Re: [Sursound] vertical precendence and summing localisation (wallis and lee 2015)

[Andy Furniss]

Joseph Anderson wrote:


effect. (Of course, all built out of the SuperCollider version of the ATK
.)


You may want to get rid of the soundofspace links as they point to 
something quite different now.



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Re: [Sursound] Binaural broadcast

[Andy Furniss]

Richard wrote:

From the program link David gave - the binaural version is
website/iplayer only.


Many thanks for the heads up. I think i'd best warn folks hear in the
K this will be best heard on the FM service, as the broadcasts via
DAB, Freeview and Satellite use Jstereo to help reduce bandwidth
usage. Apart from the damage it does to standard stereo material, it
will seriously damage the binaural information.




I believe one, possibly two, items due to be broadcast on BBC Radio
4 on Saturday 31 October as part of a 'Fright Night' strand, were
recorded in binaural stereo.

22:00 GMT  "The Stone Tape"  A story by Nigel Kneale (he of
'Quatermass' fame)

23:00 GMT  "Ring" An adaptation of Koji Suzuki's 1991 novel.

If the BBC website won't co-operate for non-UK residents, no doubt
Expat Shield will spoof a UK IP for you.

-- Peter Carbines

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Re: [Sursound] ambisonics audio for 360 film (VR)

[Andy Furniss]

David Pickett wrote:

Is everybody on the list getting gibberish (éäJ, etc), as I am?


It works for me. Maybe your mail prog didn't like it as it wasn't a
normal mail = the raw source looks like -

snip

Reply-To: Surround Sound discussion group sursound@music.vt.edu
Content-Type: text/plain; charset=utf-8
Content-Transfer-Encoding: base64
Errors-To: sursound-boun...@music.vt.edu
Sender: Sursound sursound-boun...@music.vt.edu
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snip


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Re: [Sursound] ambisonics audio for 360 film (VR)

[Andy Furniss]

Fons Adriaensen wrote:

On Tue, Aug 18, 2015 at 08:16:05PM +0100, Andy Furniss wrote:


David Pickett wrote:

Is everybody on the list getting gibberish (éäJ, etc), as I am?


It works for me. Maybe your mail prog didn't like it as it wasn't
a normal mail = the raw source looks like -

CgoKSGVuayB8IFNwb29rLmZtIDxoZW5rQC4uLj4gd3JpdGVzOgoKPiAKPiBEZWFyIFN1cnJvdW5k



IFNvdW5kIERpc2N1c3Npb24gR3JvdXAsCj4gCj4gSSBoYXZlIGJlZW4gZm9sbG93aW5nIHRoZSBl


Which is just 'base64' encoding, something any mail program is
supposed to be able to handle -- it's quite an old standard.



Yea, maybe I should have said plain instead of normal.

I was just thinking Dave was possibly seeing base64 as trashed and plain
as OK, in which case it would be a clue as to what was going on with his
client.
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Re: [Sursound] Ambisonics recording of LOUD night club or venue?

[Andy Furniss]

Eric Carmichel wrote:

Greetings Everyone,I haven't posted in ages--a move to Silicon Valley
over a year ago has occupied my time.Does anybody have a Soundfield
recording of a loud nightclub or live music venue? I mean really LOUD
electronic dance or rock music. I understand this isn't something
you'd normally take a high-end mic to, but I need an accurate
representation of the atmosphere. I have live recordings taken from
feeds, but these aren't representative of what the sound is really
like. A binaural or monaural recording (with quality mics) would
help, too, but marginal quality recordings made with a Smartphone
won't work (otherwise I'd go to YouTube and find tons of $%#@).I
checked uploaded recordings linked to the ambisonic net site: Very
cool stuff, but not what I need for a particular study.Best
regards,Eric


http://www.core-sound.com/sampler.php

Has some Binaural one of which claims to be very loud - real head though.
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[Sursound] BBC Radio 3 surround broadcast

[Andy Furniss]

May be of interest to some on here - I don't know if it will be UK only 
though.


http://rdmedia.bbc.co.uk/radio3/

http://www.bbc.co.uk/programmes/b05202hw
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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

Rupert Brun wrote:


Some people have commented that we should make a normal surround
sound stream available but I'm not at all sure what they mean by
this. For live streaming you need a codec and a transport layer. Our
normal transport layers Flash and Shoutcast. Flash won't support
more than stereo, Shoutcast can but support for this is very
limited. MPEG-DASH, on the other hand, is rapidly becoming the
standard way to transport streaming media over IP with more and more
companies supporting it.


I've not got anything against DASH, though I do find the analogy with FM
a bit weak given the amount of time content was available on both AM and FM.

So what is a normal stream -I doubt I am that exceptional as a Linux
user in not using browser/flash to access BBC content.

Take for example the R3 HD stream - I see nothing exclusively to do with
browser/flash/Shoutcast. .pls is widely supported so all I would do is -

mplayer -playlist http://www.bbc.co.uk/radio/listen/live/r3_aaclca.pls

I could even download r3_aaclca.pls with wget look in it and download
with wget using the long http:// url.

Happily the stream is aac in adts over http many many players across all
OSes will be able to play it.

If it were 4 channel aac/adts then the decoder would see it as such.


