixed typo
added audio_codec api function
added @return description
Nicolas Tizon (4):
sdl2: make window resizable (#587)
coreaudio: mediadev support (#600)
avcodec: force intra (H.263, H.264) frame request if no key frame received (#614)
audiounit: add AUConverter resampler (recorder) (#624)
Olle E. Johansson (1):
Improve mqtt module (#642)
Timmo Verlaan (2):
menu: add uafind to select ua by aor (#626)
menu: create_ua doesn't use dialbuf (#627)
juha-h (5):
Merge pull request #588 from alfredh/new_account_functions
Merge pull request #598 from alfredh/audio_codec
added account_set_answermode api function (#619)
opengles android ndk r18 update (#629)
do not accept incoming calls without srtp if account has mandatory srtp (#651)
2018-12-01 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.6.0
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.6.0
* NOTE: Requires libre v0.6.0 or later
Requires librem v0.6.0 or later
* config:
opus_complexity {0-10} # Encoder's computational complexity
opus_application {audio, voip} # Encoder's intended application
sip_cafile ca.crt # trusted Certificate Authorities
* baresip-core:
- account: added support for per account mwi using
;mwi=on|off addr-param (#530) (thanks Juha Heinanen)
per account support for call transfer (#535)
(thanks Juha Heinanen)
- audio: add audio_start()
add audio_started()
EBU/ACIP invite handler.
flush aubuf when resetting codec
- call: make call_connect() public
make call_notify_sipfrag() public
- contacts: make struct contacts opaque
add contact_uri()
- rtpkeep: code removed
- sdp: remove unused functions
- ua: add sip_cafile config option
(thanks to @wnetbal for the original patch)
add event UA_EVENT_MWI_NOTIFY (thanks Juha Heinanen)
add event UA_EVENT_CALL_TRANSFER
add event UA_EVENT_AUDIO_ERROR
add ua_uri_complete()
- vidfilt: add timestamp parameter to filter API
- vidisp: add timestamp parameter to display API
- video: add RTP timestamp state for receive
* selftest:
- test: add call transfer test (ref #538)
- test: add call via tcp test
* Modules:
* avcapture: fix video resolutions mismatches (#580) (Nicolas Tizon)
* avcodec: remove support for old versions of ffmpeg
detect MPEG4 key-frames
* avformat: remove support for old versions of ffmpeg
* ctrl_tcp: Fix netstring frame handling (#569) (José Luis Millán)
* debug_cmd: /play will always stop current file playing. (#578)
(thanks Ola Palm)
* directfb: updated vidisp api (#568) (thanks andreygursky)
* dshow: mediadev support (#550) (Nicolas Tizon)
* echo: add support for video
* h265: change rate control to ABR (Average Bitrate) (#526)
* mwi: moved printing of mwi info from mwi module to menu module
(thanks Juha Heinanen)
* opengl: properly handle linesize (#566)
* opus: add configuration parameter 'opus_complexity'
add configuration parameter 'opus_application'
(thanks José Luis Millán)
* libsrtp: module removed
* v4l2_codec: use thread instead of fd event (#558)
* winwave: add support for FLOAT sample format (#559)
mediadev support (#556) (Nicolas Tizon)
2018-09-15 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.11
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.11
* NOTE: Requires libre v0.5.7 or later
Requires librem v0.5.3 or later
* build:
* baresip-core:
- account: disable password in the SIP uri
- aucodec: add packet channels (pch) to struct (#489)
- audio: add MAGIC_CHECK to some callback handlers
- audio: fix timestamp for MPA codec
- audio: remove audio_srate and audio_channels (#488)
- audio: remove exception for MPA
- aufilt: add sample format parameter (#492)
- auplay/ausrc: Add dev_list to auplay and ausrc struct (#516)
(thanks Nicolas Tizon)
- call: dump SDP offer, ref #480
- call: Support also SIP URI with missing display name. (#512)
(thanks Christian Spielberger)
- config: update template for coreaudio
- contact: consider empty contacts file as existing
contacts file (#501) (thanks Juha Heinanen)
- main: add -n option for network interface
- main: turn off buffering to standard output (#504)
(thanks Geoff Stewart)
- opus: refresh config template
- stream: add magic number for debugging (ref #514)
- ua: do ua_register explicitly (refs #508) (#509)
- ua: added ua param to message handler (#485) (thanks Juha Heinanen)
- ua: Sending and receiving custom headers (#470)
(thanks Encamy)
- video: added more MAGIC_CHECK checks
- vidsrc: Vidsrc devices list (#491) (thanks Nicolas Tizon)
* selftest:
- test: use event-handler in mediaenc mock
- test: add sip uri with angle brackets (ref #512)
* Modules:
* alsa: remove check for alsa_sample_format config
* alsa: On termination of alsa_play wait until buffer
was processed. (#520) (thanks Christian Spielberger)
* aubridge: check to make sure the device is running before
dereferencing it. (#495) (thanks Geoff Stewart)
* avahi: fix warning when RELEASE=1
* avcapture: use mediadev API to add device names
* coreaudio: Coreaudio select device (#502)
* daala: update vidcodec API
* gtk: update message handler with struct ua pointer
* httpd: add CORS header in http reply (#517) (thanks Nicolas Tizon)
* menu: Send DTMF code command (#496) (thanks Nicolas Tizon)
* rtcpsummary: new RTCP summary module (#505)
(thanks Geoff Stewart)
* speex: remove module (#494)
* srtp: add support for AES-GCM cipher
* vidloop: use portable lock instead of pthread mutex
fix crash when resolution changes
* vidloop: Vidloop fix: video frame rendering is moved to
main thread (#481) (thanks Nicolas Tizon)
* x11: check for shared memory extension
* x11grab: remove old linker path
2018-07-04 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.10
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.10
* NOTE: Requires libre v0.5.7 or later
Requires librem v0.5.3 or later
* build:
- Updated MSVS project (thanks Encamy)
* baresip-core:
- account: add more accessor functions (thanks Juha Heinanen)
- audio: add audio_set_hold
- aufilt: add struct audio parameter
- mediaenc: add menc_event handler (thanks Juha Heinanen)
- net: add support for IP-address in 'net_interface' (thanks @Encamy)
- stream: add stream_call
- stream: check SDP_SENDONLY flag
- stream: correct flags in stream_send (thanks Andreas Hansson)
- stream: set RTP socket buffersize to 65536 (ref #415)
- ua: add events for VU level (thanks Ola Palm)
- ua: add ua_update_account
- ua: don't append domain if uri is IP address (thanks Ali Shirvani)
- ui: add ui_input_long_command
- videnc: add timestamp parameter
- video: add video_calc_rtp_timestamp_fix
- video: lock when setting encoder (ref #418) (#441)
* Modules:
* aufile: add slow cpu detection
* auloop: add samplerate and channels argument to command
* av1: add timestamp parameter to encode function
* avcodec: add timestamp parameter to encode function
set baseline profile on ffmpeg H.264 encoder
remove checks for old versions of libx264
* dshow: fix build for VC and mingw (thanks @Encamy)
add picture vertical flipping (thanks Nicolas Tizon)
* dtls_srtp: add usage of medienc event handler
* gst_video: add timestamp parameter to encode function
* gst_video1: add timestamp parameter to encode function
* h265: add timestamp parameter to encode function
