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2011-01-19Fixes #597: Error reading WAV files when there are extra chunks after DATA ↵Liong Sauw Ming
chunk. Wav player will now just play data chunks in wav files and ignore others. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3405 74dad513-b988-da41-8d7b-12977e46ad98
2011-01-13Fix #1195: assertion during thread registration in coreaudio and portaudio's ↵Liong Sauw Ming
input/output callbacks on Mac OS X and #1196: using system's default audio input/output device instead of first available device. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3404 74dad513-b988-da41-8d7b-12977e46ad98
2011-01-04As compared to iOS SDK 3.x, in iOS SDK 4.x, __IPHONE_OS_VERSION_MIN_REQUIRED ↵Liong Sauw Ming
is defined only when -miphoneos-version-min is used (see the SDK's AvailabilityInternal.h). This causes coreaudio_dev to be compiled using RemoteIO (instead of VPIO) and without Bluetooth support. This revision fixes #1194. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3403 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-19Re #1174:Liong Sauw Ming
Fixed AudioSession services error handling in factory initialization. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3400 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-17Fixed #1174, #1191, #1192Liong Sauw Ming
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3398 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-14Fix #1188:Nanang Izzuddin
- Fixed jitter buffer progressive discard by returning 'missing' frame after discarded frame(s) so the PLC will be invoked to align the audio signal. - Modified conditions in jitter buffer test data for this fix. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3396 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-06Fixed #1160: Option to bypass RTP payload type verification in the stream ↵Benny Prijono
(thanks Johan Lantz for the suggestion) git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3387 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-05Misc (re #1151): Fixed crash in coreaudio_dev when there is no call during ↵Liong Sauw Ming
an interruption. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3386 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-02Re #1165: fixed SRTP transport to not update the SRTP activation flag, i.e: ↵Nanang Izzuddin
bypass_srtp, in transport_encode_sdp(), as it shouldn't change the media session states. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3384 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-02Fix #1171, fixed issues in pjmedia_codec_mgr_set_default_param():Nanang Izzuddin
- updated the releasing of the old pool to be done after the new codec param is copied - fixed the double dec_fmtp copy loop, one of them should be enc_fmtp copy loop instead git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3380 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-02Misc (re #1151): fixed bug in pjmedia_session_enum_streams(), used wrong ↵Nanang Izzuddin
pointer for the stream info source. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3379 74dad513-b988-da41-8d7b-12977e46ad98
2010-12-01Fix #1165:Nanang Izzuddin
- Fixed pjsua_media_channel_create_sdp() to re-calculate audio index of the remote offer, instead of using existing audio index calculated by pjsua_media_channel_init(), as for subsequent SDP offer/answer, pjsua_media_channel_init() may not be called. - Fixed SRTP transport to be able to switch SRTP status from active to inactive/by-passed and vice versa. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3376 74dad513-b988-da41-8d7b-12977e46ad98
2010-11-03Misc (re #1151): fixed the compile time switch to use iLBC codec on iPhoneBenny Prijono
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3362 74dad513-b988-da41-8d7b-12977e46ad98
2010-11-03Fixed #1154 (Run-time option to disable telephone-event in outgoing SDP ↵Benny Prijono
offer (thanks Marcus Froeschl for the suggestion)) git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3360 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-29Misc (re #1151): fixed jitter buffer state 'jb_max_burst' initialization, ↵Nanang Izzuddin
MAX_BURST_MSEC should be converted to number-of-frame unit (thanks Zhefeng Du for the fix). git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3359 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-28Fix #1153: Fixed missing header dependency in pjmedia-codec/types.h, it ↵Nanang Izzuddin
should include pjmedia/codec.h git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3358 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-18Re #1140, cleaned up the mutex usages in SRTP transport:Nanang Izzuddin
- using mutex in accessing application callback pointers - releasing mutex before calling application callbacks to avoid deadlock - refine the synchronization of backend/libsrtp states git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3348 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-16Fixed #1148:Nanang Izzuddin
- Added check in processing answer, if media offer port is zero, just skip negotiation process. - Added SIPp test scenario. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3347 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-14Misc (re #1134): changed dynamic payload type starting value for internal ↵Benny Prijono