If anybody would like to set up a demonstration of live streaming of
surround content which will play natively in a browser using
different technologies then please let me know. I'm not using
MPEG-DASH to be difficult, I'm using it because we have figured out
how to make it carry good quality surround sound reliably and play
in the browser without third party  plug-ins, and because it is an
agreed standard which is widely supported. If any of you have a
better way of doing that please share it!


I don't via browser - DASH is fine for your objectives, but I hope you
see what I mean by normal stream - You could provide one - just via a
link somewhere - the browser/flash requirement/restriction is your
arbitary choice. I accept that that's the way things currently are but
this is a trial, and not using flash is the point, so there's no harm if
you wanted in providing a real link to a normal http stream just like R3 HD.

As it's a trial that requires people to have their computer plumbed into
a surround system I don't think you can argue that they would be
incapable of directly using a player capable of surround output - I
would speculate that they already have one and know how to use it.




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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

Rupert Brun wrote:

the browser/flash requirement/restriction is your arbitary choice.



I doubt I am that exceptional as a Linux user in not using
browser/flash to access BBC content.



I could even download r3_aaclca.pls with wget look in it and
download with wget using the long http:// url.


I think we are getting to the nub of the issue. Wanting something to
play in the browser isn't my arbitrary choice, it's how normal
people consume media and I need to test something which will
eventually work for normal people.

As where many of the members of this group (and I mean this in the
nicest way and include myself) are not normal - we like to
experiment, we use Linux, we access the streams in ways the
broadcasters don't intend us to, we write code. This experiment
isn't for people like us - it is to test something which will one day
allow normal people to access surround sound through the web
browser. That's what the experiment is about - testing MPEG Dash
surround through a browser, because that's a strategic solution which
I believe will become mainstream. It's also about testing the
production challenges of one person creating a surround sound
balance and a stereo balance of a live classical music concert at the
same time, because I can't afford to have 2 sound balancers and I
don't like the results from automated upmixing or downmixing. And
finally, and perhaps most importantly, it's about finding out what
works aesthetically and what doesn't when using surround sound from a
live classi cal concert.


Fair enough, I can't disagree with any of that.



Of course one could devise an experiment which would deliver
surround sound to people who want to use Linux and wget but that
would be a different experiment which I am very happy to leave to
others to try.


Just to be clear, as I didn't reiterate in my last post, though I did
say previously - I understand this is a DASH/Browser test. I am not
suggesting there should be some sort of different experiment, indeed
it's nothing new - just the same as the existing R3 HD stream.

My concern was that the content for the trial is unique and currently
not available to all. The .pls type stream could have been in addition
to the trial stream not instead of it and purely to allow access for
those who didn't want to install a new closed browser/OS just to get it.
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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

Rupert Brun wrote:

The BBC will make the BBC Proms Concerts available in 4.0 using
MPEG-DASH. The stream will be available internationally.


So the're available to the world subject to what browser/os you use, but
not to me who pays the licence fee.

Why must MPEG-DASH be used for this?

It's not like it's some UHD Video that may need bitrate considerations.

Why not just do a 4ch version of the 320kbit ADTS R3 stream, which works
fine with pretty much anything, and also gives users the choice of aac
decoders.

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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

Jörn Nettingsmeier wrote:

On 07/18/2014 03:48 PM, Andy Furniss wrote:

Rupert Brun wrote:

The BBC will make the BBC Proms Concerts available in 4.0 using
MPEG-DASH. The stream will be available internationally.


So the're available to the world subject to what browser/os you
use, but not to me who pays the licence fee.

Why must MPEG-DASH be used for this?

It's not like it's some UHD Video that may need bitrate
considerations.

Why not just do a 4ch version of the 320kbit ADTS R3 stream, which
works fine with pretty much anything, and also gives users the
choice of aac decoders.


i'd guess auntie is as much interested in testing mpeg-dash as in 4.0
:)

adaptive streaming has its nice aspects, both from the content
provider's and the listener's point of view.



I agree dash may well be the future, though from the links in Ruperts
other post I see it's not adaptive for this test.

My issue is really just the limited browser/OS choice currently, maybe
if mine worked I wouldn't care :-)

I all for the future - just this content is unique and could be easily
made available as a normal stream in addition to the trial.

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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

Rupert Brun wrote:

We can't deliver over normal broadcast channels because we don't have
the bandwidth and there would be very significant costs for a small
audience.

We are using MPEG DASH because it handles surround sound well and can
be decoded entirely within the browser without any third party
software and this will become important as more and more consumer
devices such as TVs and set top boxes have web browsers built in. It
is also rapidly becoming the standard for audio and visual content
delivery over the web and we need to explore how it works.

There are more detailed explanations in the blogs from the first
experiment using this technology.
http://www.bbc.co.uk/blogs/radio3/posts/Radio-3-in-40
http://www.bbc.co.uk/rd/blog/2014/03/media-source-extensions


Please ask any further questions  via the blog or twitter using
#BBCProms4 not through this list, so the questions and answers can
reach a wider audience as I don't want to have to answer the same
questions twice if possible.


OK, no more questions, just grumbling :-)

Thanks for the explanation and links above are interesting and informative.

