Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
--On 19 March 2014 18:47 +0100 David Pickett d...@fugato.com wrote: A bit scary that, as Paul said, Microsoft employ SRC automatically if you get it wrong... Actually, after thinking about this, I suspect it's unavoidable in this situation, even if the data rates are matched. Where's the master clock? My interface (E-MU 1616m) can be clocked from a digital input (SPDIF or ADAT), or it can be internally clocked. If I am playing back files, the interface provides the clock for the computer. But if the data is coming from the Internet, that data cannot but be separately clocked, but it can't provide the master clock for the interface - it just hasn't the stability, and buffering will get in the way and so on. My present method of recording goes through the interface, so it seems to me that resampling is unavoidable - the only way to get around that would be to have a method of recording the stream from the Internet directly without playing it out to the interface. Maybe TotalRecorder taps in to the audio stream before the SRC - I must look into that. Or maybe I just don't understand digital audio properly, or at least the Internet aspect of it. Oh, and I understand that the current Windows SRC is pretty good, actually. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
At 11:17 20-03-14, Paul Hodges wrote: I understand that the current Windows SRC is pretty good, actually. I probably know less about this than you do. But do you mean Windows 8? Windows 7, which I am using appear to have a problem, but with a secret fix that I havent tried yet: http://support.microsoft.com/kb/2653312 http://www.indexcom.com/tech/WindowsAudioSRC/ The biggest problem is in knowing what actually goes on behind the scenes. I have recorded the broadcasts with Samplitude set to 48k. I am using ASIO, so maybe the problem DOESNT exist. My interface is RME UFX (in Loopback Mode) and it claims that it is running under internal sync, from which I understand Samplitude to be the master clock. As Samplitude uses input buffering, would this not take care of potential glitches due to the asynchronous nature of the stream? RME doesnt do any D/A until I listen to it, and no errors were reported by the software. But I am guessing... David --On 19 March 2014 18:47 +0100 David Pickett d...@fugato.com wrote: A bit scary that, as Paul said, Microsoft employ SRC automatically if you get it wrong... Actually, after thinking about this, I suspect it's unavoidable in this situation, even if the data rates are matched. Where's the master clock? My interface (E-MU 1616m) can be clocked from a digital input (SPDIF or ADAT), or it can be internally clocked. If I am playing back files, the interface provides the clock for the computer. But if the data is coming from the Internet, that data cannot but be separately clocked, but it can't provide the master clock for the interface - it just hasn't the stability, and buffering will get in the way and so on. My present method of recording goes through the interface, so it seems to me that resampling is unavoidable - the only way to get around that would be to have a method of recording the stream from the Internet directly without playing it out to the interface. Maybe TotalRecorder taps in to the audio stream before the SRC - I must look into that. Or maybe I just don't understand digital audio properly, or at least the Internet aspect of it. Oh, and I understand that the current Windows SRC is pretty good, actually. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
On Thu, Mar 20, 2014 at 10:17:30AM +, Paul Hodges wrote: But if the data is coming from the Internet, that data cannot but be separately clocked, but it can't provide the master clock for the interface - it just hasn't the stability, and buffering will get in the way and so on. Not really. For the same reasons it would be impossible to resample, since this requires an accurate ans stable (i.e. at most changing very slowly) estimate of the ratio. It *is* possible to extract a stable clock from the jittery timing of the internet data (basically a SW DLL with some extensions to allow for missing packets etc.). The real problem is that most sound cards don't provide any means to use this information - they only accept a physical external clock signal. There are some exceptions, e.g. some RME cards have a software interface that allows to change the master clock frequency (a multiple of the sample rate) in very small steps. Using this the SW can sync the card to the data instead of resampling it. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
--On 20 March 2014 12:02 +0100 David Pickett d...@fugato.com wrote: I understand that the current Windows SRC is pretty good, actually. I probably know less about this than you do. But do you mean Windows 8? Windows 7, which I am using appear to have a problem, but with a secret fix that I havent tried yet: I knew about that problem, which was gross and inexcusable, and applied the fix, which is fine. But now I'm running Windows 8.1 in any case. The biggest problem is in knowing what actually goes on behind the scenes. I have recorded the broadcasts with Samplitude set to 48k. I am using ASIO, so maybe the problem DOESNT exist. But it's coming out of the browser and into the interface via the MS audio stack; you're only using ASIO for recording the data coming from the interface. My interface is RME UFX (in Loopback Mode) and it claims that it is running under internal sync, from which I understand Samplitude to be the master clock. I presume the TotalMix architecture is fairly similar to the PatchMix I'm using. In the E-MU internal sync means internal to the interface, which is providing the master clock to the software. I don't think the software ever provides the master clock. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
