[asterisk-dev] [Code Review] Media Project Phase 2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, Custom format creation in codecs.conf

[David Vossel]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1104/
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(Updated 2011-02-16 17:41:27.978959)


Review request for Asterisk Developers.


Changes
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This update includes everything I've done so far from the hd_conferencing_ftw branch.

Additional Changes:
1. ConfBridge now dynamically jumps to the best possible sample rate.  This allows for conferences to take advantage of HD audio (Which sounds awesome)
2. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
3. codec_resample now uses its own code rather than depending on libresample.


Summary
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-Functional Changes
Dynamic global format list build by codecs defined in codecs.conf
SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
Negotiation of SILK attributes in chan_sip.
SPEEX 32khz with translation
SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c
Various changes to RTP code required to properly handle the dynamic format list and formats with attributes.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

-MIS
There were several changes that I opted not to do during phase 1 that involved the proper use of formats with attributes.  I felt that the phase 1 patch was too complex to attempt these changes so I have included them here.  The primary change I held off doing the first time around was in the rtp_engine.c Payload and MIME type lists.  These lists were statically defined which made it impossible to set formats with attributes in them.  I have gone back and revised these lists to allow the flexibility required for the new architecture.


Diffs (updated)
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  /trunk/CHANGES 308199 
  /trunk/apps/app_chanspy.c 308199 
  /trunk/apps/app_jack.c 308199 
  /trunk/apps/app_mixmonitor.c 308199 
  /trunk/bridges/bridge_multiplexed.c 308199 
  /trunk/bridges/bridge_softmix.c 308199 
  /trunk/channels/chan_gtalk.c 308199 
  /trunk/channels/chan_iax2.c 308199 
  /trunk/channels/chan_jingle.c 308199 
  /trunk/channels/chan_sip.c 308199 
  /trunk/channels/chan_skinny.c 308199 
  /trunk/channels/iax2.h 308199 
  /trunk/codecs/Makefile 308199 
  /trunk/codecs/codec_resample.c 308199 
  /trunk/codecs/codec_speex.c 308199 
  /trunk/codecs/speex/arch.h PRE-CREATION 
  /trunk/codecs/speex/fixed_generic.h PRE-CREATION 
  /trunk/codecs/speex/resample.c PRE-CREATION 
  /trunk/codecs/speex/resample_sse.h PRE-CREATION 
  /trunk/codecs/speex/speex_resampler.h PRE-CREATION 
  /trunk/codecs/speex/stack_alloc.h PRE-CREATION 
  /trunk/configs/codecs.conf.sample 308199 
  /trunk/formats/format_attr_silk.c PRE-CREATION 
  /trunk/funcs/func_pitchshift.c 308199 
  /trunk/funcs/func_speex.c 308199 
  /trunk/funcs/func_volume.c 308199 
  /trunk/include/asterisk/audiohook.h 308199 
  /trunk/include/asterisk/format.h 308199 
  /trunk/include/asterisk/format_cap.h 308199 
  /trunk/include/asterisk/frame.h 308199 
  /trunk/include/asterisk/rtp_engine.h 308199 
  /trunk/include/asterisk/silk.h PRE-CREATION 
  /trunk/include/asterisk/slinfactory.h 308199 
  /trunk/include/asterisk/time.h 308199 
  /trunk/include/asterisk/translate.h 308199 
  /trunk/main/asterisk.c 308199 
  /trunk/main/audiohook.c 308199 
  /trunk/main/channel.c 308199 
  /trunk/main/data.c 308199 
  /trunk/main/format.c 308199 
  /trunk/main/format_cap.c 308199 
  /trunk/main/format_pref.c 308199 
  /trunk/main/frame.c 308199 
  /trunk/main/rtp_engine.c 308199 
  /trunk/main/slinfactory.c 308199 
  /trunk/main/translate.c 308199 
  /trunk/res/res_mutestream.c 308199 
  /trunk/res/res_rtp_asterisk.c 308199 

Diff: https://reviewboard.asterisk.org/r/1104/diff


Testing
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I used the same testing I did for phase1 with phase2.  This includes load testing with various codec negotiation scenarios.  Speex 32 and its translators were verified using Asterisk back to back.  I verified SILK using a codec translator module I'm working with.  The new slinear resample code was exercised with both Speex32 and SILK 12khz/24khz translation.


Thanks,

David

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