Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Paul Hodges]

--On 19 March 2014 18:47 +0100 David Pickett  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

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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[David Pickett]

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  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
>
>
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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Andy Furniss]

Paul Hodges wrote:


Or maybe I just don't understand digital audio properly, or at least
the Internet aspect of it.


I think it's just like playing any compressed audio file. AIUI they are
just streaming 4 channel AAC, so with the buffering that is implicit in
streaming you may as well be playing an mp3 from disk as far as your h/w
knows.

The fact that the BBC have chosen to use mpeg-dash for me at least on
linux is just a hinderance that prevents me having control.

They could IMHO just have streamed this as they do the 320kbit AAC 2ch
R3 stream, which would mean that I could just record it to directly to
disk and have total control over the playback.

As it is if I were to install chrome I would have no idea what/how good
their AAC decoder is. Taking HD TV 5.1 AAC from the proms for example.
What you hear if using say vlc as a player is going to be different
depending on whether it's using FAAD to decode or ffmpeg.

FAAD implements dynamic range control ffmpeg doesn't and FAAD is
actually buggy in it's implementation sometimes, though the bug is
probably not noticable  on a prom, the fact that it uses DRC will make a
difference.

I guess there is no DRC in the R3 test - but the variability of AAC
decoders and being forced to use an unknown one is something to consider
if critically judging sound quality.


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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Fons Adriaensen]

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)

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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Paul Hodges]

--On 20 March 2014 12:02 +0100 David Pickett  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

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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Paul Hodges]

--On 20 March 2014 11:14 + Andy Furniss  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

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[Sursound] TSL SoundField new, and free, SurroundZone2

[Daniel Courville]

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


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Re: [Sursound] TSL SoundField new, and free, SurroundZone2

[Jon Honeyball]

I bought the previous version. Was a pile of crap. Took ages to get a
64bit AU version. And it was licensed with ilok which makes me shudder.
And should have been free, given how little it did compared to Harpex. Why
lock such a tool when its really of no use to anyone who hasn¹t got a
sound field, or who works in ambisonics and thus might get a sound field?

Good to see this one is free. Lets see how well it handles 192k 24bit B
format. Wonder if it is single threaded?

jon

On 20/03/2014 12:16, "Daniel Courville"  wrote:

>
>
>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
>
>
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[Sursound] Soundfield 450 Mk2

[Jon Honeyball]

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

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[Sursound] asking for help

[umashankar manthravadi]

this is a small appeal for help.
 
we are about to complete 40 or  50 brahma microphones (both independent and 
built into zoom H2n). Calibration is going on and I am  quite pleased with the 
way the mics have turned out.
 
We will provide a microphone specific calibration file and Brahmavolver 
(windows and mac) for all who are getting this microphone.
 
I would like provide a set of links as well, so people can download software (a 
to b conversion, manipulation of B format signals, and conversion to other 
formats). many of the people buying Brahma are newcomers to surround sound so 
they will need the help. 
 
What I would like the group to do is help me put together this list of links. I 
know of quite a few programs (VVMic tops the list) but I am sure I will miss 
many things as I grow old and absent minded.
 
umashankar
  
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Re: [Sursound] asking for help

[Fons Adriaensen]

On Thu, Mar 20, 2014 at 06:35:57PM +0530, umashankar manthravadi wrote:
  
> We will provide a microphone specific calibration file and Brahmavolver
> (windows and mac) for all who are getting this microphone.
>  
> I would like provide a set of links as well, so people can download software
> (a to b conversion, manipulation of B format signals, and conversion to other
> formats). many of the people buying Brahma are newcomers to surround sound so
> they will need the help. 

I'd be happy tp support Brahma wiht TetraProc [1] provided 
the calibration file format is open and documented. As far
as I can understand it just defines a matrix of 16 IRs which
TetraProc can handle without problem.

Alternatively, you could provide the measurements (under NDA
if you prefer) and let me create the Tetraproc config file.

Ciao,

[1] http://kokkinizita.linuxaudio.org/linuxaudio/tetra-pict.html

-- 
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)

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Re: [Sursound] Soundfield 450 Mk2

[Jörn Nettingsmeier]

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

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Re: [Sursound] TSL SoundField new, and free, SurroundZone2

[Paul Hodges]

> 

Note that the Windows installer puts the 32-bit VST plugin where the
64-bit one should be and vice versa.

Paul


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Re: [Sursound] TSL SoundField new, and free, SurroundZone2

[Jon Honeyball]

Somewhat of a failure of beta testing?

On 20/03/2014 14:32, "Paul Hodges"  wrote:

>> 
>
>Note that the Windows installer puts the 32-bit VST plugin where the
>64-bit one should be and vice versa.
>
>Paul
>
>
>-- 
>Paul Hodges
>
>
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Re: [Sursound] Soundfield 450 Mk2

[Jon Honeyball]

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" 
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
>
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Re: [Sursound] TSL SoundField new, and free, SurroundZone2

[John Leonard Main]

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  wrote:
> 
> 
> 
> 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
> 
> 
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Re: [Sursound] TSL SoundField new, and free, SurroundZone2

[Jon Honeyball]

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"  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 
>>wrote:
>> 
>> 
>> 
>> 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
>> 
>> 
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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Andy Furniss]

Paul Hodges wrote:

--On 20 March 2014 11:14 + Andy Furniss 
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.



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Re: [Sursound] BBC Radio Three Surround Streaming Trial (15. to 31. March)

[Paul Hodges]

--On 20 March 2014 15:41 + Andy Furniss  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

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Re: [Sursound] calibrating ambisonic system. Was Re:Sursound Digest, Vol 68, Issue 18

[Dave Hunt]

Hi,

I've changed the subject title as I was deservedly reprimanded for  
not changing it in my initial post.



From: Fons Adriaensen 
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 
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 

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
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Re: [Sursound] Soundfield 450 Mk2

[Len Moskowitz]

Jon Honeyball  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 


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Re: [Sursound] Soundfield 450 Mk2

[Daniel Courville]

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


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Re: [Sursound] calibrating ambisonic system. Was Re:Sursound Digest, Vol 68, Issue 18

[Fons Adriaensen]

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)

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