--On 20 March 2014 11:14 + Andy Furniss adf.li...@gmail.com wrote: I think it's just like playing any compressed audio file. But it isn't, because a slight mismatch in clock speeds would mean that the playback could run ahead and eventually run out of buffered samples to play. Of course, this issue is the same for any Internet audio, and always has been. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
[Sursound] Soundfield 450 Mk2
Pity this *still* doesn¹t have a 1K tone generator at say -20dB, allowing you to properly calibrate the input levels and replay of your recorder, which almost certainly doesn¹t have ganged controls. The Sounddevices 788T can do this, but others can¹t. Missed opportunity. Jon ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Soundfield 450 Mk2
On 03/20/2014 01:57 PM, Jon Honeyball wrote: Pity this *still* doesn¹t have a 1K tone generator at say -20dB, allowing you to properly calibrate the input levels and replay of your recorder, which almost certainly doesn¹t have ganged controls. The Sounddevices 788T can do this, but others can¹t. Missed opportunity. well, since the box produces b-format, the calibration of the recorder is not quite as critical as with (say) the tetramic. but yeah, a test tone wouldn't have hurt :) i usually use it either with unity gain line inputs or with PGAs, so its absence doesn't bite me much. you could get one of those phantom-powered XLR test tone plugs to align your 788T before you plug in the soundfield. -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] TSL SoundField new, and free, SurroundZone2
http://www.tslproducts.com/soundfield/surroundzone2/ Note that the Windows installer puts the 32-bit VST plugin where the 64-bit one should be and vice versa. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] TSL SoundField new, and free, SurroundZone2
Somewhat of a failure of beta testing? On 20/03/2014 14:32, Paul Hodges pwh-surro...@cassland.org wrote: http://www.tslproducts.com/soundfield/surroundzone2/ Note that the Windows installer puts the 32-bit VST plugin where the 64-bit one should be and vice versa. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Soundfield 450 Mk2
I align my 788T with my Prism dScope, but that’s cheating :-) Quite a few cheaper devices don’t have ganged inputs, and it would help. I asked Soundfield for this several times, but clearly they disagree On 20/03/2014 14:07, Jörn Nettingsmeier netti...@stackingdwarves.net wrote: On 03/20/2014 01:57 PM, Jon Honeyball wrote: Pity this *still* doesn¹t have a 1K tone generator at say -20dB, allowing you to properly calibrate the input levels and replay of your recorder, which almost certainly doesn¹t have ganged controls. The Sounddevices 788T can do this, but others can¹t. Missed opportunity. well, since the box produces b-format, the calibration of the recorder is not quite as critical as with (say) the tetramic. but yeah, a test tone wouldn't have hurt :) i usually use it either with unity gain line inputs or with PGAs, so its absence doesn't bite me much. you could get one of those phantom-powered XLR test tone plugs to align your 788T before you plug in the soundfield. -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister VDT http://stackingdwarves.net ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] TSL SoundField new, and free, SurroundZone2
SurroundZone2 is not too bad, either. Not quite as versatile as Harpex, but a definite upgrade on version 1, and you can't argue with the price. As to the Mk 2 mic, I did enquire about the cost of upgrading the existing ST450 control unit, but was quoted just under £2,000.00 exc. VAT, which is too rich for my blood, so I'll stick with the original box for few more years. Regards, John Sent from my iPad On 20 Mar 2014, at 12:16, Daniel Courville courville.dan...@uqam.ca wrote: http://www.tslproducts.com/soundfield/surroundzone2/ Supported Input Formats: * A-Format (SPS200) * B-Format Supported Output Formats: * Stereo (and Mono) * 5.0, 5.1, 6.0, 6.1, 7.0, 7.1 * B-Format Especially a good news for Pro Tools users as this makes the SurroundZone2 a good, free, B-Format manipulator. Thanks to TSL and Pieter Schillebeeckx! - Daniel ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] TSL SoundField new, and free, SurroundZone2
I asked about upgrading my 350, or rather asked about if it was worth upgrading either the mic or the box, depending on which was the more critical. But I got nowhere ‹ they seemed to think I wanted to trade in part of the mic, which wasn¹t the case at all. Ah well. On 20/03/2014 14:40, John Leonard Main j...@johnleonard.co.uk wrote: SurroundZone2 is not too bad, either. Not quite as versatile as Harpex, but a definite upgrade on version 1, and you can't argue with the price. As to the Mk 2 mic, I did enquire about the cost of upgrading the existing ST450 control unit, but was quoted just under £2,000.00 exc. VAT, which is too rich for my blood, so I'll stick with the original box for few more years. Regards, John Sent from my iPad On 20 Mar 2014, at 12:16, Daniel Courville courville.dan...@uqam.ca wrote: http://www.tslproducts.com/soundfield/surroundzone2/ Supported Input Formats: * A-Format (SPS200) * B-Format Supported Output Formats: * Stereo (and Mono) * 5.0, 5.1, 6.0, 6.1, 7.0, 7.1 * B-Format Especially a good news for Pro Tools users as this makes the SurroundZone2 a good, free, B-Format manipulator. Thanks to TSL and Pieter Schillebeeckx! - Daniel ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
