099909e4a5
2019-06-22 Alfred E. Heggestad
* 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)
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