* httpd: no echoing of long commands
* menu: add video switch command /vidsrc (thanks Ali Shirvani)
* opensles: check state before calling Destroy
* sdl2: print renderer info
* vidloop: refactoring of timestamp routines
* vp8: add timestamp parameter to encode function
* vp9: add timestamp parameter to encode function
* vumeter: add periodic events (thanks Ola Palm)
* zrtp: add usage of medienc event handler (thanks Juha Heinanen)
2018-04-21 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.9
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.9
* NOTE: Requires libre v0.5.7 or later
Requires librem v0.5.2 or later
* build:
- Updated MSVS project to VS15 and added several files
to project settings (thanks Encamy)
* config:
video_fps 29.97 # float
* baresip-core:
- conf: add conf_get_float
- timer: add tmr_jiffies_usec
- timestamp: new file for timestamp helpers
- ua: add catchall flag to struct ua
- ua: add ua_set_catchall
- ua: uag_find: return match if catchall flag is set
- vidcodec: change rtp_ts from 32-bit to 64-bit
- videnc: change framerate to double float
- video: change framerate to double float
- vidsrc: add frame timestamp
- vidsrc: change framerate to double float
* selftest:
- mediaenc: add testcase for media encryption
* Modules:
* avcapture: add support for video frame timestamp
* avcodec: fix compiling with old ffmpeg versions
print framerate of decoded bitstream
* avformat: add support for video frame timestamp
* b2bua: add handling of all inbound SIP requests
* cairo: add support for video frame timestamp
* ctrl_tcp: Fix #369. documentation typo (#372)
Fix #370. wrong assignent (#371)
(thanks José Luis Millán)
* dshow: add support for video frame timestamp
* fakevideo: add support for video frame timestamp
add support for timer polling (no pthreads)
* menu: added "statmode_default" config variable (#359)
(thanks Juha Heinanen)
* rst: add support for video frame timestamp
* swscale: add YUV444P pixel format
* v4l: add support for video frame timestamp
* v4l2: add support for video frame timestamp
show actual framerate
* vidbridge: add support for video frame timestamp
* vidloop: add videoloop summary
* x11grab: add support for video frame timestamp
2018-02-11 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.8
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.8
* NOTE: Requires libre v0.5.7 or later
Requires librem v0.5.2 or later
* new commands:
- /aubitrate 64000 -- Set audio bitrate
* new modules:
- ctrl_tcp TCP control interface using JSON payload
(thanks José Luis Millán)
* config:
auenc_format s16 # s16, float, ..
audec_format s16 # s16, float, ..
videnc_format yuv420p # yuv420p, yuv444p, ..
* baresip-core:
- account: password in SIP uri is now deprecated
- aucodec: add encoder/decoder audio sample format (#352)
- aucodec: add bitrate to encoder param
- audio: add function to set encoder bitrate
- audio: sample format for audio encoder/decoder
- call: add call_id accessor
- call: fix memory leak in case sipsess_connect() fails
- config: add configurable video pixel format
- config: set exact installation pathes at build time (#354)
(thanks Guillaume Rousse)
- event: fix memory leak
- event: add call-id to JSON dict
- log: rename log_enable_stderr to log_enable_stdout
- metric: fix calculation of average bitrate
- reg: add display-name to SIP register
- stream: print a message when incoming RTP stream is established
- timer: add tmr_jiffies_usec
- video: save and show pixel format of incoming video
- vidutil: new file for video utility functions
* selftest:
- event: add testcase for events
- sip: make 'struct user' opaque
- ua: update password using ;auth_pass=XXX parameter
* Modules:
* account: update template with auth_pass parameter
* amr: update aucodec API with audio sample format
* avcodec: Return EPROTO when encountering missing fragments in
H264 stream, to trigger intra-frame request (#339)
(thanks Jonathan Sieber)
use AV_INPUT_BUFFER_MIN_SIZE (ref #351)
add support for YUV444P pixel format
* avformat: use av_dump_format()
* bv32: update aucodec API with audio sample format
* codec2: update aucodec API with audio sample format
* ctrl_tcp: new module for TCP control interface using JSON payload
(thanks José Luis Millán)
* g711: update aucodec API with audio sample format
* g722: update aucodec API with audio sample format
* g7221: update aucodec API with audio sample format
* g726: update aucodec API with audio sample format
* gsm: update aucodec API with audio sample format
* gst1: define _POSIX_C_SOURCE to make nanosleep visible
* l16: update aucodec API with audio sample format
* mpa: update aucodec API with audio sample format
* mqtt: update README with correct JSON syntax (ref #356)
* omx: fix compilation for Raspbian
* opus: update aucodec API with audio sample format
add support for FLOAT sample format
* silk: update aucodec API with audio sample format
* speex: deprecate, disable as autodetected module
* speex_aec: always link to libspeexdsp
* speex_pp: always link to libspeexdsp
2017-12-25 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.7
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.7
* NOTE: Requires libre v0.5.5 or later
Requires librem v0.5.0 or later
* Credits: Thanks to Swedish Radio who sponsored many new
features in this release.
* new commands:
- 'conf_reload' -- Reload config file
* new modules:
- gzrtp ZRTP module using GNU ZRTP C++ library
(thanks glenvt18)
- mqtt MQTT (Message Queue Telemetry Transport) module
(sponsored by Swedish Radio)
* config:
- audio_txmode poll|thread Set audio transmit mode
- auplay_format s16|float|s24_3le Set playback sample format
- ausrc_format s16|float|s24_3le Set source sample format
- sdp_ebuacip yes|no Enable EBU-ACIP parameters
- zrtp_hash yes|no Enable/disable ZRTP hash
* baresip-core:
- audio: add sample format conversion
- audio: add sample format for source/playback
- audio: check timestamps on incoming RTP packets
- audio: pace outgoing packets in txmode=thread
- audio: remove txmode with realtime thread
- audio: remove txmode with timer
- audio: set EBUACIP parameters in SDP
- auplay: add sample format to auplay_prm
- auplay: change write handler to any sample format
- ausrc: add sample format to ausrc_prm
- ausrc: change read handler to any sample format
- event.c: new file for generic event handling
- event: add event_encode_dict to encode event to a dictionary
- event: added UA_EVENT_CALL_RTCP for received RTCP
- log: print to stdout (ref #320)
* selftest:
- add test for different audio tx-modes
- add test for float audio sample format
* Modules:
* alsa: add support for multiple sample formats
* audiounit: add support for FLOAT sample format
* auloop: add support for multiple sample formats
* avahi: Bugfix: Destroy resolver after callback (#318)
(thanks Jonathan Sieber)
* avcodec: change x264 rate control mode to ABR (#334)
(thanks Jonathan Sieber)
* debug_cmd: add command 'conf_reload' to reload config file
* gzrtp: ZRTP module using GNU ZRTP C++ library
(thanks glenvt18)
* menu: add config 'ringback_disabled' to disable playing
of ringback tone.