From one of them

People expect to be able to consume BBC content on any platform

Yep :-) I do, and this is unique content that I can't consume. There's
nothing wrong with trialling this, but given that you know browser
support is limited you could have easily provided an additional old
style stream.


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Re: [Sursound] BBC Proms in 4.0

[Andy Furniss]

David Pickett wrote:

At 21:38 18-07-14, John Leonard wrote:
 Stream working well here - shame it's wildly out of sync with the
 broadcast picture, though.
 
 Some distortion apparent on FF passages?
 

Yes, I heard some distortion on the violins in places: it sounds digital
but I wasnt sure where it came from, as I've heard similar from the ORF
web relays (stereo).


They apologised for sound issues at the end of the TV broadcast.

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

[Andy Furniss]

Dave Malham wrote:

48 kHz is pretty well the international standard sample rate for
broadcast organisation and has been since they started upgrading from
the 32kHz used (by the Beeb) for distributing audio to FM
transmitters back in the late 60's.

Dave


True I expect, but for some reason the normal 320kbit aac R3 web
stream is 44.1k.
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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 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.



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Re: [Sursound] ambiX v0.1 Ambisonic software, mcfx v0.1

[Andy Furniss]

Matthias Kronlachner wrote:


So I finally got to get round this one tonight - though the
workaround is probably sub optimal. I did try others, but jack was
in a state from all the crashing where nothing was working, leading
me to falsely think they were all failing and dismiss them.


thanks for the patience :-)


Thanks - it turned out that it wasn't jack in a state after all, it
was cpufreq on_demand being randomly useless and leaving my 3.4 GHz
cores @ 800MHz. I should have thought of that, as I knew it's rubbish
sometimes when SIMD/threads are involved.


The problem was that as soon as I made one connection the code
behaved like the output has 2 channels, when it only had one.

you are right, it seems i did not expect the binaural decoder to be
called by less than two channels. i solved it in a slightly different
way. thanks for spotting this! should be fixed now.


Yea, working OK now.



I can get sound, though it's not right as I suppose I need to use
converter to play a wxyz - but converter also crashes, so I'll have
to look at that next.

yes, you have to use the converter to play recordings that use FuMa
channel sequence and weighting. i just checked the converter and
could not find anything suspicious which could cause a crash. maybe
you are lucky again and find the cause?


I haven't attempted my own fix, or looked much yet, but looking at the
backtrace it seems to be a similar issue.

I am now compiling first order (JFTI, though I doubt it makes any
difference).

I can avoid the crash in at least 2 ways -

Connect outputs first.
Have 2D checked on input.

To produce the crash I need to connect the inputs first - the problem
being when connecting the second.

juce::FloatVectorOperations::copy
is #1 in the trace but judging by your previous fix I guess #3
ambix_converter/Source/PluginProcessor.cpp:537

will roughly be the place to change. I see you already had some
debugging there so uncommenting them (lines 503 and 532) I see lots of -

InputCh: 0 IN_CHANNEL: 0 OUT_CHANNEL: 0
NumInputChannels: 1 Buffersize: 1

When 1 input is connected then on connecting the second -

NumInputChannels: 2 Buffersize: 2
InputCh: 0 IN_CHANNEL: 0 OUT_CHANNEL: 0
InputCh: 1 IN_CHANNEL: 2 OUT_CHANNEL: 1
Segmentation fault

I have found a couple of other issues one trivial = the octagonal preset
looks wrong (12 channels).

The second could be more tricky as it doesn't seem to be reliably
reproducable = a crash in zita convolver when changing presets while
playing something.

Anyway thanks for the software and I'll continue testing etc. At least
now I can listen to some music while testing :-)

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Re: [Sursound] ambiX v0.1 Ambisonic software, mcfx v0.1

[Andy Furniss]

Andy Furniss wrote:

Matthias Kronlachner wrote:



what exactly are the problems with the standalone build? as they
are working fine for me under debian.





ambix_binaural_standalone_o5 will segfault when I connect its output
to system playback using qjackctl connections window.


So I finally got to get round this one tonight - though the workaround
is probably sub optimal. I did try others, but jack was in a state from
all the crashing where nothing was working, leading me to falsely think
they were all failing and dismiss them.

I guess that you knowing the code better will want something different,
but at least this prevents the crash for me.

diff --git a/ambix_binaural/Source/SpkConv.cpp 
b/ambix_binaural/Source/SpkConv.cpp

index a98ccec..508326f 100644
--- a/ambix_binaural/Source/SpkConv.cpp
+++ b/ambix_binaural/Source/SpkConv.cpp
@@ -289,7 +289,9 @@ void SpkConv::process(AudioSampleBuffer 
InputBuffer, AudioSampleBuffer OutputB


 // copy buffer to output
 OutputBuffer.addFrom(0, 0, outdata_l+bcp_out, NumSamples);
-OutputBuffer.addFrom(1, 0, outdata_r+bcp_out, NumSamples);
+
+if (OutputBuffer.getNumChannels()  1)
+OutputBuffer.addFrom(1, 0, outdata_r+bcp_out, NumSamples);

 #else
out_buffer.setSize(2,NumSamples,false,false,true);

The problem was that as soon as I made one connection the code behaved 
like the output has 2 channels, when it only had one.


