Format: 1.8 Date: Wed, 28 Aug 2013 23:18:02 -0400 Source: sflphone-common-video Binary: sflphone-common-video Architecture: i386 Version: 1.2.3-rc20130828~ppa1~saucy Distribution: saucy Urgency: low Maintainer: Ubuntu Build Daemon Changed-By: Emmanuel Milou Description: sflphone-common-video - SIP and IAX2 compatible softphone - Core with video support Changes: sflphone-common-video (1.2.3-rc20130828~ppa1~saucy) saucy; urgency=low . ** SNAPSHOT 1.2.3-rc20130828~ppa1~saucy ** . * [ #14268 ] daemon: Fix a race hash collision when creating new accounts too fast (fix last commit) * [ #14268 ] daemon: Fix a race hash collision when creating new accounts too fast * #2918: fix SHMSink::resize_area() too often called * sdp: fix inverted error message logic * * #29158: sip: don't use received port if received port is 0 * sip: added debug message for answer * sip: validate STUN config * * #26628: sip: limit cipher string length to 1000 * * #28529: daemon: halve boolean port array * * #28529: daemon: better validation of port ranges * * #29034: SDP: update SDP struct on STUN update * * #29033: video: better log if socket bind fails * * #29024: dtmf over rtp: only increment timestep for new events * * #29004: dtmf over rtp: send 3 identical end packets * * #29006: dtmf: set volume value to -10 dBm. * alsa: log error message if snd_ctl_pcm_info fails * * #28675: sip: sipcall shouldn't know about sipaccount * sdp: cleanup * * #27724: test: fix sdp test * * #28691: pulseaudio: cleanup * * #28691: pulseaudio: fix resampling of captured mic input * * #28679: audio: don't create a new scratch buffer every time we resample * * #28679: audio: use correctly sized buffer for resampling * * #27724: sdp: use published IP address for STUN * * sip: #28675: add correct contact header before sending OK * * #27724: sdp: remove old contact header (if present) * * #28529: sip: fix port range unserialization * * #28529: sip: fix compilation for gcc 4.6 and off-by-one error * * #28529: sip: RTP ports now controllable via D-Bus * * #28529: sip: fix build for -std=gnu++0x * * #28529: sip: fix build for gcc < 4.8 * * #28529: sip: serialize/unserialize port ranges * * #28529: sip: ensure that ports are unique * * #28529: sip: fix build when video is disabled * * #28529: sip: add members for RTP port ranges * * #28508: daemon: make configure errors/warnings distro agnostic * * #28351: mainbuffertest: use unique_ptr instead of shared_ptr * * #28351: sdes_negotiator: use unique_ptr instead of shared_ptr * * #28351: managerimpl: use unique_ptr instead of shared_ptr * * #28351: managerimpl: use item instead of iter * * #28351: managerimpl: use range-based for loop * * #28351: mainbuffer: use range-based for loops * * #28351: daemon: rename iter to item for range-based for loops * * #28351: instantmessagingtest: use range-based for loops * * #28351: audiocodecfactory: use range-based for loops * * #28351: audiocodecfactory: use range-based for loops * * #28351: sdes_negotiator: use range-based for loops * * #28351: sdes_negotiator: use range-based for loops * * #28351: audiortprecordhandler: use range-based for loops * * #28351: libav_utils: use range-based for loops * * #28351: videov4l2: use range-based for loops * * #28351: videov4l2list: use range-based for loops * * #28351: numbercleaner: use range-based for loops * * #28351: iaxcall: use range-based for loops * * #28351: iaxvoiplink: use range-based for loops * * #28351: history: use range-based for loops * * #28351: historyitem: use range-based for loops * * #28351: historynamecache: use range-based for * * #28351: yamlemitter: use range-based for loops * * #28351: yamlnode: use range-based for loops * * #28351: sfl_config: use range-based for loop * * #28351: yamlparser: use range-based for loops * * #28351: conference: use range-based for loops * * #28351: network manager: use range-based for * * #28351: im: use range-based for loops * * #28351: sipaccount: use range based for loops * * #28351: mainbuffer: use range-based for loops * * #28351: account: use range-based for loop * * #28358: sip: ensure correct Contact Header is present for RINGING and OK * * #28358: sip: move addContactHeader into separate function * pulseaudio: cleanup * * #28351: sdp: use range based for loops * * #28351: sipvoiplink: use range-based for loops * * #28351: manager: use range-based for loops * * #28351: daemon: switch to C++11 * * #28344: audiofile: cleanup * * #28344: audiofile: add support for OGG Vorbis ringtones * * #28344: audiofile: add support for FLAC ringtones * * #28334: video: use &buffer[0] for string * * #28334: video: use vector::data() * * #28334: siputils: use vector for direct character buffer writing * * #28334: test: use vector where contiguous character buffer access is assumed * alsa: cleanup * * #28334: audiosrtpsession: use vector::data() * * #28334: audiocodec: use vector::data() * * #28334: audiocodec: remove unused dest offset argument * * #28334: audiocodec: use references rather than pointers * * #28334: main: use vector::data() * * #28334: tone: use vector::data() * * #28334: sipaccount: use vector::data() * * #27819: audio: fix playback for stereo files that require resampling * * #27819: audio: cleanup * * #27819: audiofile: fix stereo file playback for files that don't require resampling * * #27819: audiofile: fix variable name * * #27819: samplerateconverter: cleanup * * #27819: audiofile: merge RawFile and WaveFile into AudioFile * * #27819: audiofile: use sndfile for .au, .al and .ul files * * #28215: sip: on incoming calls, go to TRYING before going to RINGING * * #28047: audioloop: fix memory leak * Revert "* #28047: audioloop: allocate buffer on stack and fix memory leak" * * #28212: manager: on configuration error, automatically restore last working configuration file backup * * #28047: audio: cleanup * * #28047: audioloop: allocate buffer on stack and fix memory leak * * #28047: gnome: fix seekbar update * * #27819: audiobuffer: fix make check * Revert "* #27819: audiobuffer: iterate more efficiently" * * #28047: daemon: remove ringtone API from D-Bus * * #28047: ringtone preview and playing recorded calls now share the same code * * #27819: audio buffer: avoid # of channels bugs * * #28131: pulse: cleanup * * #28131: pulse: fix playback * * #27819: audiobuffer: rename fromInterleaved -> deinterleave * * #28047: daemon: add API to start/stop ringtone * * #27819: audiobuffer: avoid memory leaks when replacing buffer_ * * #27819: audiobuffer: iterate more efficiently * * #27819: audiofile: use sndfile for wave playback * * #27819: audiorecord: remove unused RAW format * * #27819: audiorecord: remove unused buffers * * #27819: audio: record files with libsndfile * * #26842: audio: use types consistently * * #27184: audio: fix recording * * #27693: audio: cleanup * * #27693: audio: fix audiobuffer copy constructor * * #27693: audio: cleanup setChannels * * #27693: audio: use samples size directly * * #27693: audio: remove duplicate method * * #27693: audio: opensl code should only be in OpenSLLayer * * #27693: audio: fix tests for new getChannel signature * * #27693: audio: hold a reference to main buffer instead of a pointer * * #27693: audio: fix variable names * * #27693: audio: remove redundant clear method * * #27693: audio: cleanup * * #27693: audio: getChannel should always takes a parameter * * #26842: audio: remove superfluous typedefs * * #27693: audiolayer: gain value should be neither static nor public * audioloop: cleanup * * #27570: daemon: make user-agent setting persistent * * #27570: account: fix default user-agent logic * daemon: tests: removed unused sample file * * #27558: audiortp: don't fail silently if destination is not set * * #27557: audiocodecfactory: flush dlerror() everywhere * * #27528: logger: fix ERROR log macor * * #27530: sdp: add error checking * * #27450: audiolayer: cleanup * * #27528: logger: fix logging for format strings with args * * #27528: logger: add more info to android logging, cleanup macros * * #27450: opensl: cleanup * * #27450: opensl: cleanup * * #27450: opensl: use pthread directly instead of ost::Thread * * #27450: opensl: remove unused includes * * #26839: sip: Android can use new pjsip API * * #27450: opensl: remove dead code, make methods private * * #27450: opensl: avoid pointless double assignment * * #27450: opensl: remove dead code * * #27450: opensl: cleanup * * #27450: opensl: use AudioBuffer's reset method * * #27450: opensl: move capture to a method * * #27450: opensl: audioPlayback done through method * * #27450: opensl: format with astyle * * #27450: opensl: cleanup array initialization * * #27450: opensl: remove dead code * * #27450: opensl: private members should end with _ * * #27243: fixed default audio output * managerimpl: use private members directly * * #27352: history: fix include for strerror * * #27352: daemon: add DBUSCPP CFlags to appropriate Makefiles * * #27362: changed history path for android * * #27362: history: Stop reading history file as soon as limit is hit. * * #27362: Fixed issue, put back managerimpl.i * * #27352: daemon: relax dbus pkg-config requirement * daemon: updated gitignore files * * #27352: daemon: fix dbus