* mqtt: MQTT (Message Queue Telemetry Transport) module
new module using libmosquitto as the backend.
* opus: fix encoder bitrate, ref #305
add opus_stereo config parameter (thanks Ola Palm)
add config param opus_sprop_stereo (thanks Ola Palm)
* portaudio: add support for FLOAT sample format
* pulse: add support for FLOAT sample format
remove garbage at the beginning of a recording (#323)
* quicktime: module was removed
* rst: add support for multiple sample formats
* zrtp: add signaling hash support (#311)
2017-10-14 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.6
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.6
* NOTE: Requires libre v0.5.5 or later
Requires librem v0.5.0 or later
* New Baresip logo (thanks Ernst and community)
* baresip-core:
- log: rename error to error_msg due to GNU extension clash
- ua: remove ua_sipfd()
* Modules:
* avahi: Avahi Zeroconf Module (thanks Jonathan Sieber)
* avcodec: handle fragment packet loss
* cairo: draw a dancing logo
* ice: set ICE role correctly
set retransmit count (RC) to 4
* opensles: fix recorder speaker setup (thanks Juha Heinanen)
* opus: fix encoder bitrate, ref #305
* zrtp: encrypt/decrypt RTCP packets (thanks @glenvt18)
2017-09-07 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.5
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.5
* NOTE: Requires libre v0.5.5 or later
Requires librem v0.5.0 or later
* new commands:
- insmod module.so -- Load a module
- rmmod module.so -- Unload a module
* config:
- fullscreen yes|no Enable fullscreen display
* baresip-core:
- account: optional param 'auth_pass' for password
add account_set_auth_pass()
add account_aor()
add account_auth_pass()
- contact: add update handler (thanks Jonathan Sieber)
- h264: add rtp_ts RTP Timestamp
- module: add module_load/unload
remove list of application modules
- stream: reset timer on incoming RTCP packets (fixes #271)
- ui: make the API re-entrant
- video: add RTP timestamp to videnc packet handler
add video_calc_rtp_timestamp()
add video_calc_seconds()
- video: use RTP timestamp from video encoder
* selftest:
- add test for video timestamps
* Modules:
* account: move password prompt here
* av1: use encoder PTS to calculate RTP timestamp
* avcodec: use encoder PTS to calculate RTP timestamp
use level_idc=0x1f for x264
* cons: updated UI api
* evdev: updated UI api
* gst_video: use encoder PTS to calculate RTP timestamp
* gst_video1: use encoder PTS to calculate RTP timestamp
* h265: use encoder PTS to calculate RTP timestamp
fix FU decoder bug
* httpd: updated UI api
* ice: move gathering from lib to app
(requires libre v0.5.5 or later)
* menu: updated UI api
* mwi: updated UI api
* presence: Handle contacts added at run-time
(thanks Jonathan Sieber)
* sdl: updated UI api
* sdl2: add support for fullscreen video
* stdio: updated UI api
* v4l: add support for more pixel-formats
* v4l2_codec: use encoder PTS to calculate RTP timestamp
* vp8: use encoder PTS to calculate RTP timestamp
* vp9: use encoder PTS to calculate RTP timestamp
* wincons: updated UI api
2017-06-24 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.4
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.4
* NOTE: Requires libre v0.5.4 or later
Requires librem v0.5.0 or later
* config:
- audio_level yes|no Enable audio level RTP extension
* baresip-core:
- add support for Client-to-Mixer Audio Level Indication (RFC 6464)
- add support for RTP Header Extensions (RFC 5285)
- module: dont load same static module twice
- ua: add ua_progress()
- ua: check for Accept header in incoming OPTIONS request
- use a dummy RTP port for incoming OPTIONS (ref #265)
- vidcodec: make the API re-entrant
- vidfilt: make the API re-entrant
- vidisp: make the API re-entrant
- vidsrc: make the API re-entrant
* selftest:
- add test for audio level indication in call
- add test for call progress
* Modules:
* (all video modules updated with API-changes)
* zrtp: check for RTP packet in send handler (ref #262)
(thanks to MobiSciLab for reporting the bug)
- registered zrtp_log function with zrtp engine
- improved info message on how to verify remote peer
- improved setting and printing of zrtp cache file
(thanks Juha Heinanen)
2017-05-14 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.3
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.3
* NOTE: Requires libre v0.5.3 or later
Requires librem v0.5.0 or later
* config:
- (no changes)
* build:
- detect jack module (thanks Tony Langley)
- Updated MSVS projects to vs2015 (thanks Mikhail Barg)
* baresip-core:
- aulevel: add aulevel_calc_dbov()
- audio: Set correct clock rate for telephone events
(thanks Jan Hoffmann)
- play: Add gapless repeat for tone playback (thanks Jan Hoffmann)
* selftest:
- add tests for aulevel
- add tests for audio player
- add mock aucodec/auplay
* Modules:
* gst_video1: Tune x264enc for low latency (thanks Jonathan Sieber)
* httpd: fix a crash
* ice: update to latest libre ICE-api
* omx: Fixed some problems on OMX/RaspberryPi (thanks Jonathan Sieber)
* srtp: fix SRTP for early-media (thanks Jan Hoffmann)
* vumeter: use aulevel_calc_dbov to calculate signal energy
* zrtp: update to latest libzrtp from freeswitch (thanks Juha Heinanen)
2017-04-07 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.2
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.2
* NOTE: Requires libre v0.5.0 or later
Requires librem v0.5.0 or later
* new modules:
- omx OpenMAX IL video display module (thanks Jonathan Sieber)
* config:
- (no changes)
* baresip-core:
- aucodec: make the API re-entrant
- aufilt: make the API re-entrant
- auplay: make the API re-entrant
- ausrc: make the API re-entrant
- video: using a video-source is now optional
* Modules:
* avformat: add pixelformat AV_PIX_FMT_YUVJ420P (Thanks Gary Metalle)
* cairo: print picture info, use grey background
* dtmfio: check fd before calling fclose (thanks Richard Perez)
* h265: enable YUV444P pixelformat
* oss: fix build for Solaris 11
* speex: mark the module as deprecated, see speex.org
2017-03-04 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.1
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.1
* NOTE: Requires libre v0.5.0 or later
Requires librem v0.5.0 or later
* new modules:
* config:
- stunuser STUN username for STUN/TURN/ICE
- stunpass STUN password for STUN/TURN/ICE
- snd_path Path to sndfile audio dump files
* baresip-core:
- account: add more accessor functions
- account: add 'stunuser' and 'stunpass'
- commands: make the struct commands opaque
- message: make the API re-entrant, multiple listeners
- menc: make the API re-entrant
- mnat: make the API re-entrant
* selftest:
- add tests for account
- add tests for message
* Modules:
* amr: use MOD-CFLAGS instead of global CFLAGS
* avcodec: added optional config 'avcodec_h264dec' to specify hardware
accellerated FFmpeg decoder (thanks Harald Gutmann)
* avformat: remove blocking sleep, use packet timestamp to
pace video stream (thanks Harald Gutmann)
* debug_cmd: add OpenSSL version to systems info
* gtk: fix build where USE_NOTIFICATIONS is not defined
get rid of system header warnings by using -isystem
* httpd: add support for un-escaping of URL parameters
(thanks to elektm93)
* menu: add new command 'ausrc' to switch audio source
add new command 'auplay' to switch audio player
* sdl2: add more pixelformats (ref #202)
(thanks Harald Gutmann)
* sndfile: add config to specify path for dump files (thanks Elektm93)
add test for sndfile on *BSD. (#194) (thanks jungle-boogie)
