Age | Commit message (Collapse) | Author |
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- Changed default value of PJMEDIA_HAS_VIDEO to disabled.
- Fixed code and build setting on Symbian for build correctness, it builds fine now.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3671 74dad513-b988-da41-8d7b-12977e46ad98
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specific documentation
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3669 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3668 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3667 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed compile warnings on vs2005
- Fixed compile error when PJMEDIA_HAS_VIDEO set to 0 on vs2005
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3666 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3665 74dad513-b988-da41-8d7b-12977e46ad98
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"2.0-pre-alpha-svn".
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3664 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3553 74dad513-b988-da41-8d7b-12977e46ad98
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Teluu copyright text).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3550 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3548 74dad513-b988-da41-8d7b-12977e46ad98
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using b=RS:0 and b=RR:0
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3547 74dad513-b988-da41-8d7b-12977e46ad98
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without format list.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3541 74dad513-b988-da41-8d7b-12977e46ad98
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- Updated pj_register_strerror() to just return PJ_SUCCESS when the same range
and handler is being re-registered.
- Removed the usage of static flag of error string handler registration in some
modules, which prevent the re-registration of the handler, e.g: in restarting
pjsua, as such flags never got reseted.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3455 74dad513-b988-da41-8d7b-12977e46ad98
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* Setting audio session category is now during stream creation instead of in the factory initialization.
* Reset the audio session category after an interruption.
* By default, audio route change property listener is disabled as it is no longer required.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3454 74dad513-b988-da41-8d7b-12977e46ad98
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* uri_test:
Fixes a divide by zero error when the benchmark is run on a really fast machine.
* presence:
Fixes a compiler warning about potential referencing of an uninitialized variable.
* echo_speex:
Allow for compilation when PJMEDIA_HAS_SPEEX_AEC is not defined.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3443 74dad513-b988-da41-8d7b-12977e46ad98
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from the jitter buffer.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3442 74dad513-b988-da41-8d7b-12977e46ad98
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* Update audio device's list after refreshing the device.
* Fixed WMME macro.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3440 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3438 74dad513-b988-da41-8d7b-12977e46ad98
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"Warning" for non-fatal errors
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3422 74dad513-b988-da41-8d7b-12977e46ad98
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"ippsr.lib" in auto link when IPP major version is 6 or below.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3417 74dad513-b988-da41-8d7b-12977e46ad98
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- Modified G722 frame time to 10ms (was 20ms) and frame per packet to 2 (was 1).
- Updated the detection mechanism of remote G722 frame-length in the stream to be flexible to any G722 frame length setting (was assumed to be always 20ms).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3416 74dad513-b988-da41-8d7b-12977e46ad98
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problematic G722 payloads (e.g: sized less than 160 bytes) and remote clock-rate/timestamp-span detection is active (PJMEDIA_HANDLE_G722_MPEG_BUG is set). Thanks Erik Waling for the patch.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3414 74dad513-b988-da41-8d7b-12977e46ad98
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invalid since AudioSessionInitialize can only be called once.
Re #1175: cleaning up interruption and audio route handling in coreaudio for iOS. In the case of interruption, there is no need to reinstantiate the audio unit (a simple restart will do), while for audio route change, nothing needs to be done.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3413 74dad513-b988-da41-8d7b-12977e46ad98
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RemoteIO. To use VPIO, application needs to specify a nonzero value for the PJMEDIA_AUD_DEV_CAP_EC capability.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3411 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3410 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3407 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3406 74dad513-b988-da41-8d7b-12977e46ad98
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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
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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
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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
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Fixed AudioSession services error handling in factory initialization.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3400 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3398 74dad513-b988-da41-8d7b-12977e46ad98
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- 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
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(thanks Johan Lantz for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3387 74dad513-b988-da41-8d7b-12977e46ad98
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an interruption.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3386 74dad513-b988-da41-8d7b-12977e46ad98
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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
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- 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
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pointer for the stream info source.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3379 74dad513-b988-da41-8d7b-12977e46ad98
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- 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
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3362 74dad513-b988-da41-8d7b-12977e46ad98
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offer (thanks Marcus Froeschl for the suggestion))
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3360 74dad513-b988-da41-8d7b-12977e46ad98
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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
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should include pjmedia/codec.h
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3358 74dad513-b988-da41-8d7b-12977e46ad98
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- 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
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- 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
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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
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coreaudio_dev
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3335 74dad513-b988-da41-8d7b-12977e46ad98
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overview section in pjmedia documentation
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3327 74dad513-b988-da41-8d7b-12977e46ad98
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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
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documentation when WWWDIR is specified
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3319 74dad513-b988-da41-8d7b-12977e46ad98
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