Then lets get started. Due to the relatively large number of changes, we will split them up into individual patches:
1) Support for interleaved stereo (mainly softmix and struct channel) 2) Extension of confbridge with binaural synthesis via convolution (if channel supports stereo) For the patches, we will remove the hard dependency to OPUS (although we will stick to using it) and also enable L16 with stereo. Nevertheless, there are still some open questions: 1. Storage and description of HRTFs Impulse responses are at the moment compiled into Asterisk as a header file. This header file is generated using a custom C-program converting a multi-channel wave file into a float-array - hrirs_fabian.wav was taken from the SoundScapeRenderer https://github.com/SoundScapeRenderer/ssr/tree/master/data/impulse_responses/hrirs For positioning a sound source, the HRTFs for the left and for the right ear need to be selected according to the desired angle. This information is at the moment hard-coded as we use a 720-channel wave (interleaved: left, right) to cover 360 degrees completely. Would you prefer to compile the HRTFs into Asterisk (incl. the hard-coded description) or rather make this configurable? For the second option, we would need some support in terms of how to add configuration options. 2. Configuration of positioning of conference participants The positioning of individual participants is at the moment hard-coded (compiled via header-file). This is basically an array containing the angles at which participant _n_ is going to be seated. This could also be made configurable via configuration file. Furthermore, all participants of a conference receive the _same_ acoustical environment (i.e., participant 1 always sits in front of the listener etc.). This limits the computational requirements while individual listeners cannot configure their desired seating order. In fact, the own signal of a participant is subtracted after rendering the whole environment before sending the signals back. 3. Internal sampling rate of Confbridge The binaural synthesis is at the moment conducted at 48kHz. This is actually due to our use of OPUS, which always uses 48kHz for the decoded signals. Is this ok? 4. Is the dependency to libfftw3 an issue? We look forward to your feedback and will spent some time preparing the patches. --- Dennis Guse On Wed, Jul 20, 2016 at 3:53 PM, Matthew Jordan <mjor...@digium.com> wrote: > > > On Tue, Jul 19, 2016 at 4:59 PM, Matt Fredrickson <cres...@digium.com> > wrote: >> >> On Tue, Jul 19, 2016 at 12:59 PM, Sean Bright <sean.bri...@gmail.com> >> wrote: >> > On 7/19/2016 10:35 AM, Matt Fredrickson wrote: >> >> >> >> Response below. >> >> >> >> On Mon, Jul 18, 2016 at 7:18 AM, Dennis Guse >> >> <dennis.g...@alumni.tu-berlin.de> 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. >> > > The only thing that we can't include at this time is the codec module > itself, which - if implemented correctly - should be a separate shared > object library. That means that all the rest can be put up for code review, > and anyone who wants to take advantage of the new feature(s) can simply load > an externally available codec_opus. > > So - if you're interested - please do contribute the patches back upstream. > > -- > Matthew Jordan > Digium, Inc. | CTO > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- 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 -- _____________________________________________________________________ -- 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