* swscale: get dst-size from config (ref #203)
* v4l2_codec: Video device selection bug (#218)
(thanks Richard Perez)
2016-12-23 Alfred E. Heggestad <alfred.heggestad@gmail.com>
* Version 0.5.0
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.5.0
* NOTE: Requires libre v0.5.0 or later
Requires librem v0.5.0 or later
* new modules:
- av1 Experimental AV1 video codec
- debug_cmd Debug commands for advanced users
- pcp Port Control Protocol (PCP) for NAT traversal
- swscale Video scaling using FFmpeg's libswscale
* config:
- call_max_calls Maximum number of calls per account
* baresip-core:
- call: add multiple lines
- call: start video on reinvite (thanks Gary Metalle)
- cmd: add support for long commands
- cmd: make it re-entrant
- config: add some modules to template (thanks Dmitrij D. Czarkoff)
- contact: make it re-entrant
- play: make it re-entrant
- vidcodec: add a intraframe-flag to api
- video: resend FIR until Intra frame received
* selftest:
- add test for DTMF in call
- add test for contacts
- add test for long commands
- add test for maximum calls
- add test for multiple calls
- add test for video call
- add audio-source mock
- add video-codec mock
- add video-display mock
- add video-source mock
* Modules:
* aufile: convert samples from little-endian to host-endian
* auloop: use long commands /auloop and /auloop_stop
* av1: new module for Experimental AV1 video codec
* avcodec: add config option 'avcodec_h264enc' to set encoder name
(thanks to @hargut)
* avformat: fix init and warnings (thanks Maciej Koman)
* b2bua: use long command /b2bua
* contact: use long commands
* debug_cmd: new module for advanced debug commands
* g7221: expose spandsp api (thanks to Steve Underwood)
* gtk: use long command /gtk
* h265: add 'profile-id=1' to SDP
* menu: add long commands
add command 'line' or '@' to set current call
* opengl: fix deprecated warnings on OSX 10.12
* opensles: add support for stereo
(thanks to Juha Heinanen and Vijay Pratap Singh)
* opus: add support for SDP parameter mirroring
(thanks to Sveriges Radio)
* pcp: new module for Port Control Protocol (PCP) NAT traversal
requires librew (https://github.com/alfredh/rew)
* plc: expose spandsp api (thanks to Steve Underwood)
* presence: add long commands /presence_{on,off}line
* snapshot: use long commands (thanks Dmitrij D. Czarkoff)
* sndio: use driver-suggested buffer size (thanks Dmitrij D. Czarkoff)
* swscale: new module for video filter using libswscale
* v4l2: pick up VID_FMT_NV12 and VID_FMT_NV21 formats as well (#176)
don't check for native/emulated format (#179)
(thanks Dmitrij D. Czarkoff)
* vidloop: use long commands
* vp8: add 'intra' parameter to decoder api
fix building with old versions of libvpx
* wincons: graceful closing of thread (fixes #151)
(thanks to @GGGO)
* zrtp: use long command
2016-07-22 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.20
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.4.20
* NOTE: Requires libre v0.4.17 or later
Requires librem v0.4.7 or later
* new modules:
- pulse Pulseaudio driver
- vp9 VP9 video codec
* config:
- audio_path Path to audio files
- call_local_timeout Timeout for incoming calls
- redial_attempts Number of redial attempts
- redial_delay Redial delay in seconds
* baresip-core:
- baresip: added a global baresip instance (WIP)
- call: add RTP timeout (thanks to Sveriges Radio)
- config: added call_local_timeout for incoming call timeout
- config: added compile-time configureable CONFIG_PATH
- config: added 'audio_path' config variable (thanks Juha Heinanen)
- net: made it re-entrant with struct network
- ua: added uag_set_exit_handler
- ua: fix bug with reg_uri limited to 64-chars
- video: vidfilters should not modify decoded image
* selftest:
- add test for network
- add test for sending SIP OPTIONS
- add test for RTP timeout
* Modules:
* avcodec: fix usage of deprecated API
* avformat: remove support for scaling
fix usage of deprecated API
* cons: relay log-messages to active UDP/TCP connections
https://github.com/alfredh/baresip/issues/144
* h265: fix usage of deprecated API
* menu: added support for re-dial on failure
(thanks to Sveriges Radio)
* mpa: Bug with reinit of codec structs (thanks Christian Hoene)
* natpmp: added support for RTCP
* presence: use correct struct in deref handler
* pulse: new module for Pulseaudio driver
(thanks to Matthias Apitz for testing)
* vidloop: vidfilters should not modify decoded image
* vp8: module renamed from vpx.so to vp8.so
* vp9: new module implementing VP9 video codec
* wincons: use ReadConsoleInput, thanks to GGGO (fixes #139)
https://github.com/alfredh/baresip/issues/139
2016-05-20 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.19
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.4.19
* NOTE: Requires libre v0.4.14 or later
Requires librem v0.4.7 or later
* new modules:
- mpa MPA Speech and Audio Codec (thanks Christian Hoene)
* baresip-core:
- audio: remove is_g722 exception
use aucodec's rtp clockrate for calculating RTP timestamp
plc: make sure sampc is exactly one ptime frame
- aucodec: split srate into DSP srate and RTP clockrate
(these are different for e.g. G.722 and MDA)
- mos: add mos_calculate() (thanks Lorenzo Mangani)
- net: use configured dns servers only, if specified
- ua: fix potential NULL-pointer crash for uag.cfg
* selftest:
- add test for SIP registration with DNS
- add test for SIP registration with authentication
- add test for MOS calculations
- added a mock DNS Server
- added a mock SIP Server
* Modules:
* aucodec: add support for NV12 and YUVJ420P pixel formats
* daala: update to libdaala version 0.0-1564-g79787c7
* gtk: fix autodetection of libgtk+ 2.0 (thanks Charles Lehner)
* h265: remove call to x265_cleanup, caused crash on OpenBSD
* mpa: new module that implements MPA Speech and Audio Codec
(this module was contributed by Christian Hoene)
* opus: added new configuration parameters:
opus_cbr {yes,no} # Constant Bitrate (inverse of VBR)
opus_inbandfec {yes,no} # Enable inband FEC
opus_dtx {yes,no} # Enable DTX
* presence: improved interoperability, allow white space before
xml element closing tags (thanks Juha Heinanen)
* x11: added borderless window (thanks Doug Blewett)
2016-03-12 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.18
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.4.18
* NOTE: Requires libre v0.4.14 or later
Requires librem v0.4.7 or later
* baresip-core:
- call: fix SIP INFO with dtmf-relay (thanks Gary Metalle)
- ua: add event UA_EVENT_CALL_CLOSED for ua_hangup()
* selftest:
- add tests for answer a call and hangup
* Modules:
* alsa: fix potential crash (thanks Gary Metalle)
* audiounit: fix compilation for iOS (issue #91)
* avcodec: fix compilation for FFmpeg 3.0
* avformat: fix compilation for FFmpeg 3.0
* gtk: always handle incoming calls (thanks Charles Lehner)
* h265: fix compilation for FFmpeg 3.0
* menu: add config 'menu_bell off/on' to enable Bell alert
add command 'A' for switch audio device (thanks AlexMarlo)
* v4l2_codec: add list of encoders (fixes #99)
2016-01-17 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.17
* GIT URL: https://github.com/alfredh/baresip.git
* GIT tag: v0.4.17
* NOTE: Requires libre v0.4.14 or later
Requires librem v0.4.7 or later
* new modules:
- echo Echo server module
- jack JACK Audio Connection Kit audio-driver
* baresip-core:
- config: keep config object in memory
- ua: moved playing of ringtones out of core, to "menu" module
(let's keep the core nice and slim..)
