[asterisk-dev] [Code Review] Asterisk media architecture conversion - no more format bitfields

Terry Wilson reviewboard at asterisk.org
Wed Feb 2 16:37:08 CST 2011 

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I think this is the last bit of review!


/trunk/include/asterisk/rtp_engine.h
<https://reviewboard.asterisk.org/r/1083/#comment6539>

    s/F/f/



/trunk/include/asterisk/rtp_engine.h
<https://reviewboard.asterisk.org/r/1083/#comment6540>

    s/asterisk/Asterisk/



/trunk/include/asterisk/translate.h
<https://reviewboard.asterisk.org/r/1083/#comment6541>

    Not yours, but C++-style comment



/trunk/include/asterisk/translate.h
<https://reviewboard.asterisk.org/r/1083/#comment6542>

    



/trunk/include/asterisk/translate.h
<https://reviewboard.asterisk.org/r/1083/#comment6543>

    What is the reasoning for the specific numbers chosen? Is the number order all that is important or is arithmetic done on them at some point?



/trunk/main/app.c
<https://reviewboard.asterisk.org/r/1083/#comment6544>

    ast_format_clear



/trunk/main/channel.c
<https://reviewboard.asterisk.org/r/1083/#comment6545>

    I may be misunderstanding, but it seems like we should send everything capable of being translated.
    
    What if we determine the best codec to be a codec that isn't available in the destination channel due to a peer config? Say the requestor supports ulaw and g729 and we determine that ulaw is the best. We then send the call to a SIP device that is configured in sip.conf to only allow g729. In this case, wouldn't we do an unnecessary transcode since jointcaps for the peer/pvt in chan_sip would be empty when it might not have been if we had sent everything?


