Age | Commit message (Collapse) | Author |
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- Initial source of G.722.1/Annex C integration.
- Disabled some "odd" modes of L16 codec (11kHz & 22kHz mono & stereo) while releasing some payload types.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2563 74dad513-b988-da41-8d7b-12977e46ad98
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well
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2394 74dad513-b988-da41-8d7b-12977e46ad98
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- Added "dec_fmtp" and "enc_fmtp" fields to pjmedia_codec_param.setting.
- Codec factory puts its default parameters in "dec_fmtp" field.
- pjmedia_stream_info_from_sdp() puts the "fmtp" attribute in SDP to pjmedia_codec_param.
- Special treatment for fmtp "bitrate" parameter (of G722.1) during SDP negotiation
- Added maxptime field in stream_info.
- Replaced iLBC's fmtp "mode" implementation to use general fmtp mechanism.
- Added some test scripts for G722.1 bitrate negotiation.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2236 74dad513-b988-da41-8d7b-12977e46ad98
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- rearranged some codec properties, e.g: codec name, enable/disable, payload type
- fixed bug VAD setting on init USC codec
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2218 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2216 74dad513-b988-da41-8d7b-12977e46ad98
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copyright notice
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2039 74dad513-b988-da41-8d7b-12977e46ad98
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telephone-event (thanks Esbjorn Dominique)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@1070 74dad513-b988-da41-8d7b-12977e46ad98
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2007 in copyright notice in all sources
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@974 74dad513-b988-da41-8d7b-12977e46ad98
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- configurable default decoder mode (20 or 30),
- encoder mode follows the mode specified in SDP fmtp from
the remote's SDP,
- silence detector uses pjmedia's,
- PLC uses iLBC's PLC,
- perceptual enhancement (penh) is configurable via codec
param, as usual.
- iLBC mode is configurable in pjsua with --ilbc-mode option.
- Added packet lost simulation in pjmedia's UDP transport and
in pjsua (with --rx-drop-pct and --tx-drop-pct options).
- Increase default buffer count in DirectSound to 32 frames
to make it more resilient to CPU disruption.
- Specify and parse fmtp mode in SDP for codecs that need it.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@637 74dad513-b988-da41-8d7b-12977e46ad98
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-pedantic is used, also when g++ is used
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@622 74dad513-b988-da41-8d7b-12977e46ad98
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parsing and validation, (2) fixed bug in RTCP attribute generation in SDP, (3) configurable telephone-event payload type
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@571 74dad513-b988-da41-8d7b-12977e46ad98
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handling for case remote media is restarted
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@411 74dad513-b988-da41-8d7b-12977e46ad98
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specific codec during compilation
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@320 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@176 74dad513-b988-da41-8d7b-12977e46ad98
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