- ui: added ui_password_prompt()
* selftest:
- silence debug/info log by default, only print warnings
(use -v to see verbose logging)
* Modules:
* alsa: added config option to specify the sample format
"alsa_sample_format {s16,float,s24_3le}"
thanks to Ola Palm for valuable feedback
* audiounit: fix recording on OSX (thanks Sebastian Reimers)
print hardware samplerate in debug mode
* auloop: add support for 44100 Hz samplerate
* daala: update to latest libdaala API (thanks Dmitrij D. Czarkoff)
* echo: new module which implements a simple Echo-server, to
be used in combination with the aubridge.so module.
contributed by Sebastian Reimers
* gtk: fixes to support C89 compiler (thanks Dmitrij D. Czarkoff)
* jack: new module which implements audio-driver for JACK
* menu: playing of ringtones moved here, from ua.c
* sndio: fix crash when device open fails (thanks Dmitrij D. Czarkoff)
2015-12-01 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.16
* GIT URL: https://github.com/alfredh/baresip.git
* GIT commit bed2241da3261e472f09b21958f0cc1324a94f27
* GIT tag: v0.4.16
* NOTE: Requires libre v0.4.14 or later
* new modules:
- v4l2_codec Video4Linux2 video codec (H264 hardware encoding)
- vidinfo Video info overlay module
* baresip-core:
- audio: add audio_set_source() and audio_set_player()
- audio: flush tx-buffer for all modes (thanks Thibault Gueslin)
- call: add call_is_outgoing()
- call: check address-family of incoming SDP offer (thanks Olle)
- h264: move H.264 packetization code to core
- main: add -u option to append extra global UA parameters
- main: pre-load modules after all arguments are parsed
- ua: add events UA_EVENT_SHUTDOWN,UA_EXIT
- ua: add ua_hold_answer()
- ua: add ua_set_media_af()
- ua: delay mod-unloading if mods has a ref to struct ua
* build:
- add verbose build with V=1 (thanks Dmitrij D. Czarkoff)
- add pkg-config file (thanks William King)
- add travis.yml file for Github build-system
* Modules:
* alsa: fix memory leaks
* avcodec: move common H.264 packetization code to core
* cairo: use pkg-config in makefile
* daala: update to latest libdaala (thanks Dmitrij D. Czarkoff)
* gst_video: use H.264 packetization API from core
* gst_video1: use H.264 packetization API from core
* gtk: fix segmentation fault on window close
* mwi: add 500ms delay after closing subscription
* oss: use pthread for ausrc instead of fd_listen (fixes FreeBSD)
* presence: use sipevent_sock instance from UA core
add 500ms delay after closing subscription
* v4l2_codec: new module
* vidinfo: new module
* zrtp: fix ZRTP over TURN by moving helper to layer 10
fix ZID verification (thanks Ingo Feinerer)
2015-09-26 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.15
* GIT URL: https://github.com/alfredh/baresip.git
* GIT commit 86262a6fc17e19e2be82eb8a2a05ec0f884d3d38
* GIT tag: v0.4.15
* NOTE: Requires libre v0.4.13 or later
* added selftest binary
* baresip-core:
- audio: fix televent when pt != 101 (reported by AndyJRobinson)
- magic: use __func__ for C99 or later
- sip: make sip_req_send() public
- ua: add UA_EVENT_CALL_DTMF_START/END, thanks Gary Metalle
* Modules:
* alsa: added extra logging
* gtk: add support for libnotify (thanks Charles Lehner)
* video: fix potential null deref (thanks Tomasz Ostrowski)
* zrtp: added 36-bytes preamble for TURN-header
2015-08-08 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.14
* GIT URL: https://github.com/alfredh/baresip.git
* GIT commit ebac23b0692de71ee4c3a436f0372013150c937f
* GIT tag: v0.4.14
* NOTE: Requires libre v0.4.13 or later
* new modules:
- gtk GTK+ 2.0 UI (thanks Charles E. Lehner)
- gst1 Gstreamer 1.0 audio module
- gst_video1 Gstreamer 1.0 video module (thanks Thomas Strobel)
- daala Experimental video-codec using Daala
* baresip-core:
- baresip: added -m argument to pre-load modules
- config: add kqueue to sample config (thanks Dmitrij D. Czarkoff)
- log: make code C89 compliant (thanks Victor Sergienko)
- module: added module_preload()
- ua: add CALL_EVENT_TRANSFER_FAILED
- ua: skip initial white space from uri (thanks Juha Heinanen)
- ua: ua_prev_call()
- videnc: move videnc_packet_h to update-handler
* build:
- added optional $(MOD)_CFLAGS for local module CFLAGS
- added project file for Visual C++ Express 2010
- freebsd: add include path to $(SYSROOT)/local/include
(thanks Hellmuth Michaelis)
* Modules:
* avcodec: make code C89 compliant (thanks Victor Sergienko)
* cons: make code C89 compliant (thanks Victor Sergienko)
* daala: new module
* dshow: updates for VC2010 (thanks Victor Sergienko)
* gst1: new module
* gst_video1: new module
* gtk: new module
* menu: fix crash when 0 UAs (thanks Hans Petter Selasky)
added command 'H' to hold previous call (thanks xanm)
* wincons: make code C89 compliant (thanks ggcoding)
2015-06-20 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.13
* GIT commit 2e3e825ef5532dfde5a8b52de9ebaac51aa20a9c
* NOTE: Requires libre v0.4.12 or later
* new modules:
- aufile Audio module for using a WAV-file as audio input
- b2bua Back-to-Back User-Agent (B2BUA) module
- codec2 CODEC2 audio codec
- gst_video Gstreamer video codec
- h265 H.265 (HEVC) video codec
* baresip-core:
- contact: add support for access-control (thanks Doug Blewett)
- ausrc: change base-class to a const pointer
- auplay: change base-class to a const pointer
- vidsrc: change base-class to a const pointer
- vidisp: change base-class to a const pointer
- video: smooth sending of video packets
* Modules:
* amr: added support for octet-align mode (thanks to Stefan Sayer)
* aubridge: copy audio-samples if resampler not needed
* aufile: new module for using a WAV-file as audio source
* avcapture: only register 1 video source
* avformat: fix segfault on recent versions of libav
* b2bua: new experimental module
* codec2: new module for CODEC2 audio codec
* dtls_srtp: uppercase fingerprint, interop (thanks Juha Heinanen)
alternative SDP protocols for interop
* dtmfio: unregister event handler on close (thanks Hellmuth Michaelis)
* gst_video: new module using Gstreamer as a video codec
(Thanks to Victor Sergienko and Fadeev Alexander)
* h265: new module for H.265 video codec
* httpd: added raw mode (thanks Lorenzo Mangani)
* menu: create user-agent with a command 'R' (thanks Lorenzo Mangani)
* opus: add configuration of "opus_bitrate"
(thanks to Juha Heinanen)
* speex: add configuration of "speex_mode_nb" and "speex_mode_wb"
(thanks to Dmitrij D. Czarkoff and Juha Heinanen)
* vidloop: add VIDLOOP_INTERNAL_FMT and split encoder/decoder
* x11: catch Window delete (thanks to Doug Blewett)
* zrtp: initialize remote_zid (thanks to Ingo Feinerer)
2014-12-24 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.12