Paul Hodges wrote: --On 20 March 2014 11:14 + Andy Furniss adf.li...@gmail.com wrote: I think it's just like playing any compressed audio file. But it isn't, because a slight mismatch in clock speeds would mean that the playback could run ahead and eventually run out of buffered samples to play. Of course, this issue is the same for any Internet audio, and always has been. Yea, I did also mention buffering, just that I assume the buffer is normally big enough so that it doesn't run out/overflow in a reasonable time. I guess broadcast is in the same situation, mp2/aac/ac3 are all ahead by 600 ms in transport streams, though I don't know why, it could be to give some leeway. Transport streams do have extra clock timestamps, but I wonder if TVs/receivers that have internal dacs + spdif + hdmi out really can/do slave the clocks that control them to the clock reference in the stream or whether they rely on buffering to help. Looking at a sample of R3 AAC downloaded to disk I see it has presentation time stamps - I am not sure how (or if) players handle the soundcard clock being at a different rate to the master clock. The open source video players I use base their a/v sync on sound, so will adjust video to fit the sound card clock. There are exeptions, XBMC does have an option to sync to video and resample sound to fit. ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)
--On 20 March 2014 15:41 + Andy Furniss adf.li...@gmail.com wrote: Yea, I did also mention buffering, just that I assume the buffer is normally big enough so that it doesn't run out/overflow in a reasonable time. This is not the kind of programming I have ever done, and it makes me uncomfortable - not that I can see any good alternative. Paul -- Paul Hodges ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] calibrating ambisonic system. Was Re:Sursound Digest, Vol 68, Issue 18
Hi, I've changed the subject title as I was deservedly reprimanded for not changing it in my initial post. From: Fons Adriaensen f...@linuxaudio.org Date: 19 March 2014 20:21:34 GMT To: sursound@music.vt.edu Subject: Re: [Sursound] Sursound Digest, Vol 68, Issue 17 On Wed, Mar 19, 2014 at 07:50:43PM +, Dave Hunt wrote: Surely the best approach is to feed the noise signal (post decoder) into each speaker channel in turn and adjust the amplification on each channel until the level measured at the centre listening point is the same for each speaker. That would be a prerequisite for the method I explained. But it still leaves you with an uncalibrated system, as the decoder gain (no matter how you define it) isn't included. Agreed. I was just stating the prerequisite, as this seemed to have been missed. From: Fons Adriaensen f...@linuxaudio.org Date: 19 March 2014 20:30:01 GMT To: sursound@music.vt.edu Subject: Re: [Sursound] Sursound Digest, Vol 68, Issue 17 Reply-To: Surround Sound discussion group sursound@music.vt.edu On Wed, Mar 19, 2014 at 07:50:43PM +, Dave Hunt wrote: The panning approach won't work, as all speakers would be excited at various different levels. Anything to substantiate that claim ? Practice ? Theory ? W will excite all speakers equally, assuming the decoder is correct and all speaker gains matched to produce the same level at the centre. Higher orders will excite more than one speaker at differing levels and polarity. This crosstalk (for want of a better word) falls as the order (and number of speakers) rises. FYI, I have used this method a number of times, with excellent results. I've also done the maths that show it do be correct. I'm sure it is, assuming that the basic installation and set up is well and thoroughly done. It would be very confusing and time consuming to attempt to use your method without having done the prerequisite. It is always a bit disconcerting to hear how, with even a set of identical good speakers producing the same level of pink noise, each sounds different. Yes, rooms and positions in them sound different, but this is often observable when just swopping one speaker for another in the same location. Ciao, Dave -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20140320/7f5a05b5/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Soundfield 450 Mk2
Jon Honeyball j...@jonhoneyball.com wrote: Pity this *still* doesn't have a 1K tone generator at say -20dB, allowing you to properly calibrate the input levels and replay of your recorder, which almost certainly doesn't have ganged controls. The Sounddevices 788T can do this, but others can't. The Tascam DR-680 has ganged controls. Len Moskowitz (mosko...@core-sound.com) Core Sound LLC www.core-sound.com Home of TetraMic ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Soundfield 450 Mk2
Le 2014-03-20 08:57, Jon Honeyball a écrit : Pity this *still* doesn¹t have a 1K tone generator at say -20dB, allowing you to properly calibrate the input levels and replay of your recorder, which almost certainly doesn¹t have ganged controls. The Sounddevices 788T can do this, but others can¹t. Missed opportunity. While ganged level controls are quite necessary for the SPS200 or the TetraMic, the ST350/450/450 MkII output at line level and numerous input devices, either portable or studio-dedicated, have fixed line level inputs. - Daniel ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] calibrating ambisonic system. Was Re:Sursound Digest, Vol 68, Issue 18
On Thu, Mar 20, 2014 at 06:37:48PM +, Dave Hunt wrote: Agreed. I was just stating the prerequisite, as this seemed to have been missed. It may not be essential. If you start with all the channels at more or less the right gain (e.g. by setting analog controls to the same position), and then repeat the panned measurement a number of times, each time adjusting the channel panned to, the gains will converge. It's the same process as inverting a matrix with dominant diagonal elements by iteration. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
