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- updated missing doxygen documentation from various PJMEDIA-CODEC headers
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2875 74dad513-b988-da41-8d7b-12977e46ad98
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handling burst of lost packets
WSOLA improvements:
- Introduce fade-out and fade-in effect
- Limit the number of continuous synthetic samples (only take effect when fading is used)
- Export many settings as macros:
- PJMEDIA_WSOLA_DELAY_MSEC (was HANNING_PTIME)
- PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC (was TEMPLATE_PTIME)
- PJMEDIA_WSOLA_MAX_EXPAND_MSEC
PLC:
- added compile time macro PJMEDIA_WSOLA_PLC_NO_FADING to disable fading (default enabled)
Stream:
- fixed bug when stream is not PLC-ing subsequent packet loss (only the first)
- also add maximum PLC limit just as precaution if PLC doesn't limit number of synthetic frames
- unrelated: fixed warning about unused send_keep_alive() function
Delaybuf:
- modified to NOT use fading in WSOLA since we don't expect it to generate many continuous synthetic frames
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2850 74dad513-b988-da41-8d7b-12977e46ad98
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receive the reflected packets
- This ticket allows the same loop media transport instance to be attached to more than one streams, and allow application to control which stream(s) receives the reflected packets.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2845 74dad513-b988-da41-8d7b-12977e46ad98
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stream/session
- added pjmedia_session_get_stream_stat_jbuf() and pjmedia_session_get_stream_stat_jbuf()
- fixed const correctness in pjmedia_jbuf_get_state(), jb_framelist_size(), and pj_math_stat_get_stddev(),
- modify the jitter buffer statistic log message printed by stream (it contains newlines)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2844 74dad513-b988-da41-8d7b-12977e46ad98
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- Added default ilbc mode into codec passthrough setting.
- Added iLBC mode 'negotiation' in iLBC codec_open().
- Updated stream_create() to prioritize codec_open(), that may update the codec params, over stream initializations involving codec params.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2834 74dad513-b988-da41-8d7b-12977e46ad98
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- Added support for Nokia VAS 2.0.
- Fixed wrong value assigned to last downstream state var in downstream callback.
- Minor fix in config_site_sample.h related to VAS Direct setting.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2833 74dad513-b988-da41-8d7b-12977e46ad98
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- Added a new API pjmedia_codec_passthrough_init2().
- Updated the initialization steps of passthrough codec in pjsua_media.c, to configure the codecs (of passthrough codec) to be enabled based on audio device extended/encoded formats.
- Minor update: added passthrough.h into pjmedia_codec.vcproj.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2825 74dad513-b988-da41-8d7b-12977e46ad98
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- Added new audio device VAS for Symbian platform.
- Updated symsndtest to use the latest audio device framework.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2821 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2763 74dad513-b988-da41-8d7b-12977e46ad98
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to media stream.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2759 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2755 74dad513-b988-da41-8d7b-12977e46ad98
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- Added pjmedia synchronizer port.
- Updated affected components, i.e: sound port, AEC, conference bridge.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2747 74dad513-b988-da41-8d7b-12977e46ad98
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- Added calls to delay buf destructor in conference.c and echo_common.c.
- Moved mutex creation to the end of pjmedia_delay_buf_create().
- Deprecated pjmedia_conf_add_passive_port().
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2728 74dad513-b988-da41-8d7b-12977e46ad98
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config.h: removed phrase 'under development', G722.1 remains disabled by default.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2680 74dad513-b988-da41-8d7b-12977e46ad98
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value (default is 1) to make it compatible with some other app
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2620 74dad513-b988-da41-8d7b-12977e46ad98
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- Added build config for GNU autoconf & make.
- Fixed some G.722.1 codes for linux & mingw32 targets, e.g: types
defs, collision function name 'round'.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2601 74dad513-b988-da41-8d7b-12977e46ad98
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initialization.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2597 74dad513-b988-da41-8d7b-12977e46ad98
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- Updated loop condition in put_frame() to avoid possibility of infinite loop.
- Added JB capabilities to handle sequence restart & jump.
- Updated jitter calculation, e.g: reset max_hist_level after updating prefetch, avoid updating prefetch when burst level is exceeding max_burst.