- Terry


On 2011-01-24 15:25:45, David Vossel wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1083/
> -----------------------------------------------------------
> 
> (Updated 2011-01-24 15:25:45)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> This patch is the foundation of an entire new way of looking at media in Asterisk.  The code present in this review is everything required to complete phase1 of my Media Architecture proposal.
> 
> For more information about this project visit the link below.
> https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
> 
> 
> Diffs
> -----
> 
>   /trunk/addons/chan_mobile.c 303557 
>   /trunk/addons/chan_ooh323.h 303557 
>   /trunk/addons/chan_ooh323.c 303557 
>   /trunk/addons/format_mp3.c 303557 
>   /trunk/addons/ooh323cDriver.h 303557 
>   /trunk/addons/ooh323cDriver.c 303557 
>   /trunk/apps/app_alarmreceiver.c 303557 
>   /trunk/apps/app_amd.c 303557 
>   /trunk/apps/app_chanspy.c 303557 
>   /trunk/apps/app_confbridge.c 303557 
>   /trunk/apps/app_dahdibarge.c 303557 
>   /trunk/apps/app_dictate.c 303557 
>   /trunk/apps/app_dumpchan.c 303557 
>   /trunk/apps/app_echo.c 303557 
>   /trunk/apps/app_fax.c 303557 
>   /trunk/apps/app_festival.c 303557 
>   /trunk/apps/app_followme.c 303557 
>   /trunk/apps/app_ices.c 303557 
>   /trunk/apps/app_jack.c 303557 
>   /trunk/apps/app_meetme.c 303557 
>   /trunk/apps/app_milliwatt.c 303557 
>   /trunk/apps/app_mixmonitor.c 303557 
>   /trunk/apps/app_mp3.c 303557 
>   /trunk/apps/app_nbscat.c 303557 
>   /trunk/apps/app_originate.c 303557 
>   /trunk/apps/app_parkandannounce.c 303557 
>   /trunk/apps/app_record.c 303557 
>   /trunk/apps/app_rpt.c 303557 
>   /trunk/apps/app_sms.c 303557 
>   /trunk/apps/app_speech_utils.c 303557 
>   /trunk/apps/app_talkdetect.c 303557 
>   /trunk/apps/app_test.c 303557 
>   /trunk/apps/app_voicemail.c 303557 
>   /trunk/apps/app_waitforsilence.c 303557 
>   /trunk/bridges/bridge_multiplexed.c 303557 
>   /trunk/bridges/bridge_simple.c 303557 
>   /trunk/bridges/bridge_softmix.c 303557 
>   /trunk/channels/chan_agent.c 303557 
>   /trunk/channels/chan_alsa.c 303557 
>   /trunk/channels/chan_bridge.c 303557 
>   /trunk/channels/chan_console.c 303557 
>   /trunk/channels/chan_dahdi.c 303557 
>   /trunk/channels/chan_gtalk.c 303557 
>   /trunk/channels/chan_h323.c 303557 
>   /trunk/channels/chan_iax2.c 303557 
>   /trunk/channels/chan_jingle.c 303557 
>   /trunk/channels/chan_local.c 303557 
>   /trunk/channels/chan_mgcp.c 303557 
>   /trunk/channels/chan_misdn.c 303557 
>   /trunk/channels/chan_multicast_rtp.c 303557 
>   /trunk/channels/chan_nbs.c 303557 
>   /trunk/channels/chan_oss.c 303557 
>   /trunk/channels/chan_phone.c 303557 
>   /trunk/channels/chan_sip.c 303557 
>   /trunk/channels/chan_skinny.c 303557 
>   /trunk/channels/chan_unistim.c 303557 
>   /trunk/channels/chan_usbradio.c 303557 
>   /trunk/channels/chan_vpb.cc 303557 
>   /trunk/channels/h323/ast_h323.cxx 303557 
>   /trunk/channels/h323/chan_h323.h 303557 
>   /trunk/channels/iax2-parser.h 303557 
>   /trunk/channels/iax2-parser.c 303557 
>   /trunk/channels/iax2-provision.c 303557 
>   /trunk/channels/iax2.h 303557 
>   /trunk/channels/sip/include/globals.h 303557 
>   /trunk/channels/sip/include/sip.h 303557 
>   /trunk/codecs/codec_a_mu.c 303557 
>   /trunk/codecs/codec_adpcm.c 303557 
>   /trunk/codecs/codec_alaw.c 303557 
>   /trunk/codecs/codec_dahdi.c 303557 
>   /trunk/codecs/codec_g722.c 303557 
>   /trunk/codecs/codec_g726.c 303557 
>   /trunk/codecs/codec_gsm.c 303557 
>   /trunk/codecs/codec_ilbc.c 303557 
>   /trunk/codecs/codec_lpc10.c 303557 
>   /trunk/codecs/codec_resample.c 303557 
>   /trunk/codecs/codec_speex.c 303557 
>   /trunk/codecs/codec_ulaw.c 303557 
>   /trunk/codecs/ex_adpcm.h 303557 
>   /trunk/codecs/ex_alaw.h 303557 
>   /trunk/codecs/ex_g722.h 303557 
>   /trunk/codecs/ex_g726.h 303557 
>   /trunk/codecs/ex_gsm.h 303557 
