Format: 1.8 Date: Fri, 09 Nov 2012 23:02:19 -0500 Source: sflphone-common Binary: sflphone-common sflphone-common-dbg Architecture: amd64 Version: 1.2.0-rc20121109~ppa1~precise Distribution: precise Urgency: low Maintainer: Ubuntu Build Daemon Changed-By: Emmanuel Milou Description: sflphone-common - SIP and IAX2 compatible softphone - Core sflphone-common-dbg - debugging symbols for sflphone-common Changes: sflphone-common (1.2.0-rc20121109~ppa1~precise) precise; urgency=low . ** SNAPSHOT 1.2.0-rc20121109~ppa1~precise ** . * video: remove unused frame number variable * daemon: remove unused mailnotify flag * daemon: remove unused/deprecated tests * * #17184: daemon should have no mention of the addressbook * sip: removed unused sip_utils::createRouteSetList function * * #17351: logger: don't use long long unsigned in format string * * #16724: video: fix regression in preview * * #17351: test: add -pthread to flags * * #17351: logger: add pthread_self() output to logs * * #16724: video: don't create a temporary SDP file * * #17340: video: switch to single threaded decoder to handle resolution changes * * #17246: video: call sws_getCachedContext whenever we are about to scale * * #16841: sipaccount: preincrement iterators (++iter), and use const_iterator * #16841: Fix cipher array size determination * * #16808: daemon: Fix configuration file path when XDG_CONFIG_HOME is set * * #16729: video: remove sizeof(unsigned char) * * #16719: pulse: check device index before using * * #16719: manager: lock mutex properly before getting telephone tone * * #16428: audiortp: don't send dtmf if session is NULL * * #16378: pjsip: don't build unused sound module * * #16377: Add build definitions for SH4 * * #16376: gcc 4.7 fixes * * #16230: video: close output context to avoid leaking sockets every call * * #16180: sip: fixed crasher on incoming calls from SIPML web client * * #16007: video: don't leak AVDictionary objects * * #16070: sip: catch SocketError * * #15545: sip: restore old Call API but handle invite with no SDP correctly * * #16023: sip: don't call sip_inv_answer for terminated invite sessions * * #16029: audiortprecordhandler: don't access elements in empty codec list * * #14077: siptransport: default listening address should be anyhosts, not our local IP * * #15867: sipvoiplink: fix status codes for hangup * * #15866: sipvoiplink: simpler invite state checking * * #15866: sipvoiplink: don't try to end invite session if in invalid state * * #14077: sip: put all supported methods in our allowed set * * #14077: daemon/video: better error messages on avformat_find_stream_info failure * * #15545: respond to INVITE that has no SDP with an OK that has an SDP * * #14077: video: defer sdp removal until later * * #14077: sdp/video: less hackish way of parsing SDP * * #15528: audiortp: fix regression from commit 21410c238df235490b3958fec6ac51377dff3f3b * * #15528: audiortp: fixed sdp parsing * * #15528: audiortp: use separate variables for encoder and decoder payload types * * #15641: audiortp: use less CPU intensive for loop for fade in * * #15528: sip/audiortp: support asymmetric audio codecs * * #15528: audiortp: store vector of audioCodecs * * #15528: audiortp: issue warning when receiving wrong RTP packets * * #15493: daemon: fix bad copyright indention in codecs * * #15493: daemon: fix FSF address indentation * * #15493: fix FSF address for daemon/tests * * #15493: fix FSF address for daemon/src/video * * #15493: fix FSF address for daemon/src/sip * * #15493: fix FSF address for daemon/src/{history,hooks,iax,im} * * #15493: fix FSF address for daemon/src/{dbus,config} * * #15493: fix FSF address for daemon/src/audio * * #15493: fix FSF address for daemon/src/ * * #15503: update copyright for daemon/test * * #15503: update copyright for daemon/src/video * * #15503: update copyright for daemon/src/sip * * #15503: update copyright for daemon/src/im * * #15503: update copyright for daemon/src/iax * * #15503: update copyright for daemon/src/hooks * * #15503: update copyright for daemon/src/history * * #15503: update copyright for daemon/src/dbus * * #15503: update copyright for daemon/src/config * * #15503: fix copyright years for daemon/src and daemon/src/audio/* * * #15502: audio: removed unused echosuppress/echocancel code * * #14529: sip: don't overwrite routeset in hangup * * #15456: sip: remove unused contact header field * * #14529: sip: store addresses as vector since we want to preserve order * * #14529: sip: end session with code = 0 * * #14529: sip: don't resolve host URLs for outgoing calls * * #14529: sip: removed unused updateContactHeader code * * #14529: sipvoiplink: removed unused pool * * #14529: sip: minor fixes in getIPList * * #14529: sip: add list of routes (IPs of proxy service route) to BYE request * * #15266: dtmf: Use correct event codes for DTMF over RTP * * #14529: sip: move getAccountIdFrom... to sipvoiplink * * #15091: daemon: cleanup configure.ac * * #14615: managerimpl: simplify API for getting all accounts * * #8034: configure: ilbc now enabled by default * * #8034: ilbc: convert to float before encoding/after decoding, and ilbc_decode must be called in "normal" mode * * #8034: audiocodec: remove unused bandwidth_ * * #8034: ilbc: cosmetics * opus: cosmetics * * #8034: pjsip: added shell script to compile pjsip * * #8034: daemon: use ilbc implementation that comes with pjsip * * #14910: Use PCRE CFLAGS, not PCRE_LIBS * * #14603: audiocodecs: resolve all undefined symbols before dlopen() returns and fail if this cannot be done * * #14603: fix includes for opus * * #14603: g729: load symbols in constructor * * #14603: opus: fix double free and lifetime issues * #14881: Add mode ZEROS for playback which only provide empty buffers to playback engine * * #14603: opus: avoid payload type conflicting with video * * #14602: opus: only import functions that we need to call directly * * #14602: opus: cosmetics * * #14885: mainbuffer: lock mutex when dumping info * * #14668: manager: catch Socket * which is thrown when sockets can no longer be bound * #14818: Get MainBuffer as a reference in iaxvoiplink * #14881: Should be a pointer to a AudioLayer * #14881: Set audio playback mode for answer/hangup actions * #14881: Init playbackMode to NONE * #14881: Add playback modes in audio layer * #14848: rename samplerate variable in manager when loading a file for clarity * #14848: Remove explicite reference to audioCodec->getFrameSize move transportRate to audioRtpSession instead of audioSymmetricRtpSession * #14818: Get MainBuffer as a reference instead of a pointer * #14615: Fix compilation without iax * * #14615: manager: voiplinks should cleanup their own accountmaps * * #14727: video: video receive thread should not call pjsip or touch the callmap * * #14615: sipvoiplink: make some data and methods non-static * * #14745: gnome: make sure codecs are set to active if in the "active" list * * #14569: yamlparser: guarantee that the file exists for the duration of yamlparser's lifetime * #14615: Fix compilation without iax * * #14615: siptest: fix tests for new API * #14615: Use DEBUG instead of ERROR for getVoipLink * * #14615: iax: only clear call map if removing the last account * * #14615: iax: don't explicitly call constructors * * #14615: sipvoiplink: don't need to invoke constructors explicitly * #14615: Make sure the account is valid when returning the associated voip link * #14615: Make sure the account pointer is valid in placeFirstAccountCall * #14615: Fix default audio codec when creating new account * * #14687: video: cleanup SDP files immediately * * #14687: video: avoid deadlocks in VideoReceiveThread * #14615: Make the serializable interface use a reference to a yamlnode