A similar fix needed for non ZITA case I expect.

I can get sound, though it's not right as I suppose I need to use 
converter to play a wxyz - but converter also crashes, so I'll have to 
look at that next.

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Re: [Sursound] ambiX v0.1 Ambisonic software, mcfx v0.1

[Andy Furniss]

Matthias Kronlachner wrote:

On 06/02/14 21:34, Andy Furniss wrote:

Matthias Kronlachner wrote:

hi!

i finally got my ambisonic plug-ins to a stage where i'd like to
share them with the community. these are working under windows,
osx and linux as vst, lv2 and standalone applications. (the linux
version still needs some treat - help wanted!)


Thanks for getting this out.

I have tried just the standalone build on linux and can see there
are some issues.

what exactly are the problems with the standalone build? as they are
working fine for me under debian.


Maybe I am not the best testcase as I am using a pure 64 bit Linux from
scratch setup, though I don't have any issues with jack using
supercollider/ffmpeg/mplayer/ambdec.

Additionally because I don't yet have anything to plug the plugins into
and steinberg wanted me to register to get the SDK, I didn't get it, so
am building with VST disabled with ccmake.

Minor issue - have to build single threaded, despite the make -j advice
in the readme.

Running ambix_encoder_standalone_o5 if I abuse it by continually
moving the mouse around left button held, it will after a while lock
needing a pkill.

ambix_binaural_standalone_o5 will segfault when I connect its output to
system playback using qjackctl connections window.

Here's what valgrind says abut that -

==2142== Thread 4:
==2142== Invalid read of size 8
==2142==at 0x69864F: juce::FloatVectorOperations::add(float*, float 
const*, int) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0x697283: juce::AudioSampleBuffer::addFrom(int, int, 
float const*, int, float) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0x677BA3: 
Ambix_binauralAudioProcessor::processBlock(juce::AudioSampleBuffer, 
juce::MidiBuffer) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0x73E48B: 
juce::AudioProcessorPlayer::audioDeviceIOCallback(float const**, int, 
float**, int, int) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0x6883B4: 
juce::AudioDeviceManager::audioDeviceIOCallbackInt(float const**, int, 
float**, int, int) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0x6931D1: 
juce::JackAudioIODevice::processCallback(unsigned int, void*) (in 
/mnt/sdb1/Src64/ambix/ambix/build/_bin/standalone/ambix_binaural_standalone_o5)
==2142==by 0xB16E5E6: Jack::JackClient::CallProcessCallback() (in 
/usr/lib/libjack.so.0.1.0)
==2142==by 0xB16E4ED: Jack::JackClient::ExecuteThread() (in 
/usr/lib/libjack.so.0.1.0)
==2142==by 0xB16C52A: Jack::JackClient::Execute() (in 
/usr/lib/libjack.so.0.1.0)
==2142==by 0xB18AE8B: Jack::JackPosixThread::ThreadHandler(void*) 
(in /usr/lib/libjack.so.0.1.0)

==2142==by 0x6FFDD79: start_thread (pthread_create.c:308)
==2142==by 0x7D0DA9C: clone (clone.S:115)
==2142==  Address 0x0 is not stack'd, malloc'd or (recently) free'd
==2142==
==2142==
==2142== Process terminating with default action of signal 11 (SIGSEGV)
==2142==  Access not within mapped region at address 0x0



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Re: [Sursound] ambiX v0.1 Ambisonic software, mcfx v0.1

[Andy Furniss]

Matthias Kronlachner wrote:

hi!

i finally got my ambisonic plug-ins to a stage where i'd like to
share them with the community. these are working under windows, osx
and linux as vst, lv2 and standalone applications. (the linux version
still needs some treat - help wanted!)


Thanks for getting this out.

I have tried just the standalone build on linux and can see there are
some issues.

I am not used to cmake/C++/JUCE but time/luck permitting I'll try to
understand what is going on - it's not going to be quick or easy, though :-)
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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Stefan Schreiber wrote:

Andy Furniss wrote:



Nothing to do with this thread and I am not saying that any players
use it, but I did see the words Ambisonic and WXYZ in the spec, so
there is some provision for carrying and flagging as special
b-format in a DTS extension stream.



Maybe hidden in the TrueHD spec, but Blu-Ray doesn't support anything
 like WXYZ. Having worked a bit on disc standards before, and never
saw anything of this...


I don't claim any expertise in anything I write here and am often wrong
:-) but ...

I would say that's debatable, in that bluray players should accept
Dolby/DCA and both are specified so that decoders skip
substreams/extensions that they don't know about.

Additionally if instructed, the player can just pass on bitstreams
without decoding to a receiver without caring what it contains.

Of course there may not currently exist and decoders that could use
wxyz, but from a compatibility point of view I don't thing it would be
impossible for current disks/players to handle a stream with wxyz
embedded - they wouldn't know or care. Even a normal decoder wouldn't
care and could just do what it knows about. If a stream with wxyx were
not primary - and bluray can mux many soundtracks, you could even have
the core compatibility stream contain sound telling you that you need a
special setup to use this track.