>   /trunk/codecs/ex_lpc10.h 303557 
>   /trunk/codecs/ex_speex.h 303557 
>   /trunk/codecs/ex_ulaw.h 303557 
>   /trunk/formats/format_g719.c 303557 
>   /trunk/formats/format_g723.c 303557 
>   /trunk/formats/format_g726.c 303557 
>   /trunk/formats/format_g729.c 303557 
>   /trunk/formats/format_gsm.c 303557 
>   /trunk/formats/format_h263.c 303557 
>   /trunk/formats/format_h264.c 303557 
>   /trunk/formats/format_ilbc.c 303557 
>   /trunk/formats/format_jpeg.c 303557 
>   /trunk/formats/format_ogg_vorbis.c 303557 
>   /trunk/formats/format_pcm.c 303557 
>   /trunk/formats/format_siren14.c 303557 
>   /trunk/formats/format_siren7.c 303557 
>   /trunk/formats/format_sln.c 303557 
>   /trunk/formats/format_sln16.c 303557 
>   /trunk/formats/format_vox.c 303557 
>   /trunk/formats/format_wav.c 303557 
>   /trunk/formats/format_wav_gsm.c 303557 
>   /trunk/funcs/func_channel.c 303557 
>   /trunk/funcs/func_frame_trace.c 303557 
>   /trunk/funcs/func_pitchshift.c 303557 
>   /trunk/include/asterisk/abstract_jb.h 303557 
>   /trunk/include/asterisk/astobj2.h 303557 
>   /trunk/include/asterisk/audiohook.h 303557 
>   /trunk/include/asterisk/bridging.h 303557 
>   /trunk/include/asterisk/bridging_technology.h 303557 
>   /trunk/include/asterisk/callerid.h 303557 
>   /trunk/include/asterisk/channel.h 303557 
>   /trunk/include/asterisk/data.h 303557 
>   /trunk/include/asterisk/file.h 303557 
>   /trunk/include/asterisk/format.h PRE-CREATION 
>   /trunk/include/asterisk/format_cap.h PRE-CREATION 
>   /trunk/include/asterisk/format_pref.h PRE-CREATION 
>   /trunk/include/asterisk/frame.h 303557 
>   /trunk/include/asterisk/frame_defs.h 303557 
>   /trunk/include/asterisk/image.h 303557 
>   /trunk/include/asterisk/mod_format.h 303557 
>   /trunk/include/asterisk/pbx.h 303557 
>   /trunk/include/asterisk/rtp_engine.h 303557 
>   /trunk/include/asterisk/slin.h 303557 
>   /trunk/include/asterisk/slinfactory.h 303557 
>   /trunk/include/asterisk/speech.h 303557 
>   /trunk/include/asterisk/translate.h 303557 
>   /trunk/main/abstract_jb.c 303557 
>   /trunk/main/app.c 303557 
>   /trunk/main/asterisk.c 303557 
>   /trunk/main/astobj2.c 303557 
>   /trunk/main/audiohook.c 303557 
>   /trunk/main/bridging.c 303557 
>   /trunk/main/callerid.c 303557 
>   /trunk/main/ccss.c 303557 
>   /trunk/main/channel.c 303557 
>   /trunk/main/cli.c 303557 
>   /trunk/main/data.c 303557 
>   /trunk/main/dial.c 303557 
>   /trunk/main/dsp.c 303557 
>   /trunk/main/features.c 303557 
>   /trunk/main/file.c 303557 
>   /trunk/main/format.c PRE-CREATION 
>   /trunk/main/format_cap.c PRE-CREATION 
>   /trunk/main/format_pref.c PRE-CREATION 
>   /trunk/main/frame.c 303557 
>   /trunk/main/image.c 303557 
>   /trunk/main/indications.c 303557 
>   /trunk/main/manager.c 303557 
>   /trunk/main/pbx.c 303557 
>   /trunk/main/rtp_engine.c 303557 
>   /trunk/main/slinfactory.c 303557 
>   /trunk/main/translate.c 303557 
>   /trunk/main/udptl.c 303557 
>   /trunk/pbx/pbx_spool.c 303557 
>   /trunk/res/res_adsi.c 303557 
>   /trunk/res/res_agi.c 303557 
>   /trunk/res/res_calendar.c 303557 
>   /trunk/res/res_clioriginate.c 303557 
>   /trunk/res/res_fax.c 303557 
>   /trunk/res/res_fax_spandsp.c 303557 
>   /trunk/res/res_musiconhold.c 303557 
>   /trunk/res/res_rtp_asterisk.c 303557 
>   /trunk/res/res_rtp_multicast.c 303557 
>   /trunk/res/res_speech.c 303557 
>   /trunk/tests/test_format_api.c PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/1083/diff
> 
> 
> Testing
> -------
> 
> Below are the major areas I tested during development.  I will continue testing as this patch is being reviewed.
> -Local Channel + IAX2 channel load testing
> -SIP Calls with and without video
> -IAX2 Calls
> -AudioHooks and apps using audiohooks
> -Masquerades
> -DTMF Attended Transfers
> -SIP Transfers
> -Gtalk
> 
> 
> Thanks,
> 
> David
> 
>

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