- Updated shrinking method to be less agressive (only shrink JB when JB size is twice larger than burst level).
- Updated the way JB switching status from 'initializing' to 'processing' by waiting for some OP switch cycles.
- Few simplifications in framelist process, e.g: replacing fields 'empty' & 'tail' with 'size'.
- Minor updates: comments, shortened framelist field names, added some JB states for reporting/monitoring purpose.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2578 74dad513-b988-da41-8d7b-12977e46ad98
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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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2521 74dad513-b988-da41-8d7b-12977e46ad98
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aps-direct branch to trunk.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2506 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed Symbian APS G.711 frame size variation issue.
- Fixed APS implementation to regard 'samples_per_frame' setting.
- Added APIs for u-law/a-law <-> PCM bulk conversions.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2416 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2386 74dad513-b988-da41-8d7b-12977e46ad98
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MAXUINT16+1 (feature is disabled).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2368 74dad513-b988-da41-8d7b-12977e46ad98
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codecs to enable octet-align from FMTP settings.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2363 74dad513-b988-da41-8d7b-12977e46ad98
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- Configurable setting to enable/disable AMR bitstream reordering (sensitivity order to/from encoder bits order).
- Updated AMR codec to regard in-band Change Mode Request from remote encoder.
- Updated AMR settings (octet-align, etc) to be configured upon codec opening, instead of hardcoded in the encode, decode, parse.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2359 74dad513-b988-da41-8d7b-12977e46ad98
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codec helper.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2358 74dad513-b988-da41-8d7b-12977e46ad98
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to disable, otherwise to enable).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2317 74dad513-b988-da41-8d7b-12977e46ad98
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report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2313 74dad513-b988-da41-8d7b-12977e46ad98
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accuracy, and updated tonegen.c with the results from ARM9 tests
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2294 74dad513-b988-da41-8d7b-12977e46ad98
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- Changed rem_rtp/rtcp_addr to src_rtp/rtcp_addr in media transport info
- Updated behaviour of media transport get info, when the transport hasn't receive any packet src_rtp/rtcp_addr will not be set.
- Fixed bug in pjsua_call_dump that rem_addr was unset.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2293 74dad513-b988-da41-8d7b-12977e46ad98
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- Deprecate the automatic selection of algorithm
- Introduced various constants for tonegen backends
- Allow user to specify/override the algorithm by setting
- Fix the floating-point approximation backend
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2292 74dad513-b988-da41-8d7b-12977e46ad98
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- Added new fields in media transport info: remote address originating RTP & RTCP.
- Updated transport UDP & ICE to fill above fields in getting transport info.
- Updated pjsua call dump, instead of showing remote RTP address from SDP, it will show address of RTP originator.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2291 74dad513-b988-da41-8d7b-12977e46ad98
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patch).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2290 74dad513-b988-da41-8d7b-12977e46ad98
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the tone generator, added PJMEDIA_TONEGEN_VOLUME setting to control the default amplitude, and increase default tone volume by about 50%
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2284 74dad513-b988-da41-8d7b-12977e46ad98
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used in the tone generator, and added fade-in and fade out options
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2281 74dad513-b988-da41-8d7b-12977e46ad98
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transmission to source address if remote doesn't use ICE.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2276 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2275 74dad513-b988-da41-8d7b-12977e46ad98
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bit A-law/U-law.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2270 74dad513-b988-da41-8d7b-12977e46ad98
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how SRTP media transport works
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2262 74dad513-b988-da41-8d7b-12977e46ad98
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- Introduced new API pjmedia_rtp_session_init2() to enable intializing RTP session with non-default initial settings
- Updated stream so it can be created with non-default initial RTP settings.
- Updated pjsua-lib to make sure RTP timestamp and sequence contigue when stream session is restarted.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2241 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2225 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2222 74dad513-b988-da41-8d7b-12977e46ad98
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- Added codec AMR-WB
- Updated AMR & AMRWB to utilize quality flag in the AMR payload header
- Updated callback interface (frm_attr_cb() -> predecode_cb())
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2219 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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setting the threshold too high thus cutting audio signal (e.g. when streaming long continuous signal)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2215 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2214 74dad513-b988-da41-8d7b-12977e46ad98
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