instead of mappingnode * #14615: Use static cast for sessionAudioMedia since used only for audio codecs... * #14615: Use static_cast in getCurrentAudioCodecName (same as video) * #14615: Remove dynamic casts from audio_rtp_factory * #14615: Remove dynamic_cast associated to account in callmanager * #14615: Use getSipAccount in configuration manager instead of getAccount and cast * #14615: Make RTP sessions to return empty vector if not a secured session * #14615: Get sip account directly in audio rtp factory * #14615: Fix yaml to use references instead of pointers * * #14669: sipvoiplink: don't try to set a dialog's transport if either of them are NULL * #14615: Add getCall for iax and sip, which prevent from dynamic_cast * #14615: Clear call map in voip destructorw * #14615: Remove callMap from VoipLink to IAX and SIP links respectively * [ #14666] rtp: make transport rate dynamic * #14615: Move SIP and IAX account maps to their respective voip link * #14615: Move AccountMap definition from ManagerImpl Account * [#14602] codec/rtp: Change delay from 20 ms to 10 ms (G729 and Opus) * [#14602] alsa: Better handle alsa error detection * #14615: Make audiocodec factory to deal with casting to Codec to AudioCodec * * #14605: preferences: fix error message * #14615: Remove dynamic_cast for getAccount in IAXVoIPLink * #14615: Remove dynamic cast related to getAccount in SIPVoIPLink * #14615: Add tow different map to store SIP and IAX account separately * * #14631: history: store codecs used in call * [#14602] codec: free opus codec memory in destructor * [#14602] codec: Multiply input buffer to convert 16bit->2x8bit * [#14602] codec: Changes values back to 16kHz * * #14605: audiopreferences: don't allow config file to set a non- writable recording path * #14615: Remove dynamic casting in yaml configuration, using inheritance instead * * #14605: audiorecord: don't allow non-writable paths to be set for recording * * #14605: audiorecord: use safer HOMEDIR macro * * #14605: audiorecord: remove non-filename characters from filename before recording * * #14600: SIPAccount: refactor account matching * #14607: dlerror returns const char * instead of char * on certain implementation * [#14602] codec: Ajust OPUS size to 12k (untested) * [#14602] codec: Ajust OPUS to work at 48k * #14569: Use static instance for main document in yaml parser * [#14602] codec: Initial Opus implementation (untested) * #14569: Use dynamic cast instead of static ones in yaml parser * * #14568: gnome: auto answer feature working * #14569: Refactor configuration file existence test and exception handling * [ #14363 ] Add proper dynamic detection, add 'init' capability to codecs. If there is an exported init function, then it is called at runtime to check loading * [ #14363 ] Fix decoder * [ #14363 ] Change decoder value to 160 * [ #14363 ] Split G729 frame in 2 * [ #14363 ] Fix some G729 parameters, still can't proprely send audio * * #14363: g729: add create and destroy functions for dl_open to find/use * [ #14363 ] Add to factory * [ #14363 ] G729 now compile * [ #14363 ] Innitial code for g729 support Checksums-Sha1: e7ec5810d95d961fcb2093fc838f5fd4d2aacda2 1050108 sflphone-common_1.2.0-rc20121109~ppa1~precise_amd64.deb 2b2e917de939b79cff70656cef91e3283e3313ea 5464572 sflphone-common-dbg_1.2.0-rc20121109~ppa1~precise_amd64.deb Checksums-Sha256: f4333e6652d290221d4373382855d5261745df16f318ea4ab4c304b54a9ed7d9 1050108 sflphone-common_1.2.0-rc20121109~ppa1~precise_amd64.deb b6476ee8bddc266c4dc3fe1d8dd60a73c386f6d52237faeebdd3bc00bd30a2a3 5464572 sflphone-common-dbg_1.2.0-rc20121109~ppa1~precise_amd64.deb Files: 7b281051d0cd90299db6be022b64b3d9 1050108 gnome optional sflphone-common_1.2.0-rc20121109~ppa1~precise_amd64.deb e5ed0e7d973fd49662e85273a868411b 5464572 debug extra sflphone-common-dbg_1.2.0-rc20121109~ppa1~precise_amd64.deb