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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Augustine Leudar wrote:

Hi Eric,
this is what I have done in a couple of permanent installations. I use
a motu ultralite and a mini pc  (or mac). This is fine for an
installation that will never be touched. However even using linux or a
mac there are problems with this setup for permanent installations if
the staff want to turn the system on and off each day - even doing my
best to secure all usb and power cables even in a few short months I
have had to go up a couple of times just to reconnect cables that have
been jogged etc Plus power cuts can cause problems with software on
any system.   I want to simplify the system - one dedicated box with
an on/off switch and a play button that staff can easily operate and
has the minimum amount of variables that can go wrong.
I think this program might do it :

http://www.dts.com/professionals/audio-software/dts-hd-master-audio-suite/overview.aspx

anyone used it ?


If you need 8 full range channels then it doesn't look like that will do 
it -


Up to 7.1 ch. Discrete for Primary Audio for Blu-ray Disc

What is this one dedicated box going to contain?

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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Stefan Schreiber wrote:


I would say that's debatable, in that bluray players should
accept Dolby/DCA and both are specified so that decoders skip
substreams/extensions that they don't know about.

Additionally if instructed, the player can just pass on
bitstreams without decoding to a receiver without caring what it
contains.

Of course there may not currently exist and decoders that could
use wxyz, but from a compatibility point of view I don't thing it
would be impossible for current disks/players to handle a stream
with wxyz embedded - they wouldn't know or care. Even a normal
decoder wouldn't care and could just do what it knows about.



This would be G format?


No b-format wxyz is the example given of an audio asset that may be
flagged as not directly being for speakers.


But I believe we already should aim for something far more complete.


Fair enough - I am in no way arguing in favor of DTS - I just mentioned
something I was surprised to see when looking at the spec for unrelated
reasons.



I am all in favour to find  some  more or less  agreed  solution,
 but what you describe would be  still a standard extension.

But if we are at this: I will post some proposal how to define some
Ambisonics based real-world standard (CE standard) during the next
days. (This  one will use more or less existing components.) In fact
there have been some private discussions, and I believe there have
been some results which might prove useful.


Of course a bespoke and open/free standard is going to be better



Using (currently) BD doesn't help quite a lot. You can use the
channels for B format/HOA channels (WXYZ etc.), but you can't feed
this into a normal decoder. (You would also need some flags to
bypass. I fear even this is not very well defined in HDMI etc. For
example, you can transmit/bypass TrueHD/DTS HD MA in a bitstream to
a receiver, but not WXYZ. At least not yet. The mixing stage you
have described is actually in  the BD player, but then the receiver
will expect a HDMI PCM stream? Most probably...)


The DTS spec is referring to DTS-HD - so the player AIUI could just pass
the whole stream to a decoder which if ambisonic aware could get the
wxyz from the stream and if not ignore it.


Personally, I also believe that some disc based solution  should 
exist, but this is more the 2nd step. IMHO, you would define the
Ambisonics CE standard, and then the distribution via file formats,
 on-disc etc. (You would also define typical output configurations,
 like 5.1/6.1/7.1, hexagon, octagon, binaural.)

If you don't do this, you are getting stuck with FOA. More important
(if you don't care), some backward-compatible hack extension won't
permit to progress to higher orders in the future, say SOA/TOA.


Maybe, but DTS-HD and DTD_MA are by your definition hack extensions
themselves - again I don't in any way favor DTS over something better
and certainly don't even understand the spec well enough from just
skimming it to be sure of anything, but the there is mention of up to 32
channels in there somewhere.

FWIW it is freely available - just google ETSI TS 102 114 V1.4.1 pdf



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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Augustine Leudar wrote:

Seeing as there is a dearth of 8 channel players - I was thinking I could
just use a 7.1 DVD on a loop for an eight channel sound installation - as
long as I can send a seperate audio signal to each of the 8 (the LF sends a
full range signal too) there shouldnt be a problem. I encoded 5.1 DVD ages
ago  and I vaguely remember I needed several programs, one for encoding
AC3, one for authoring etc etc - does anyone know  programs would I need to
encode a 7.1 DVD ?


According to wikipedia DVD-Audio only does 5.1 - it could be wrong of 
course.


The DVD-Video page claims 8ch PCM support is in the spec, but says it's 
not well supported by players.


AFAIK in practice 7.1 is a bluray thing.

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Re: [Sursound] status of ambisonia.com?

[Andy Furniss]

Paul Hodges wrote:

--On 16 December 2013 16:14 +0100 Jörn Nettingsmeier
netti...@stackingdwarves.net wrote:


what's the status of ambisonia.com? it appears that while the site is
distributing torrent metadata, nobody is actually seeding the files.


I agree that some seem unseeded and it was quite a while ago I tried - I 
did manage to get some and am still seeding these and some are being 
leeched occasionally.




I am currently seeding my files; the rest are available on my torrent
server, but I need to fix the seeding which broke when I had to
re-install the software - I guess this is a prompt to get on with it!

There are very few downloads, so I'm not sure if the torrents from
ambisonia.com are linking with my server correctly, or whether it's
just that no one knows it's there; but a few of my files get downloaded
somewhat regularly (Duruflé's Requiem and the excerpts from Purcell's
King Arthur are the top choices).