pkg-config check * * #27352: daemon: android client has no need for dbus * * #27352: daemon: dbus is now optional * android: Fixed missing method declaration * Removed managerimpl.i * daemon: added build-aux to .gitignore * * #27364: client: merge APIs * * #26839: client: hide DBus only includes * * #26839: client/android: add updatePlayback signal * Updated configure-android * * #26839: client: move xml into client, since it's shared * * #26839: client: move video_controls.h into client * * #26839: client: move configurationmanager.h into client/ * * #26839: client: move callmanager.h into client * * #26839: dbus: move into client/dbus * video: remove unused variable * * #26839: daemon: remove Android.mk files * * #26839: preferences: fix formatting with astyle * * #26839: preferences: fix preprocessor logic * * #26839: video: fix includes * * #26839: tests: fix missing includes * * #26839: audiocodecfactory: use ifdef to be consistent * * #26839: sipvoiplink: cleanup * * #26839: siptransport: cleanup * * #26839: sipaccount: cleanup * * #26839: managerimpl.h: cleanup * * #26839: sipvoiplink: cleanup * * #26839: managerimpl: cleanup * * #26839: daemon: remove dead code * * #26839: history: remove useless debug * * #26839: global.h: cleanup * * #26839: callmanager: cosmetics * * #26839: client: cosmetics * * #26839: audiocodecfactroy: remove redundant check * * #26839: audio: use standard types (not ccrtp types) * * #26839: remove unused header * * #26839: audiortprecord: format with astyle * * #26839: audioloop: cosmetics * * #26839: audiolayer: more constness * * #26839: audiolayer: format with astyle * * #26839: audiolayer: more constness, fix copy constructor * * #26839: android: buffer should reset itself * * #26839: android: cosmetics * Revert "* #27232: android: remove unused BUILD_ALSA" * * #27232: android: remove unused BUILD_ALSA * * #27232: android: Add configure script * * #26839: android: delete unused header * * #26839: android: fix paths for jni files * * #26839: daemon: load history limit at startup * * #26839: daemon: ensure that HAVE_DBUS is defined * * #26839: audiolayer: fix warnings * * #26839: daemon: fix build after android merge * * #27201: libiax: avoid out of bounds read in a more obvious way * * #27201: libiax: fix memory leak * * #27196: audiofile: fix mismatched delete * * #27201: libiax: removed dead code * * #27201: libiax: remove redundant NULL check * * #27201: libiax: remove redundant NULL check * * #27201: libiax: guarantee that buffer is NULL-terminated * * #27232 Modified build process and organisation of JNI layer * * #27201: libiax: guarantee that buffer is NULL-terminated * * #27201: libiax: fix out of bounds memory access * * #27201: libiax: remove dead code * * #27201: libiax: don't pass out of bounds pointer to memmove (minbuf) * * #27201: don't pass out of bounds pointer to memmove * * #27201: iax: fix memory leak * * #27201: iax: fix uninitialized value * * #27196: pulse: pass structs by const reference * * #27196: audiofile: fix illegal delete [] * * #27196: audiofile: fix memory leak * * #26839: iax: fix warnings * sdp: fix warning if compiling without video support * * #26839: audio: fix formatting * * #26839: audio: fix warnings * * #26839: audiobuffer: fix warnings * * #26839: audio: fix warning * * #26839: pulse: fix TODO * * #26839: pulselayer: fix formatting with astyle * * #26839: ringbuffer: fix formatting * * #26839: ringbuffer: fix warnings * * #26839: nameComparator: don't inherit from deprecated unary_function * * #26839: audiofile: fix formatting, delete commented code, add FIXME * * #26839: audiofile: fix warnings * * #26839: audiortp: fix warnings, formatting * * #26839: opus: fix warnings, formatting * * #26839: audiobuffer: fix formatting with astyle * iax: downgrade false positive errors to debug messages * * #27096: iax: use union instead of dereferencing type-punned pointer * * #27095: iax: zero out IAXContext correctly * * #26839: speexcodec.h must include global.h * audio: define SFLDataFormat as int16_t and use SFLDataFormat instead of short * * #26839: alsa: fix comments * * #26839: audio: fix formatting with astyle * * #26807: pulse: fix build for older pulseaudio (< 1.0.0) * * #20661: opus: ensure that HAVE_OPUS is defined * * #26896: daemon: remove eclipse project files * * #25537: daemon/dbus renamed daemon/client * * #25537: daemon: allocate bus dispatcher on heap * * #25537: daemon: move dbusmanager.* to client.