[asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)
André Valentin
avalentin at marcant.net
Fri Nov 27 02:30:44 CST 2015
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?
>
>
>
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