codecs to 96 (from 101) as we are running out of constants. Also added more comments to clarify the restriction on the value, i.e. it must be less than 128 (thanks Robbie Hanson for the suggestion) git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3345 74dad513-b988-da41-8d7b-12977e46ad98
2010-10-10Misc (re #1134): Fixed memory leak after getting audio route capability in ↵Liong Sauw Ming
coreaudio_dev git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3335 74dad513-b988-da41-8d7b-12977e46ad98
2010-09-30Re #1134 (misc fixes): fixed various doxygen warnings, as well as added ↵Benny Prijono
overview section in pjmedia documentation git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3327 74dad513-b988-da41-8d7b-12977e46ad98
2010-09-28Fix #1135: added RTP padding checks and RTP payload length adjustment ↵Nanang Izzuddin
according to padding length. Payload padding in outgoing RTP investigation results: - the RTP does not specify RTP payload alignment. - most codecs also do not specify RTP payload alignment, usually only octet-alignment is specified and this seems to be done. - SRTP, RFC3711 states: - None of the pre-defined SRTP encryption transforms uses any padding; for these, the RTP and SRTP payload sizes match exactly. - Message authentication codes define their own padding. - Encryption transforms that use padding are vulnerable to subtle attacks, especially when message authentication is not used. So, currently payload padding in outgoing RTP is not necessary. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3325 74dad513-b988-da41-8d7b-12977e46ad98
2010-09-23Misc (re #1134): fixed error in Makefile target for generating doxygen ↵Benny Prijono
documentation when WWWDIR is specified git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3319 74dad513-b988-da41-8d7b-12977e46ad98
2010-09-22Close #1133: added processing to remote SDP attribute "ptime".Nanang Izzuddin
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3315 74dad513-b988-da41-8d7b-12977e46ad98
2010-09-20Closed #1129:Nanang Izzuddin
- Added run-time configuration for activating/deactivating stream keep-alive (non-codec-VAD mechanism), also added this config to account settings. - Fixed bug wrong session info pointer "si" in pjsua_media_channel_update() when call audio index is not zero. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3313 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-24Fix #1114:Nanang Izzuddin
- Fixed bytes_per_frame calculation in stream port. - Fixed sample streamutil.c to use codec info/param for codec bandwidth calculation (was using bytes_per_frame info of stream port). - Doc fix for bytes_per_frame field in pjmedia_port_info. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3292 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-18Fix #1112: fixed IPP G.729 codec to update the application regarding VAD ↵Nanang Izzuddin
status, disabled/off, after receiving 'fmtp:18 annexb=no' in the SDP. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3288 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-11Re #1106: Reverted back the IPP codec APIs ↵Nanang Izzuddin
pjmedia_codec_ipp_set/get_config() as currently to set PCM signal level adjustment can be done using the existing G722.1 API pjmedia_codec_g7221_set_pcm_shift(). git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3263 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-11Misc (re #1068): fixed various compilation warnings with Visual Studio 6 and ↵Benny Prijono
2005 git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3262 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-10Fix #1106:Nanang Izzuddin
- Added PCM signal adjustment in IPP G722.1 implementation. The default setting is configurable via (the existing compile-time config) PJMEDIA_G7221_DEFAULT_PCM_SHIFT. - Added new APIs to get and set IPP codecs settings: pjmedia_codec_ipp_set/get_config(). At run-time, the G722.1 PCM signal adjustment setting can be set using these functions. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3261 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-06Misc (re #1068): temporary fix address parsing in SDP connection info.Nanang Izzuddin
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3256 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-06Misc (re #1068): fix compile errors/warnings on Symbian S60 5th ed.Nanang Izzuddin
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3255 74dad513-b988-da41-8d7b-12977e46ad98
2010-08-05Re #1072: fixed buffer overflow bugs introduced by r3202.Nanang Izzuddin
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3250 74dad513-b988-da41-8d7b-12977e46ad98
2010-07-15Re #1103:Nanang Izzuddin
- Added (back) raw jitter statistics into RTCP statistics, with the new name "rx_raw_jitter". - Added IPDV statistics into RTCP statistics. - Added new compile-time settings to enable/disable raw jitter and IPDV statistics. - Updated call dump in pjsua-lib. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3239 74dad513-b988-da41-8d7b-12977e46ad98
2010-07-15Re #1103:Nanang Izzuddin