* * * #25537: daemon: rename DBusManager => Client * * #25537: daemon: rename dbus_ to client_ * * #26807: pulse: create context with pa_context_new_with_proplist * * #26796: daemon: remove managerimpl_registration * * #14255: dbus: added "started" signal for when daemon is running * * #11942: dbus: always notify client when recording playback stopped * sipaccount/tls: cleanup * * #26628: TLS: limit number of ciphers to avoid abort in pjsip * * #Added callback isConferenceParticipant * * #7078: audio: suspend audio processing if peer hungup and no calls remain * manager: no need to call .get() on shared_ptr * * #7037: manager: added comment explaining checkAudio() * * #26544: dbus: improve recording API * * #7037: audio: stop audio stream if user starts then stops dialing * daemon: removed dead code * * #26184: daemon: stop ringing if there are no more waiting calls * * #26453: sip: use HAVE_TLS around all ssl dependent code * * #26302: pjsip: fix build of daemon for bi-endian systems * * #26416: daemon: warn if pkg-config is missing * * Changes on conference callbacks * * #26295: sip: don't crash if transport could not be created * audiortp: fix variable name * * #26183: sdp: add telephone-event payload to media description * * #25974: pjproject add all needed configure files * * #25974: pjsip: added configure script to tree * * #25974: daemon: fix pjproject path * * #25974: daemon: update to pjsip 2.1.0 * * #26115 Modified callbacks * * #25528: audio: both macros must have updated version numbers * [#25528] Update doc with new version number, and jenkins script * * #25946: sip: fix calling IP2IP calls from history * * #25935: pulse: don't add duplicate devices to list * * #23661: remove audio stream if no calls remain * * #25921: daemon: only backup valid configuration files * * #25916: sip: restrict RTP port ranges * * #25295: sip: update published address for STUN * * #25295: audiortp: get ports from STUN * sipvoiplink: cleanup * * #25295: sipaccount: remove dead code * * #25295: audiortp: allow socket descriptor retrieval * * #25900: sip: fixed broken stripPrefix code and added regression test * * 25787: sip: use STUN address in VIA sent-by * * #25656: preferences: noise suppress should be off by default * * #25472: sip: fix port number calculation * * #25367: dbus: remove isValidCall from D-Bus interface * audiortp: remove verbose log * * #25393: siptransport: fix Log crash * manager: cleanup * * #23661: daemon: use getAccountID directly * * #23661: manager: use isValidCall * * #23661: daemon: restore getCallList * * #23661: audiortp: get accountid from call * account: move split_string into Account * manager: move join_string into Account. * * #23661: daemon: removed unneeded hasCalls * managerimpl: cleanup * * #23661: daemon: removed callAccountMap * * Extracted JNI callbacks in new file * Added accounts state changed jni callbacks * * #25242: daemon: close codec handle on exception * * #25237 Resolved multiple issues * * #25012: video: fix typo in comment * numbercleaner: fix comparison * * #23666: daemon: ensure that calls are only in one conference at a time * managerimpl: simplify detachParticipant * managerimpl: remove unused parameter * * #23739: daemon: fix unhold conference * managerimpl: cleanup * * #25076: sip: fix forward declarations of structs * * #25073: video: catch exception if socket fails * * #25021: pad SHMHeader so that shared memory buffer is 16-bit aligned. * * #24936: socket: fix build for g++-4.6 * * #24936: socket: initialize all member variables * * #24936: socket_pair: clean up socket if bind failed * * #24936: socket: remove dead code and gotos * * #24936: video: fix NULL check found by coverity * * #24936: socket_pair: fix unsigned comparison found by coverity * * #24936: video: fix NULL check found by coverity * * #24936: v4l2: fix uninitialized array found by coverity * * #24936: video: fix bug found by coverity * Added log in opensllayer Added getCallList in callmanager.i * callmanager: cleanup xml tags * callmanager: remove unused placeCallFirstAccount * * #24789: callmanager: placeCall returns bool * * #24789: callmanager: return bool in joinConference * * #24789: callmanager: return bool in joinParticipants * * #24789: callmanager: return bools in detachParticipant * * #24789: callmanager: add/addMainParticipant return bools * * #24789: callmanager: hold/unholdConference return bools * * #24789: callmanager: accept/refuse now return bool * * #24789: callmanager: return true on success for hold/offhold * * #24789: dbus: hangupCall/Conference returns a boolean * * #24789: daemon: return true if hangupCall succeeded * * #14077: video: send and receive RTP on one socket * * #24637: codecs: clear dlerror() before loading codecs * * #24613: iax: Fix SEGFAULT when call is not found * * #24212 Fixed codec linking * * #24121 Additionnal changes to IM integration * * #24121 Saving before sipvoiplink merge * * #24106: sip: fix instant messaging regression * * #24017 Modified src/Android.mk to include im module * * Modified callmanager.i and JNI layer to implement conference callbacks * Small changes in daemon src, related to #23448 Modified callmanager.i to expose attended transfer and conference API * * #23362 Modified swig interface for getHistory implementation * #23361 Added log in siptransport.cpp * build fixes * Modified README * comments for Android NDK presentation * Opus changes * Use the first PA device if the prefered one is not available. * updated copyright headers * cleanup rtp_record_handler; iostream overloads for AudioStream * SDP protocol support for Opus; using only one codec class for mono/stereo in Opus * First Opus stereo implementation * channel number information transmitted through dbus and printed in gnome client * Remove DEBUG called too frequently * RingBuffer backed by one AudioBuffer * code cleanup * Use a dedicated structure to hold PA device infos instead of pa_sink_info and pa_source_info * no more dependency to AudioBuffer in codec shared libraries * minor changes * unit tests for audiobuffer * unit test conversion * use references instead of pointers to AudioBuffer * multichannel compiles * * #21631: daemon: void NULL pointer dereference on unexpected case * * #21631: siptransport: guarantee that ifr_name is NULL-terminated * * #21631: audiozrtp: fix broken override * * 21631: sip: avoid buffer overrun * * #21631: audiosrtpsession: fix broken override of set salt and key methods * * #21631: audiorecord: check if fseek succeeded * * #21631: alsa: fixed error case * * #21631: yamlemitter: don't close invalid file descriptor * * #21631: yamlparser: don't use NULL pointer * * 21631: iax: NULL check calling_number before using it * * #21631: manager: don't answer fake calls * * #21507: sipcall: handle exception if audiortp is no longer in scope * * #21631: sipvoiplink: fix broken NULL check * * #21631: mainbuffer: NULL check call id set * * #21631: manager: NULL check account before using it * * 21631: sipvoiplink: NULL check account before using it * * #21631: audiocodecs: don't use codec handle until it's NULL- checked * * #21631: sipaccount: don't use account before NULL check * * #21631: yaml: add all members to initializer list * * #21631: daemon: remove dead code * * #21631: daemon: fix off by 1 error in account * * #21631: audiocodecfactory: don't leak codec handle if dlopen fails * * #21631: daemon: don't allow access to deleted call * * #21631: fileutils: don't return pointer to function-local object * * #21625: daemon: fix various warnings * * #21507: daemon: get audio codec names from audiortpfactory, not SDP * * #21556: sip: re-register if registration attempt failed with timeout * dbus: fix bug in name-for-bindings * * #21496: daemon: warn if libtoolize or autoreconf are missing * * #21210: sip: don't print an error message every time we add a call * initial ALSA adaptation to multichannel * * #21275: video: fallback to av_freep for older libavcodec * * #21275: video: allocate/deallocate, and reset frames properly * * #20736: daemon: cleanup coverage flags * * #20736: hooks: added test, fixed code * * #20736: daemon: added --enable-coverage option and make targets * * #21210: daemon: avoid bogus error messages when dialing * * #21210: daemon: debug instead of err for "hangup of non-created call" case * * #21210: dbus: added isValidCall method * first set of change to support multichannel * * #21184: daemon: remove unused echo state * audiorecord: cleanup * * #21171: history: readability fix * * #21171: daemon: Always use get_home_dir() * * #14350: daemon: fix broken zidfile path * * #17498: video: restrict RTP port to [10500-20000] * * #20940: sip: Do nothing if non-existant call is requested, just print an error * * #20851: daemon: backup configuration file on errors * * #20793: accounts: enforce stricter rules for account loading * * #20793: yamlnode: ensure that const char * arguments call correct constructor * * #20802: dbus: return instead of exitting * * #20461: ulaw/alaw: use real size parameters * * #20661: gsm: don't use magic numbers for encoded size * * #20785: sip: handle exceptions thrown when transferring with a bogus account * * #20712: sip: handle transfer status updates correctly * * #20712: sip: remove redundant pointer check * * #20661: opus: fix Makefile.am flags * * #20627: video: fix #ifdef