Hello Alexander, great work!! If you ask, it would be nice if transcoding support is there for GSM-EFR and both AMR Codes. If you need testing, I can support you.
Kind regards, André Am 24.11.2015 um 16:08 schrieb Alexander Traud: > Thanks to the codec/format changes which were introduced with Asterisk 13, > adding new trancoding modules is possible within one working day. Thanks to > format-attribute modules, the debugging of the SDP/fmtp-negotiation resides > in one source file. Therefore, I was able to port five formats to > Asterisk 13: <http://github.com/traud>. > > > Question #1: Level of Integration? > > Now, I plan to submit those modules into Asterisk. However, which level of > integration is desired for which format: pass-through only (like G.729), > pass-through plus library detection (like Opus), everything (like Speex)? > > pass-through including fmtp negotiation (level 1) > | pass-through plus library detection in ./configure (level 2) > | | transcoding module in codecs/codec_* (level 3) > | | | > [x] [x] [x] Codec 2 <http://www.rowetel.com/codec2.html> > [x] [x] [x] iLBC 20; the other mode iLBC 30 is available already > [ ] [ ] [ ] SILK; deprecated since September 2012 in favor of Opus > [x] [x] [ ] GSM-EFR; GSM-FR is available already > [x] [x] [ ] AMR and AMR-WB > > Your opinion? Please, set/change your checkmarks as desired! Of course, I am > able to find a contra position for each codec. Of course, I would like to > see complete support for all codecs (level 3). Anyway, some arguments why I > implemented those codecs: > * SILK/24 is the only default HD codec in the famous CSipSimple for Android. > All other HD codecs must be enabled in CSipSimple manually. > * GSM-EFR is default in Stock-Android, optional in Voice-over-LTE (VoLTE) > Beside AMR and GSM-FR the only codec with compression (the rest is G.711). > * AMR(-WB) are mandatory when linked to VoLTE (3GPP TS 26.103 chapter 7) > * Codec 2 does not have a MIME media-type specification, yet. > However, Codec 2 is supported in FreeSWITCH and CSipSimple. > * iLBC 20 was an apprentice piece thanks to the patch in ASTERISK-18094. > > > Question #2: Format-attribute Keys as Header Files? > > In Asterisk 13, some formats do have a header file in include/asterisk/, > like CELT and Opus (*_attr_keys). However internally, nobody consumes those > headers anymore. Some format-attribute modules offer "format_attribute_set" > but nobody uses that either. Is it OK, to create new header files? For which > use-case should I implement "format_attribute_set"? > > > Question #3: Module Loading Priority > > H.26x modules load very late (AST_MODPRI_DEFAULT). The Opus Codec module > loads very early (AST_MODPRI_CHANNEL_DEPEND). Which one is correct? Or are > both and there is a reason why video modules load later than audio modules? > > > Question #4: Orphan Format-attribute module CELT > > Currently, Asterisk 13 does not offer even pass-through of SILK or CELT. > Still their header files and their format-attribute modules are present. Is > there a reason for this? > For example, the SILK module contains several bugs, like creating several > fmtp lines instead of one, is able to parse only a single parameter, and > does not return a default when used with an internal transcoding module. > > > Final Sentence > > Although it would be nice to see at least some of this work in Asterisk, I > am mainly interested in a code-review. Is it possible to submit everything > for code-review even if there is no chance to pass? > > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