* GIT commit 67993e35d980375458348b264c4a35a944bb5180
* NOTE: Requires libre v0.4.11 or later
* baresip:
- account: add regint and pubint
- audio: fix checking of sample-rate range
- config: remove the "input" block
- config: added support for quoted device parameters
- config: fix conversion of bandwidth to kbit/s
- config: generate more relevant config for FreeBSD and OpenBSD
(thanks Dmitrij D. Czarkoff)
- reg: add support for extracting GRUU parameter
- main: add -p option to set path to audio files
- sipreq: make response-handler optional
- ua: add support for GRUU (RFC 5627)
(many thanks to Juha Heinanen for starting this work and
helping out with the testing)
- ua: moved presence-status to each struct ua instance
- ua: add presence status to each User-Agent instance
- ua: use public-GRUU if set, otherwise local cuser
- ui: make UI single instance
- video: add VIDENC_INTERNAL_FMT (suggested by Victor Sergienko)
* docs: added sample configuration files
* account: added pubint for Publishing Interval
* avcodec: upgrade to recent ffmpeg/libav APIs
either FFmpeg or libav can be used
* celt: deleted module (replaced by opus)
* cons: update usage of struct ui, added output handler
added config: cons_listen 0.0.0.0:5555
* evdev: update usage of struct ui, added output handler
added config: evdev_device /dev/input/event0
* httpd: added ui output handler
* menu: added command 'o' for sending OPTION request
(thanks to Juha Heinanen)
added command 'D' for accepting incoming calls
* mwi: subscribe to MWI after Registration succeeded
(thanks to Juha Heinanen)
* opensles: add double-buffering and some tuning
(thanks to Francesco Bradascio)
* opus: added config "opus_bitrate" (thanks to Sebastian Reimers)
* presence: added support for PUBLISH (thanks to Juha Heinanen)
interop fixes and tuning
* stdio: update usage of struct ui, added output handler
* uuid: use internal version of generating UUID
* v4l2: use memory mapped mode only
* vumeter: dont call tmr_start from non-RE thread
* wincons: update usage of struct ui, added output handler
* winwave: fix bug when closing player device
(thanks to Tomasz Ostrowski)
add support for mapping device name to index
* zrtp: add support for verify SAS (thanks to Ingo Feinerer)
2014-06-21 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.11
* GIT commit 7a465f2eb92f4e32740093e5ad4970d528908c51
* baresip:
- audio: added audio_ismuted() to get audio mute status
- audio: fix timestamp generation for stereo-streams
- audio: send outgoing audio-packets as soon as possible
- audio: upgrade to sample-based ausrc/auplay API
- auplay: change API to use samples instead of 8-bit buffer
- auplay: remove option to specify sample format (always S16LE)
- ausrc: change API to use samples instead of 8-bit buffer
- ausrc: remove option to specify sample format (always S16LE)
- call: added support for X-RTP-Stat header (thanks Lorenzo Mangani)
- call: check for common audio-codecs (thanks Juha Heinanen)
- logging: use info() instead of DEBUG_INFO();
- logging: use warning() instead of DEBUG_WARNING()
- play: convert WAV-file from little-endian to native-endian
- removed support for Symbian OS
* debian: upgrade debian files
* avcapture: also build for MacOSX
* alsa: fix sample-endianess with SND_PCM_FORMAT_S16
upgrade to sample-based ausrc/auplay API
* audiounit: upgrade to sample-based ausrc/auplay API
* auloop: upgrade to sample-based ausrc/auplay API
* coreaudio: upgrade to sample-based ausrc/auplay API
* dtls_srtp: use DTLS code from libre (needs libre v0.4.9 or later)
use SRTP code from libre (needs libre v0.4.9 or later)
* dtmfio: new module to send DTMF-events via FIFO file
(contributed by Aaron Herting)
* fakevideo: new module for fake video input/output driver
* gst: upgrade to sample-based ausrc/auplay API
* ice: set default candidates for ICE-lite
* libsrtp: module 'srtp.so' renamed to 'libsrtp.so'
* mda: Symbian MDA audio driver was deleted
* menu: fix issue with audio-mute on multiple calls
* opensles: upgrade to sample-based ausrc/auplay API
* oss: upgrade to sample-based ausrc/auplay API
* portaudio: upgrade to sample-based ausrc/auplay API
* rst: upgrade to sample-based ausrc/auplay API
* selftest: new module for testing the baresip core api
* sndio: new module for OpenBSD audio driver
(It was contributed by Dmitrij D. Czarkoff, thank you!)
* srtp: module is now using SRTP-stack from libre (v0.4.9 or later)
* syslog: use logging framework to get messages
* v4l2: add format negotiation and OpenBSD support
(contributed by Dmitrij D. Czarkoff)
* winwave: upgrade to sample-based ausrc/auplay API
2014-01-23 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.10
* baresip:
- account: add account_set_display_name() -- thanks Dimitris
- audio: use both srate/channels to check if resampler is needed
- aufilt: change from frame_size to ptime
- auplay: change from frame_size to ptime
- ausrc: change from frame_size to ptime
- config: add optional ausrc_channels and auplay_channels
- config: create config dir with mode 0700 (suggested by Jann Horn)
- play: update auplay usage with ptime
* alsa: update to new ausrc/auplay API with ptime
fix bug when snd_pcm_readi() returns -EPIPE (thanks Remik)
open device from main thread instead of alsa-thread (thanks EL)
(caused problems with Sennheiser Century SC 660 + USB adapter)
* auloop: minor cleanups and improvements
* coreaudio: update to new ausrc/auplay API with ptime
* gst: update to new ausrc/auplay API with ptime
* l16: fix a bug with sample count
* opus: fix a memory corruption error in opus_decode_pkloss()
* oss: update to new ausrc/auplay API with ptime
* plc: update to new aufilt API with ptime
* portaudio: update to new ausrc/auplay API with ptime
fix bugs when using channels=2 (stereo)
configure device index using "device" parameter
* rst: update to new ausrc/auplay API with ptime
* speex_aec: update to new aufilt API with ptime
* speex_pp: update to new aufilt API with ptime
* winwave: update to new ausrc/auplay API with ptime
* zrtp: update to use libzrtp from Travis Cross' github
use config dir to store ZRTP cache-file (thanks Juha Heinanen)
2014-01-06 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.9
* new modules:
- zrtp Media Path Key Agreement for Unicast Secure RTP
* build:
- added support for LLVM clang compiler
* baresip:
- account: add account_laddr()
- audio: upgrade to new librem auresamp API
- config: use oss,/dev/dsp as default device for FreeBSD
- log: added new logging framework
- main: added new verbose debug argument (-v)
- net: added sanity check for HAVE_INET6 build flag
- play: added play_set_path() -- thanks to Dimitris P.