for older libavutil * * #20661: opus: use system opus if available * * #20661: gsm: include iostream for printing to cerr * * #20661: audio: catch exception if gsm can't be instantiated * * #20661: codecs: remove superfluous "virtual" declarations * * #20661: audio/codecs: only constructors/destructors are publicly visible * * #19940: account: get rid of "type" field * * #20627: daemon: don't distribute unused opus files * * #20627: daemon: fix make distcheck * * #20627: daemon: fix autogen warnings * * #20627: video: fix build for older libavutil * * #20627: video: fix build for older libav/ffmpeg * * #20603: daemon: fix fast picture update * * #20595: daemon: fix automake warnings * v4l: fix compilation on Debian Sid * * #19964: sip: don't let via_addr go out of scope * * #19649: cosmetics: avoid using memset * * #18944: daemon: fix tests, print message to stderr if pidfile creation fails * * #18944: daemon: create, lock and remove pidfile properly * * #19242: codecs: use system ilbc instead of pjsip's * * #19246: build: bump version numbers for 1.2.2 * * #19058: account: ringtone should be enabled by default * * #19129: iax: fix deadlock on registration * * #19032: sip: don't decrement transport reference on error * [#19045] Update audiocodec library version * [#19045] Update code with new release number (1.2.1) * * #18663: audiocodec: only encode/decode and destructor should be virtual * * #18776: audiortp: remove unused inheritance..more cleanup * * #18668: audiortp: remove duplicated code * * #18668: audiortp: use pthreads, favour composition over inheritance * * #18668: daemon: removed cc_thread.h header * * #18668: pulse: use usleep instead of ost::Thread::sleep * * #18668: alsa: use pthread instead of ost::Thread * * #18668: audiorecorder: use pthread instead of ost::Thread * daemon: audiozrtp: hide spurious warning * * #18668: iax: use usleep * * #18668: daemon: only join threads that have been created * * #18668: manager: use pthread_mutex_t * * #18668: manager: fix audiolayer synchronization * * #18668: audio_rtp_record_handler: use pthread_mutex * * #18668: audiortpfactory: use pthread_mutex * * #18668: iaxvoiplink: use pthread_mutex * * #18668: audiolayer: use pthread_mutex * * #18668: mainbuffer: use pthread_mutex * * #18668: sipvoiplink: use pthread * * #18668: call: use pthread_mutex * * #18668: history: use pthread_mutex * * #18668: daemon: move scoped lock into its own files * * #18667: video_v4l2_list: don't leak mutex * * #18666: eventhread: use pthread instead of cc++/ucommon thread * * #18644: video: remove any reference to cc++/ccrtp * * #18664: video: fix compiler warnings * * #18663: daemon: remove Codec class * * #18631: video: cleanup capture and encoding * * #18603: libs: disable video when building pjsip * * #18627: v4l2_list: fix initialization order warning * * #18581: sip: use pjsip_regc_set_via_sent_by to re-register after receiving a 606 error code * * #18603: daemon: migrate to pjsip 2.01 * * #17588: video: shm buffer must be sized for BGRA textures * Revert "* #18494: sip: don't create new transport unnecessarily" * * #18494: sip: ensure that transport is non-NULL before decrementing it * * #18494: networkmanager: remove unused enum * * #18569: history: moved into the sfl namespace * * #17588: video: use proper avpicture_get_size calls * * #18494: sip: don't create new transport unnecessarily * * #18494: sip: remove unused method * audiortp: remove superfluous namespace names * * #17588: video_send_thread: calculte encoder buffer size correctly. * * #17588: video: use the raw frame, not the scaled frame. * * #17558: video: fix local video dimensions and pixel format * * #17558: video: move buffer size code into VideoProvider * * #17558: video: shm buffer filling callback should be a pure virtual method * * #17558: video: align members vertically * * #17558: video_receive: refactor * * #18294: video/shm: permissions need not be member variables * * #17558: videosend: refactor * * #18294: video: shared video memory should not be group readable. * * #17558: video: cleanup * * #18286: sip: don't set ports to random values unless they are uninitialized * * #18261: sdp: add RTCP info (for audio) to SDP * * #18261: sdp: fix function name * * #18261: audiortp: use forgetDestination in the same manner as addDestination * * #18262: audiortp: remove unused return codes * * #18237: sip: respond with 415 error code if dealing with unsupported media * * #17601: v4l: set probing to true before launch v4l thread * * #17647: pjsip: remove unused files and modules * * #17601: video: remove cc++ thread tests