- Updated RTCP jitter statistics calculation (in receiving direction) to use "interarrival jitter" (was using "difference D") of RFC 3550. - Added APIs to reset RTCP statistics. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3237 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-26Fixed segfault Re #740Benny Prijono
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3226 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-26Fix #740:Nanang Izzuddin
- Fixed zeroed/unset RTP timestamp in RTCP sender report. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3224 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-24Fix #1075:Nanang Izzuddin
- Fixed SRTP transport to only (re)start the SRTP state when the SRTP crypto settings are updated. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3221 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-23Re #1094: fixed compilation error in previous fix in r3218Benny Prijono
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3220 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-23Fixed #1079 (Media transport should be kept alive during double-hold ↵Benny Prijono
scenario). Details: - now the stream will be destroyed but the media transport will be kept alive during doublehold scenario - small fix in SRTP to also negotiate crypto even when the media is marked as inactive, otherwise it's possible that an "optional" endpoint would create RTP/AVP offer and send it to "mandatory" endpoint, which would be rejected and cause the media port to be set to zero git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3219 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-23Fixed #1094: Transmit RTCP keep alive even when the stream is paused (thanks ↵Benny Prijono
Johan Lantz for the suggestion) git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3218 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-23Another unterminated SDP offer/answer negotiation fix (Re: #1045):Benny Prijono
- fixed unterminated negotiation if our media transport rejects incoming offer (e.g. due to mismatch SRTP transport) with 488. - to fix the above, modified the SDP negotiator (sdp_neg.[hc])'s pjmedia_sdp_neg_cancel_offer() to also be able to cancel in remote offer state - also fixed the bug introduced previous Session Timer fix (Re: #1047), which cause SDP negotiator's state to be cleared after failed UAC UPDATE transaction is terminated, which means UPDATE can only be sent 5 seconds after the last UPDATE if the last UPDATE failed. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3217 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-21Misc (re #1068): added new API pjmedia_master_port_wait() to poll master ↵Nanang Izzuddin
port clock when it is created with PJMEDIA_CLOCK_NO_ASYNC flag. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3214 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-15Misc (Re #1068): initialize rtpmap string with NULL as it generates ↵Benny Prijono
uninitialized memory ptr read under Valgrind. Thanks Jones Desougi for the patch. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3204 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-11Close #1072:Nanang Izzuddin
- Added API pjmedia_codec_g722_set_pcm_shift() to enable configurable level-adjusment setting. - Also added macro PJMEDIA_G722_DEFAULT_PCM_SHIFT (default value is 2) to accomplish 14-16 bit PCM conversion for G722 input/output. - Added a feature in G722 to stop level-adjusment/PCM-shifting when clipping occured, compile-time configurable via PJMEDIA_G722_STOP_PCM_SHIFT_ON_CLIPPING macro. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3202 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-11Fix #1093:Nanang Izzuddin
- Fixed memory leak of CVoIPFormatIntfc instances in S60 VAS. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3201 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-07Fix #1085:Nanang Izzuddin
- Added new codec G721, as alias for G726-32. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3199 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-04Re #668:Nanang Izzuddin
- Fixed process_answer() of SDP negotiation, when no common format in a media, instead of returning error, it should just deactivate the media (offer & answer) and continue negotiating next media. - Generalized the way of deactivating media: set port to 0 and remove all attributes. - Added new API pjmedia_sdp_media_clone_deactivate() to clone media and deactivate the newly cloned media. - Updated PJMEDIA SDP negotiation test. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3198 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-03Fix #1073:Nanang Izzuddin
- Fixed SDP negotiation in processing answer: when the answer has less media count, generate disabled-media to match the media count. - Added python test. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3195 74dad513-b988-da41-8d7b-12977e46ad98
2010-06-02Fix #1088:Nanang Izzuddin
- Fixed SDP negotiation to ignore disabled media (with port 0) in the answer. - Added a SIPp scenario for reproducing the issue. git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3192 74dad513-b988-da41-8d7b-12977e46ad98
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