- ua: added uag_find_param()
- ua: fix param-bug in ua_connect() -- thanks to Juha Heinanen
* aubridge: upgrade to new librem auresamp API
* avcodec: use new av_frame_alloc() api
* celt: deprecate CELT-module, use OPUS instead
* opengles: fix warnings (thanks to Dimitris P.)
* opensles: fix bugs in player and recorder
* opus: encode/decode sdp parameters as of I-D
* speex_resamp: module removed, replaced by librem's resampler
* zrtp: new module for ZRTP media encryption (use ;mediaenc=zrtp)
2013-12-06 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.8
* new modules:
- dtls_srtp DTLS-SRTP media encryption module (RFC 5763,5764)
- aubridge Audio Bridge to connect auplay->ausrc
- vidbridge Video Bridge module to connect vidisp->vidsrc
* baresip:
- added RFC 5576 Source-Specific Media Attributes in SDP
- audio: set SDP bandwidth only if "rtp_bandwidth" config set
- play: do not store a copy of global config
- stream: save RTCP statistics from Sender-reports
- stream: add SDP ssrc attribute
- stream: added metrics for packets/bytes transmit/receive
- ua: added uag_current()/_set() to get/set current User-Agent
- video: set maximum RTP packet-size to 1024 bytes
* config:
- added "video_display module,device" for Video Display
- added "rtp_stats {off,on}" for RTP Statistics after Call
- default RTP bandwidth is now 0-0
* contact: dynamic command description for "Message" handling
dial from current UA (thanks to Simon Liebold)
* isac: upgrade to draft-ietf-avt-rtp-isac-04
* srtp: added auto-negotiation of RTP-profile for incoming calls
(RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)
* vidloop: fix memory leak
2013-11-12 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.7
* new modules:
- httpd HTTP webserver UI module
* baresip:
- added RFC 5506 Support for Reduced-Size RTCP
- audio: minor cleanups
- cmd: ignore RELEASE key in editor mode
- conf: add conf_get_sa()
- mnat: add address family (af) to session handler
- realtime: fixes for iOS (thanks Dimitris)
- ua: make ua_register() public
- ua: add ua_calls() to get list of calls
- ua: only create register client if regint > 0
* debian: update dependencies (thanks Juha Heinanen)
* rpm: added RPM package spec file
* alsa: open device from thread to avoid blocking re-main loop
* avcodec: build fixes for Debian Testing
* avformat: use sys_msleep()
* contact: improve matching logic (thanks EJC Lindner)
* dshow: initialize variables (found with cppcheck)
* evdev: fix formatted printing (found with cppcheck)
* ice: use address family (AF) from call
* ilbc: update to separate encoder/decoder states (thanks Dimitris)
* snapshot: initialize variables (found with cppcheck)
* stun: use address family (AF) from call
* turn: use address family (AF) from call
* uuid: fix usage of strncat()
2013-10-11 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.6
* new modules:
- directfb DirectFB video display module (thanks Andreas Shimokawa)
- dshow Windows DirectShow vidsrc (thanks Dusan Stevanovic)
- wincons Console input driver for Windows
* baresip:
- audio: print audio-pipelines in console/debug
- aufilt: split into separate encoder+decoder states
- call: add local uri/name, dtmf-handler
- call: fix decoding of DTMF/SIP-INFO for '*' and '#'
- export CALL_EVENT_* in public API
- fix various clang warnings
- sipreq: use outbound proxy if specified (thanks EJC Lindner)
- ua: add possibility to specify 'struct call' for hangup/answer
- ua: move SIP extensions into a dynamic vector container
- ua: move playing of tones from call.c to ua.c
- vidfilt: split into separate encoder+decoder states
- vidisp: remove input handler
* menu: improve call-transfer handling
* plc: update to separate encoder/decoder states
* selfview: update to separate encoder/decoder states
* snapshot: remove state which was not needed
* sndfile: update to separate encoder/decoder states
print unique timestamp to saved files
* speex_aec: update to separate encoder/decoder states
* speex_pp: update to separate encoder/decoder states
* vidloop: update to separate encoder/decoder vidfilt states
* vumeter: update to separate encoder/decoder states
* wincons: new module for Console input on Win32
2013-08-31 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.5
* new modules:
- account Account loader module
- natpmp NAT-PMP client (RFC 6886)
- sdl2 Video display using libSDL2
* baresip:
- account: added SIP account parser and container
- config: split conf.c into conf.c and config.c
- config: move enum audio_mode to struct config
- config: move uuid to struct config
- more usage of the #ifdef USE_VIDEO macro
- message: add handling of SIP MESSAGE send/recv
- mediaenc: added rtp_sock parameter to media-handler
- ua: cleanup public struct ua API
- vidisp api: remove unused 'parent' parameter
- call: handle incoming DTMF in SIP INFO (application/dtmf-relay)
- sdp: added sdp_decode_multipart()
- net: fix bug on IP-refresh when 'net_interface' is used
- video: minor cleanups
handle incoming RTCP_RTPFB_GNACK
* isac: fix encode_update() signature
* menu: move dialbuffer here from ua.c
added command 'g' to print current config
* mwi: multiple MWIs for multiple UAs
* presence: include supported methods in SIP messages
* srtp: improved interop and debugging
handle incoming RTP/RTCP-demultiplexing
* uuid: write loaded UUID directly to struct config
* vidloop: added video-filters
2013-05-18 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.4
* new modules:
- g726 G.726 audio codec
- mwi Message Waiting Indication
- snapshot Save video-stream as PNG images
* config:
- added 'sip_certificate' to use a Certificate for SIP/TLS
- added 'ausrc_srate' and 'auplay_srate' to force DSP samplerate
* baresip:
- added a simple BFCP client
- aufilt: improved API
- mediaenc: improved API with session state
- ua: added event handler framework
- aucodec: improved API with separate encode/decode state
- vidcodec: improved API with separate encode/decode state
- sdp.c: added SDP helper functions
- ua: move registration client to reg.c
- audio: added internal resampler
* auloop: added config option 'auloop_codec' for setting codec
* ice: remove old 'ice_interface' config option
* menu: move handling of status-mode here
* selfview: added config option 'selfview_size'
* vp8: upgrade to draft-ietf-payload-vp8-08
* winwave: cleanup and minor fixes
2013-01-01 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.3
* new modules:
- selfview Video selfview as video-filter module
- vumeter Audio-filter module to display recording/playback level
* config:
- added 'net_interface" to bind to a specific network interface
- added accounts 'regq' parameter for SIP Register client
* baresip:
- added video-filter plugin API (vidfilt)
- audio.c: cleanups, split into transmit/receive part
- ua: added SIP Allow-header (thanks Juha Heinanen)
- ua: added Register q-value (thanks Juha Heinanen)
- ua: fix DTMF end event bug
* avcodec: fix x264 fps bug (thanks Trevor Jim)
* ice: only include ufrag/pwd in session SDP (thanks Juha Heinanen)
2012-09-09 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.2
* new modules:
- auloop Audio-loop test module
- contact Contacts module
- isac iSAC audio codec
- menu Interactive menu
- opengles OpenGLES video output
- presence Presence module
- syslog Syslog module
- vidloop Video-loop test module
* baresip:
- added support for call transfer
- added support for call waiting
- added multiple calls per user-agent
- added multiple registrations per user-agent
- cmd: added new command interface
- ua: handle SIP Require header for incoming calls
- ui: cleanup, use dynamic interactive menu
* config:
- added 'audio_alert' for ringtones etc.