since we don't use them anymore for video * * #17601: v4l2_list: use pthreads instead of cc++ threads * * #17601: video: make receive_thread exit if v4l device was not opened * * #17601: video: use atomic functions to modify shared variables while threads are running * * #17601: video: send/encode thread uses pthreads instead of cc++ threads * * #17601: video: don't use atomic counter as boolean flag * * #17601: video_receive_thread: use pthreads instead of cc++ threads * * #17601: video: cleanup video resources in video threads, not main or sip threads * * #17601: video: fix concurrent access to libav/ffmpeg with pthreads primitives * * #17601: video: close video device even if we haven't parsed streams yet. * * #17601: video: remove unused code * * #16856: sdp: look for connection in SDP and SDP media * video: use consistent naming * 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 * #17208 Include password in along with username when calling getAccountDetails in daemon * #15835: Fix first ougoing call fails to instantiate codec * #17001: Add Call Type (incomming/outgoing) to call state * * #16841: sipaccount: preincrement iterators (++iter), and use const_iterator * #16852: Implement on_call_state_change * #16852: Add new_call_created signal handler * #16841: Fix cipher array size determination * * #16808: daemon: Fix configuration file path when XDG_CONFIG_HOME is set * * #16729: video: remove sizeof(unsigned char) * #16702: Implement account state chaged callback * * #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 * #16032: Add %template(StringVect) vector in jni * * #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 * #15939: Add configuration manager methods * #15939: Separate jni_interface.i in three files (call and config) * #15939: Rename callmanager.i to jni_interface.i * #15939: Include std::vector and std::map in callmanagerJNI.h * #15985: Add getAccountDetails call in configuration manager * #15939: Add configuration Manager in SFLPhone android service * * #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 * #15779: enable logger capability in swig * #14652: rename sflphoneservice and handle new "service" java package * * #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 * #14652: attach current thread as daemon in SWIG * #14652: export CallManager services to sflphone-service * * #15528: sip/audiortp: support asymmetric audio codecs * * #15528: audiortp: store vector of audioCodecs * * #15528: audiortp: issue warning when receiving wrong RTP packets * #15414: run swig directly from Android makefiles, handle deps * #14399: add logs to various constructors * #15414: Generate SWIG JNI interface for basic functions * #15414: remove original JNI calls before swig implementation comes * * #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 * #15416: catch exceptions while parsing arguments * #14399: remove manager unused methods * #14650: stub run() method used only with dbus * * #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 * #15416: initial xml parser to generate sflphone-service.i SWIG file * * #15266: dtmf: Use correct event codes for DTMF over RTP * #15223: Fix audiortp too slow bug using usleep added audio record- to-disk debug * #15162: make incoming call from java UI * #14371: Add logger.h in opensllayer.cpp * #14371: Use Android Logging system * #14371: Add delay in capture callback * #15046: implement hangUp * #15046: release string pointers in placeCall * #14371: Uncomment start capture in opensl * #14650: release string pointers as java's GC does not do it * * #14529: sip: move getAccountIdFrom... to sipvoiplink * * #15091: daemon: cleanup configure.ac * * #14615: managerimpl: simplify API for getting all accounts * #15046: implement placeCall * #14741: stub history to complete manager init, fix warning * #14399: fix debug message * #15046: make outgoing call from java UI * #14698: get SIP log level from java env * #14399: remove warnings and clean up * * #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 * #14650: fix manager init from Java UI and add JNI_OnUnLoad * #14741: fix history path * * #14910: Use PCRE CFLAGS, not PCRE_LIBS * #14987: Move sleep in opensl callback instead of FillAudioBuffer * #14987: Remove common implmentation from opensl layer * #14987: Move common audio logic to AudioLayer * #14917: fix example JNI functions * #14919: Rename output files for debuging * #14917: example JNI functions with callback * #14650: change android configdir * #14741: function to get env variable * #14399: add debug in SIP * #14881: Playback fully implemented, fixed