My files from Ambisonia (and John Leonard's, with permission) are also
available from a very fast download site through links from my webpages
at http://www.ambisonic.info/audio.html.


Thanks for that site, coincidently I was actually downloading some 
earlier today and found a dead link/mistake -


Farmer - A Little Pretty Bonny Lass AMB DTS

The AMB link points to the DTS dir and so doesn't work.

https://ambisonic-files.s3.amazonaws.com/DTS/pwh-Farmer-ALittlePrettyBonnyLass.amb






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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Stefan Schreiber wrote:

Andy Furniss wrote:



According to wikipedia DVD-Audio only does 5.1 - it could be wrong
of course.

The DVD-Video page claims 8ch PCM support is in the spec, but says
 it's not well supported by players.



Starts with the interface problems. Today you would use HDMI and USB
for this. Not in any DVD specs...


Yea, though it does hold out some hope that a blu-ray player that does
dvds may support it and send 8xPCM over hdmi to a receiver.


AFAIK in practice 7.1 is a bluray thing.



Or you define a file format. (AAC and FLAC actually would support
7.1. This doesn't mean people are using such an option yet.


I read the OPs post as meaning he wanted to use a consumer player - if a
computer is an option then problem solved :-)



P.S.: BDs would support Dolby/DTS 7.1, which means discrete 7.1.
Certainly not anything like Ambisonics, so you would need in any case
a new format definition.


I think that PCM if possible would be the answer - I know there are open
encoders for the core AC3/DTS 5.1 codecs, but not the TrueHD/MA that 7.1
blu-ray use, I guess certified commercial encoders are not cheap.

Having had a look at the DTS spec (ETSI TS 102 114 v1.4.1) recently,
though there are many possible channel layouts in there, it may be that
blu-ray players only do 7.1 and in that case the .1 really is only for
LFE - it is not stored/decoded as a full range channel. I don't know
about TrueHD - the spec seems to be secret.

Nothing to do with this thread and I am not saying that any players use
it, but I did see the words Ambisonic and WXYZ in the spec, so there is
some provision for carrying and flagging as special b-format in a DTS
extension stream.




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Re: [Sursound] Encoding a 7.1 audio DVD ?

[Andy Furniss]

Bearcat M. Şándor wrote:

If the bluray spec only supports up to 7.1, why are we seeing receivers
with 11.2 outputs?  Are those just matrixed channels with some reverb
thrown in? I'll be thinking about this for my ambisonics setup.


When I said

Having had a look at the DTS spec (ETSI TS 102 114 v1.4.1) recently,
though there are many possible channel layouts in there, it may be that
blu-ray players only do 7.1 and in that case the .1 really is only for
LFE - it is not stored/decoded as a full range channel. I don't know
about TrueHD - the spec seems to be secret.


I was really meaning what every player/decoder should support as a 
minimum - to be blu-ray compatible.


I think if the player passes on the whole DTS/Dolby stream to a more 
advanced decoder then it's quite possible that if present 11.2 real 
channels could be on the disk.


Looking at wikipedia again I see it doesn't even say 7.1 is max for 
TrueHD and DTS-MA, it says 8ch so I guess I was wrong in saying 7.1 was 
max and what I wrote about LFE is wrong as long the channels are all 
flagged a full range. OP may still want to use PCM of course as there is 
the issue of encoders.


I also realise I misread/skimmed the first line in this thread and took 
dearth of 8 channel players  to mean there are plenty of 8ch players - 
strange

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Re: [Sursound] Linux ambdec b-format channel order

[Andy Furniss]

Marc Lavallée wrote:


In my experience, the order of channels in jack are stable enough to
use this simple channels remapping trick:

mplayer -channels 4 -af channels=4:4:0:0:1:3:2:1:3:2 -ao
jack:port=ambdec AJH_eight-positions.amb


FWIW, while looking at mplayer code it occured to me that -

1. My suggestion in my last post to make mplayer sort by name is silly
and would be wrong for higher orders where channel number matters (I
can't reply to my own post because of the stupid way gmail works).

2. Rather than taking time to extend mplayer to take input names, though
useful, I could just use qjackctl patch bay. With that I can set up
connections how I want and handle all the variety of higher orders
(though I did notice an 8 channel limit in the mplayer code - easily
changed).

So if I did need to use mplayer for many different types of .amb I would
just set up different mappings in patchbay, load my settings on starting
jack and then with recent (older doesn't have noconnect) mplayer do
something like -

mplayer -channels 4 -ao jack:name=MP4:noconnect some-bformat.amb

mplayer -channels 7 -ao jack:name=MP7:noconnect some-3rdhoriz.amb

etc. MP4 and MP7 being names used in my patchbay config the right
connections will be made when mplayer starts.


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Re: [Sursound] Linux ambdec b-format channel order

[Andy Furniss]

Fons Adriaensen wrote:

On Thu, Nov 14, 2013 at 12:00:10AM +, Andy Furniss wrote:


I am not experienced with ambdec, but while trying something today
I noticed something confusing when feeding direct with mplayer -

mplayer -channels 4 -ao jack:port=ambdec pwh-VoiCE-Round.amb

The channels are crossed as in the screenshot. This doesn't happen
if I were to use the same mplayer command to a different sink.