- added 'outboundX=proxy' for multiple outbound proxies
- added 'module_tmp' for temporary module loading
- added 'module_app' for application modules
* avcodec: upgrade to latest FFmpeg and fix pts bug
* natbd: register command 'z' for status
* srtp: fix memleak on close
* uuid: added UUID loader
2012-04-21 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.1
* baresip: do not include rem.h from baresip.h
rename struct conf to struct config
vidsrc API: move size to alloc handler
aucodec API: change fmtp type to 'const char *'
add SDP fmtp compare handler
vidcodec API: added enqueue and packetizer handlers
remove size from vidcodec_prm
remove decoder parameters from alloc
change fmtp type to 'const char *'
add SDP fmtp compare handler
remove aufile.c, use librem instead
audio: fix Telev timestamp (thanks Paulo Vicentini)
configurable order of playback/source start
ua_find: match AOR for interop (thanks Tomasz Ostrowski)
ua: more robust parsing for incoming MESSAGE
ua: password prompt (thanks to Juha Heinanen)
* build: detect amr, cairo, rst, silk modules
* config: split 'audio_dev' parameter into 'audio_player/audio_source'
order of audio_player/audio_source decide opening order
rename 'video_dev' parameter to 'video_source'
added optional 'auth_user=NAME' account parameter
(idea was suggested by Juha Heinanen)
* alsa: play: no need to call snd_pcm_start(), explictly started when
writing data to the device. (thanks to Christof Meerwald)
* amr: more portable AMR codec
* avcodec: automatic size from encoded frames
detect packetization-mode from SDP format
use enqueue handler
* avformat: update to latest versions of ffmpeg
* cairo: new experimental video source module
* cons: added support for TCP
* evdev: added KEY_KPx (thanks to ccwufu on OpenWRT forum)
* g7221: use bitrate from decoded SDP format
added optional G722_PCM_SHIFT for 14-bit compat
* rst: thread-based video source
* silk: fix crash, init encoder, bitrate=64000 and complexity=2
(reported by Juha Heinanen)
* srtp: decode SDES lifetime and MKI
* v4l, v4l2: better module detection for FreeBSD 9
do not include malloc.h
(thanks to Matthias Apitz)
* vpx: auto init of encoder
* winwave: fix memory leak (thanks to Tomasz Ostrowski)
* x11: add support for 16-bit graphics
2011-12-25 Alfred E. Heggestad <aeh@db.org>
* Version 0.4.0
* updated doxygen comments (thanks to Olle E. Johansson)
* docs: added modules description
* baresip: add ua_set_aumode(), configurable audio-tx mode
vidsrc API: added media_ctx shared with ausrc
ausrc API: add media_ctx shared with vidsrc
audio_encoder_set() - stop audio source first
audio_decoder_set() - include SDP format parameters
aufile: add PREFIX to share path (thanks to Juha Heinanen)
natbd.c: move code to a new module 'natbd'
get_login_name: check both LOGNAME and USER
ua.c: unique contact-user with address of struct ua
ua.c: find correct UA for incoming SIP Requests
ua_connect: param is optional (thanks to Juha Heinanen)
video: add video_set_source()
* amr: minor improvements
* audiounit: new module for MacOSX/iOS audio driver
* avcapture: new module for iOS video source
* avcodec: fixes for newer versions of libavcodec
* gsm: handle packet-loss
* natbd: move to separate module from core
* opengl: fix building on MacOSX 10.7
(thanks to David Jedda and Atle Samuelsen)
* opus: upgrade to opus v0.9.8
* rst: use media_ctx for shared audio/video stream
* sndfile: fix stereo mode
2011-09-07 Alfred E. Heggestad <aeh@db.org>
* Version 0.3.0
* baresip: use librem for media processing
added support for video selfview
aubuf, autone, vutil: moved to librem
ua: improved API
conf: use internal parser instead of fscanf()
vidloop: cleanup, use librem for processing
* config: add video_selfview={pip,window} parameter
* amr: new module for AMR and AMR-WB audio codecs (RFC 4867)
* avcodec, avformat: update to latest version of FFmpeg
* coreaudio: fix building on MacOSX 10.5 (thanks David Jedda)
* ice: fix building on MacOSX 10.5 (thanks David Jedda)
* opengl: remove deps to libswscale
* opensles: new module OpenSLES audio driver
* opus: new module for OPUS audio codec
* qtcapture: remove deps to libswscale
* rst: new module for mp3 audio streaming
* silk: new module for SILK audio codec
* v4l, v4l2: remove deps to libswscale
* x11: remove deps to libswscale, use librem vidconv instead
* x11grab: remove deps to libswscale
2011-05-20 Alfred E. Heggestad <aeh@db.org>
* Version 0.2.0
* baresip: Added support for SIP Outbound (RFC 5626)
The SDP Content Attribute (RFC 4796)
RTP/RTCP Multiplexing (RFC 5761)
RTP Keepalive (draft-ietf-avt-app-rtp-keepalive-09)
* config: add 'outbound' to sipnat parameter (remove stun, turn)
add rtpkeep={zero,stun,dyna,rtcp} parameter
audio_codecs parameter can now specify samplerate
add rtcp_mux for RTP/RTCP multiplexing on/off
* alsa: set buffersize and fix samplesize (thanks to Luigi Rizzo)
* avcodec: added support for MPEG4 video codec (RFC 3016)
wait for keyframe before decoding
* celt: upgrade libcelt version and cleanups
* coreaudio: fix buffering in recorder
* ice: several improvements and fixes
added new config options
* ilbc: handle asymmetric modes
* opengl: enable vertical sync
* sdl: upgrade to latest version of libSDL from mercurial
* vpx: added support for draft-westin-payload-vp8-02
* x11: handle remote display with optional shared memory
* x11grab: new video-source module (thanks to Luigi Rizzo)
* docs: updated doxygen comments
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