Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
On Tue, Jul 19, 2016 at 12:59 PM, Sean Bright wrote: > On 7/19/2016 10:35 AM, Matt Fredrickson wrote: >> >> Response below. >> >> On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse >> wrote: >>> >>> Technical Details (at the moment the modifications are based upon >>> 13.6.0): >>> * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) >>> * Extended softmix for stereo support (downmixing) >>> * Extended the default confbridge (basically added a convolution engine) > > > If Opus is a required part of the implementation - and from reading the > description of the work being done it appears to be - wouldn't that make > this ineligible for inclusion? My hope is that there is a large amount of core Asterisk work which does not explicitly require Opus for it to be merged. I'm hoping that our current Opus legal impasses will not go on forever, but until things clear up with regards to it, this is the best response I can give. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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
Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
On 07/19/2016 01:56 PM, Sean Bright wrote: > On 7/19/2016 4:00 PM, Bruce Ferrell wrote: >> On 07/19/2016 10:59 AM, Sean Bright wrote: >>> On 7/19/2016 10:35 AM, Matt Fredrickson wrote: Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse wrote: > Technical Details (at the moment the modifications are based upon 13.6.0): > * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) > * Extended softmix for stereo support (downmixing) > * Extended the default confbridge (basically added a convolution engine) >>> If Opus is a required part of the implementation - and from reading the >>> description of the work being done it appears to be - wouldn't that make >>> this ineligible for inclusion? >>> >>> Kind Regards, >>> Sean >>> >> Just for clarification, why? speex is included. similar if not the same >> license? > > The last update I've seen on Digium's position in regards to the inclusion of > Opus support in the core is here: > > http://lists.digium.com/pipermail/asterisk-dev/2013-May/060419.html > > I don't believe they have publicly changed their stance on the topic since > then, but I'd be happy to be proven wrong. > > Kind regards, > Sean > > > great answer Sean. Thanks -- _ -- 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
Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
On 7/19/2016 4:00 PM, Bruce Ferrell wrote: On 07/19/2016 10:59 AM, Sean Bright wrote: On 7/19/2016 10:35 AM, Matt Fredrickson wrote: Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse wrote: Technical Details (at the moment the modifications are based upon 13.6.0): * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) * Extended softmix for stereo support (downmixing) * Extended the default confbridge (basically added a convolution engine) If Opus is a required part of the implementation - and from reading the description of the work being done it appears to be - wouldn't that make this ineligible for inclusion? Kind Regards, Sean Just for clarification, why? speex is included. similar if not the same license? The last update I've seen on Digium's position in regards to the inclusion of Opus support in the core is here: http://lists.digium.com/pipermail/asterisk-dev/2013-May/060419.html I don't believe they have publicly changed their stance on the topic since then, but I'd be happy to be proven wrong. Kind regards, Sean -- _ -- 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
Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
On 07/19/2016 10:59 AM, Sean Bright wrote: > On 7/19/2016 10:35 AM, Matt Fredrickson wrote: >> Response below. >> >> On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse >> wrote: >>> Technical Details (at the moment the modifications are based upon 13.6.0): >>> * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) >>> * Extended softmix for stereo support (downmixing) >>> * Extended the default confbridge (basically added a convolution engine) > > If Opus is a required part of the implementation - and from reading the > description of the work being done it appears to be - wouldn't that make this > ineligible for inclusion? > > Kind Regards, > Sean > Just for clarification, why? speex is included. similar if not the same license? -- _ -- 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
Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
On 7/19/2016 10:35 AM, Matt Fredrickson wrote: Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse wrote: Technical Details (at the moment the modifications are based upon 13.6.0): * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) * Extended softmix for stereo support (downmixing) * Extended the default confbridge (basically added a convolution engine) If Opus is a required part of the implementation - and from reading the description of the work being done it appears to be - wouldn't that make this ineligible for inclusion? Kind Regards, Sean -- _ -- 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
Re: [asterisk-dev] Asterisk goes Spatial Conferencing: the STEAK project
Response below. On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse wrote: > Hello, > > spatial audio for telephone conferencing has been proclaimed as a > silver bullet for speech-based telephone conferencing. > For research, we needed a _production-ready_ system that provides a > centralized conferencing bridge being able to provide binaural > synthesized spatial audio (i.e., for a pair of headphones). > As none was available (except for proprietary solutions), we chose to > get our hands dirty and extended Asterisk. > Most important, the modifications were required to be complaint to > VoIP standards (no fancy protocol extensions). > We succeeded and the modifications are now at a stage that makes them > suited to be merged back upstream. > Or at least start the discussion on this topic. > > Technical Details (at the moment the modifications are based upon 13.6.0): > * Enabled OPUS (with incoming stereo and outgoing stereo [interleaved]) > * Extended softmix for stereo support (downmixing) > * Extended the default confbridge (basically added a convolution engine) > > We chose OPUS (beside being used for WebRTC) as it allows to > RTP-channels with stereo (L16 or AMR-WB+ are not really alternatives). > Convolution was implemented via libfftw3 and the required HRTFs are at > the moment compiled into Asterisk. > > We would now like to bring the changes upstream as we think this is an > important feature (that might set Asterisk apart) and is not too hard > to maintain. > > More detailed information are available at our website: > http://steakconferening.de > and in the source code (branch >>steak-13.6.0<<): > https://github.com/steakconferencing/asterisk > > In addition, we host WebRTC- based demo (the real system): > https://demo.steakconferencing.de That sounds like a really exciting project! In order to move things forward and get things merged back upstream, I would advise 1.) Reading over our patch submission process [1] as well as our coding guidelines [2]. 2.) Opening up a code review on gerrit.asterisk.org [3] [4]. 3.) While not mandatory for the master branch, it is strongly encouraged to submit tests for any code submitted [5], particularly for code that you would like to function for any length of time in the future. If it's any code that involves deep architectural changes, expect much (presumably positive) discussion and feedback on it :-) Hope that helps, and I look forward to seeing your changes on gerrit. Matthew Fredrickson [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2] https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines [3] https://wiki.asterisk.org/wiki/display/AST/Code+Review [4] https://wiki.asterisk.org/wiki/display/AST/Code+Review+Checklist [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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