I suppose it does match the order on the config screen, but not the
jack input names.

http://imageshack.com/a/img203/1880/1xzw.png


Mplayer's Jack support is broken, it doesn't allow you to specify the
ports to connect to, only the application name. And Jack itself has
no notion of any 'order' of the ports, whatever order there is has to
be inferred from the port names. Mplayer doesn't do this, and the
result is more or less random connections.


OK, so do you mean that in the absence of being able to specify names
(or even if implemented to save user typing) that just sorting the
inputs by name would be the correct thing to do?



In this case the sequence matches the one in the config window,
probably because Ambdec creates its ports in that order, and when
asked for Ambdec's ports, Jack happens to list them in the same order
again. But that is not documented in Jack's API, so you can't rely on
it.

The only solution is to fix Mplayer.



Thanks for the info, and thanks also to Marc and Michael for their posts.

It's going to bug me till I find it, but I think that a link recently on
this list suggested the very mplayer command that I was testing.

@Fons - have you built ambdec with a recent linux set up?

Though I am not a very typical/good test case as I use LFS (actually
CLFS on this PC) I had to add -lpthreads to the makefile as ld (binutils
from last year) threw an error. It was self explanatory and searching it
seems that the default behavior has changed so you have to explicitly
specify libs that previously would have been included if referenced by
other libs that were specified. As I don't use distros I don't know if
they work around so have no idea if any one else will see this.

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[Sursound] Linux ambdec b-format channel order

[Andy Furniss]

I am not experienced with ambdec, but while trying something today I
noticed something confusing when feeding direct with mplayer -

mplayer -channels 4 -ao jack:port=ambdec pwh-VoiCE-Round.amb

The channels are crossed as in the screenshot. This doesn't happen if I
were to use the same mplayer command to a different sink.

I suppose it does match the order on the config screen, but not the jack
input names.

http://imageshack.com/a/img203/1880/1xzw.png


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Re: [Sursound] B-Format test signals 5.1

[Andy Furniss]

Eero Aro wrote:

I am afraid I binned the discussion thread about the test signals.

However, if anybody's interested, I made quickly a couple of files.
I panned a pink noise pulse train into the speaker directions
of ITU775, the 5.1 layout.

The files are 44.1 kHz/16 bit four channel wav:s. The Z channel
has no content, as it is horizontal only.

No such thing as LFE in B-Format.

As others already said, the correct placing of 5.1 speakers is
better to do by measuring. But here's some noise from those
directions:

https://dl.dropboxusercontent.com/u/22100835/CF_0_dgr.wav
https://dl.dropboxusercontent.com/u/22100835/LF_315_dgr.wav
https://dl.dropboxusercontent.com/u/22100835/LB_250_dgr.wav
https://dl.dropboxusercontent.com/u/22100835/RB_110_dgr.wav
https://dl.dropboxusercontent.com/u/22100835/RF_45_dgr.wav


ITU775 has FL/R at 30 degrees not 45.


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Re: [Sursound] Flying Scotsman programme

[Andy Furniss]

Dave Malham wrote:

I was watching the BBC programme on the Flying Scotsman the other nigh when
I spotted something at just over 37'40 in - in a film from between 1963
and 1968, the guy recording an interview with Alan Pegler, the then owner
of the famous loco, was using a dummy head - or at least a head sized
sphere with mics either side. Does anyone know anything about this, who it
was and so on? The programme is still available on iPlayer so at least the
Brits (and those of you with IP faking) should be able to pick this up.


If no one here knows, it may be worth asking on usenet, there are a few 
retired BBC people who lurk on uk.tech.broadcast.



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Re: [Sursound] Very inexpensive surround speakers

[Andy Furniss]

Bearcat M. Şándor wrote:

Since we're talking about this, I've had my eye on Emotiva Stealth
8s http://www.emotivapro.com/products/stealth-8 . I assume they (or
the upcoming Stealth 88 which adds an additional 8 driver) would do
just fine for an ambisonic installation?

Now i just need to be able to afford 18 of those things..


:-) well if I were a millionaire the Genelec HT series look very nice.

More seriously, I did some time ago do a bit of searching about active
speakers just for home use. One possible issue could be back ported vs
front ported, your link looks like back ported, which could (so I have
read - no experience) have bass issues if placed close to walls.

Another thing I read albeit by an amateur reviewer about a specific
studio monitor, is that he said they wouldn't be much use for home hifi
use as the sweet spot was very narrow. Just one opinion about one
speaker of course, but something worth looking into. I have also read
the studio monitors are fine for home hifi use.
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Re: [Sursound] Volume question WRT 7.1 sound recorded at listening position.

[Andy Furniss]

Aaron Heller wrote:

On Mon, Sep 23, 2013 at 3:55 PM, Andy Furniss adf.li...@gmail.com wrote:






I don't quite understand the in phase though, are you saying that they
artificially adjust phase for the same sound that comes out of more than
one speaker to affect the mixdown?



The Recording Academy recommendations for surround sound say (sec 4.3)

One potential problem that can arise from routing a signal into two or more
speakers is the danger of increased, and increasingly complex, comb
filtering. This problem multiplies as more speakers are engaged and can
become critical if downmixing is ever employed by the playback system.
Therefore, many experienced surround mixers selectively turn off channels
when bringing a sound inside the surround bubble or when dynamically
panning a sound from one area in the surround space to another.