buffer size, add dump to disk * #14881: Fix audio thread for OpenSL * #14881: Add index increment * * #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: Put buffers in OpenSL playback queue only if filled * #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 * #14371: Update settings for playback buffers * #14371: Remove update OpenSL mixer settings * #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 * Revert "#14399: use libgnustl_static" * #14399: fix libsflphone and sflphoned build issue * #14399: use libgnustl_static * #14818: use one unique Android.mk at project's root * #14818: Keep HAVE_DBUS in config.h * #14399: remove sip_thread_client.cpp build * #14371: Start audio capture * #14371: Remove MainBuffer dump info from OpenSL * #14818: Use MainBuffer as a reference in OpenSL implementation * #14818: Get MainBuffer as a reference instead of a pointer * #14615: Fix compilation without iax * #14371: Fill buffers with zeros in OpenSL when no audio to be played * #14371: Instantiate audio layer, add linking flags for OpenSL in sflpnoned * * #14615: manager: voiplinks should cleanup their own accountmaps * * #14727: video: video receive thread should not call pjsip or touch the callmap * #14371: Implement audio playback and capture method * * #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 * #14631: Fix android build for manager * #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 * #14698: make pjsip log in android logcat sip level harcoded to 5 because calling __system_property_get() from NDK generates a segfault... * #14399: fix codec path and package name for android * #14399: fix program path and configdir * #14650: stub dbus in manager class * #14598: specify APP_NAME log tag in daemon libraries * #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 * #14617: Create sample example of JNIexport and shared lib * #14598: Modify logger class to print in android logcat * #14617: make libsflphone a shared lib to export it to Android JNI * #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 * #14399: fix lib deps * [#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 * #14607: Renamed codecs on Android to libcodec_ * * #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: Add android's config path * #14569: Refactor configuration file existence test and exception handling * #14379: Fix undefined AudioLayer::PCMType * [ #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 * #14523: dependency cleanup * #14470: fix lib dependencies * #14470: fix lib dependencies * #14465: Link sflphoned with zlib * 14470: Cross-compile libsflphone and sflphoned for android * #13891: missing initTLS in TLS optional support * #14470: Refactor hooks build for android * #14470: Refactor history build for android * #14470: Refactor libaudio build for android * #14470: Refactor audiortp build for android * #14479: minor removal * #14470: Remove dependency on ccrtp for codecs * #14470: Refactor sound static library Android.mk * #14470: Refactor Android.mk for opensl layer * #14379: Remove alsa from preferences * #14470: Cross-compile sflphoned for android * #14480: Add Android.mk to hooks subfolder * #14479: Cross-compile daemon/config for android * #14476: Add Android.mk in history subfolder * #14464: Cross-compile audio subfolder * #14463: Cross-compile audiortp for android * #14371: Add speex and dbus-c++ path to compile rtp layer * #14371: Add aduiortp mk file * #14451: add daemon/sip Android.mk * #14343: fix pjproject path in daemon/dbus * #14371: Add audio codec factory to Android mk file * #14371: Add android mk file for ulaw and alaw * #14399: add config.h for android * #14371: Add libsound android mk file * #14371: rename libaudio module as libopensl * #14371: Derive opensl layer from audiolayer * #14371: Add mk files for android * #14371: audiolayer tested externally for capture and playback * #14343: change include path of pjproject, fix CPPFLAGS * #14343: fix daemon/dbus include paths * #14371: Refactor buffers management in opensl audio layer * #14343: add Android.mk to sflphone/daemon/dbus * #14379: Make alsa an optional feature at compile time * #14371: Add basic opensl layer structure in source repository Checksums-Sha1: 33dccfece4da68d19f4c440c0a322890b802a732 8297414 sflphone-common-video_1.2.3-rc20130828~ppa1~saucy_i386.deb Checksums-Sha256: 2dc32021f32dce84f77f33cebc477856de32e14be39cf1501fe110a45748b9a4 8297414 sflphone-common-video_1.2.3-rc20130828~ppa1~saucy_i386.deb Files: 7675f2ff7ed07f96b39fbcbc4d4dc8a4 8297414 gnome optional sflphone-common-video_1.2.3-rc20130828~ppa1~saucy_i386.deb