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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