It is recommended that whenever signal is placed into three, four, or five
speakers, it be decorrelated.



http://www2.grammy.com/PDFs/Recording_Academy/Producers_And_Engineers/SurroundRecommendations.pdf


Ahh, thanks for that.

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Re: [Sursound] Volume question WRT 7.1 sound recorded at listening position. (dw)

[Andy Furniss]

Andrew Levine wrote:

See Bob Katz' K-20:

http://www.digido.com/how-to-make-better-recordings-part-2.html


Thanks for the link, looks interesting, though I haven't had time to 
read properly yet.


Accepting the above may give insight into my query, just to be clear I 
am not a producer in any way - just thinking about reproduction of 
others work whether that is good, bad or inbetween.


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Re: [Sursound] Volume question WRT 7.1 sound recorded at listening position.

[Andy Furniss]

dw wrote:

On 22/09/2013 12:51, Andy Furniss wrote:

Hi

I do not have a 7.1 sound system so can't actually test this, also
as may become apparent I don't know much about sound :-)

I would be grateful if someone could correct/confirm the
following.

If I were to mix down a digital 7ch to mono I have to reduce by 1/7
 amplitude to prevent clipping, so one channel will be -16.9dBfs.



This assumes that all 7 channels are driven with the same maximum
level, in phase, and that this would get past the mastering stage,,
I'm sure they bear in mind the possibility of stereo, or mono
mixdown.


Yea I guess all channels full at once only really happens on the front
three in practice, mine was just an example to make the figures as
different as possible.

I don't quite understand the in phase though, are you saying that they
artificially adjust phase for the same sound that comes out of more than
one speaker to affect the mixdown?

Level wise, I don't think my method is any different from and software
decoder, except of course various channels get different weighting
before normalisation.
Peaking at 0dbfs is evident on a couple of film soundtracks, disk and
BBC broadcast I've just looked at (without compressing using the DRC
metadata, of course).

Of course in practice consumer 7.1 currently means TrueHD or DTS MA
which seems to be mixed up rather than down - so there is a studio
stereo mix there in the stream to be decoded directly.


If I play the track over one speaker the volume difference between
one and all channels will be 16.9dB.

If I were to measure the volume at listening position (assuming
anechoic and equal speaker distance) with a real 7 speaker setup
then the volume difference, because the speakers are not close,
would add up using power not amplitude so the difference
heard/measured between 1 and 7 at full power would only be 8.45dB,
so there is quite a large dynamic range discrepency?


You would add powers if the 7 channels had random relative phase
(which is less likely with anechoic and equal speaker distance


Ok, so would infinite distance planewaves playing the same sound add as
if the speakers were coupling (I thought you needed  1/2 wavelength for
this) to gain twice as much as power alone even though they are not in
the same direction? Or is this a just a mathematical perfect position
effect that wouldn't really happen in a real 7.1 setup even if the
speakers were playing the same sound?


The real reason for this question is more to do with simulation
than real life, so perhaps that will make a difference - if the
speakers are infinitely far producing planewaves and a soundfield
is at listening position would that change anything for what levels
the virtual omni would hear.


Sorry for all the questions - I would just like to understand if I use
supercolliders 7.0 ambisonic encoder so I can then do either personal
hrtf or uhj stereo decode, whether the simulation it uses is correct in
the sense of the levels I would get with a real soundfield measuring
real speakers.

Thanks.

Andy.



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Re: [Sursound] Volume question WRT 7.1 sound recorded at listening position. (dw)

[Andy Furniss]

Ken Landers wrote:

While -16.9 might keep you safe, a better option might be -20 dBFS.
Gives some headroom in case you need it.  Also, many consumer
playback devices may not handle full scale output.


Interesting, I am not a producer of anything as such, but do see that a
lot of digital music and some films push 0dBfs (though at least the
films aren't anywhere near RMS)

Do you mean that consumer DACs can't handle it properly, or that the
analogue side doesn't being driven full voltage?

Andy.
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[Sursound] Volume question WRT 7.1 sound recorded at listening position.

[Andy Furniss]

Hi

I do not have a 7.1 sound system so can't actually test this, also as 
may become apparent I don't know much about sound :-)


I would be grateful if someone could correct/confirm the following.

If I were to mix down a digital 7ch to mono I have to reduce by 1/7 
amplitude to prevent clipping, so one channel will be -16.9dBfs.


If I play the track over one speaker the volume difference between one 
and all channels will be 16.9dB.


If I were to measure the volume at listening position (assuming anechoic 
and equal speaker distance) with a real 7 speaker setup then the volume 
difference, because the speakers are not close, would add up using power 
not amplitude so the difference heard/measured between 1 and 7 at full 
power would only be 8.45dB, so there is quite a large dynamic range 
discrepency?


The real reason for this question is more to do with simulation than 
real life, so perhaps that will make a difference - if the speakers are 
infinitely far producing planewaves and a soundfield is at listening 
position would that change anything for what levels the virtual omni 
would hear.


TIA

Andy.


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