15 Dec
2011
15 Dec
'11
2:59 a.m.
Hi,
(TL;DR: skip to "The main question...")
I'm planning a hobby project to build an all-digital surround/hi-fi
system. Regular surround/hi-fi systems convert digital input sources
to analog in the surround processor/preamp, but I want to keep the
signal in the digital domain as far as possible (by using all-digital
class D power amps, aka. "PowerDACs" [*] that convert a PCM signal to
an amplified PWM signal that is feeds the speaker directly through an
LPF. The heart of this system is an all-digital surround
processor/preamp that should do the following:
1. Switch between multiple digital inputs. (typically from HDMI or SPDIF)
2. Decode and/or upsample the input (if necessary) into a suitable
format, typically 96kHz or 192kHz 24-bit PCM in 8 channels (for a 7.1
system).
3. Perform signal processing on the PCM signal, like room correction
and volume control.
4. Output each of the 8 channels on separate digital outputs suitable
for connection to the power amp associated with each speaker.
AFAIK there are currently (very few or) zero preamps for sale that
meet the above requirements. Requirement #4 seems especially hard to
satisfy.
I'm therefore planning to build my own preamp based on the following components:
- A HDMI switch with audio split (SPDIF) and RS-232 control.
- A Linux computer that controls the HDMI switch, and contains the
audio processing software (e.g. a gstreamer pipeline doing the
decoding and manipulation).
- A suitable audio interface with at least 8 digital outputs.
I believe this rig should be able to meet the above requirements.
The main question at this point is what audio interface I should
choose for the rig. Some points to consider:
- Must work very well with Linux; bonus points if the vendor is Linux-friendly.
- Must be able to output 8 channels of 96kHz/24bit PCM, preferably
using AES/EBU or SPDIF (as that's what most digital power amps seem to
accept as input).
- USB, Firewire, PCI or PCI express? Which is more stable/supported?
- Bonus points if it also has 8 analog outputs, as I want to
prototype this on an analog power amp before investing in digital
power amps.
- Since this is still in a proof-of-concept phase, I'd like to spend
not more than about $1000 on the interface.
So far I've been browsing interfaces like:
- MOTU 828mk3 (capable, but seems to be poorly supported by Linux)
- RME HDSPe AES (very capable, lacks analog outputs, a bit pricey)
- RME Multiface II + HDSPe interface card (has analog outputs, not
sure if 1 SPDIF plus 1 ADAT can give me 8 channels of digital output
in 96/24, pricey)
- Focusrite Saffire PRO 40 (seems similar to RME Multiface II, but
cheaper. Unsure how well it is supported by Linux)
What other audio interfaces should I check out? Are there other things
I should keep in mind when picking an audio interface?
Have fun! :)
...Johan
[*]: Examples of all-digital power amps include the NAD Direct Digital
amps (M2 and C390DD), the Tact True Digital Amplification series, etc.
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
18 Dec
18 Dec
12:30 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On 12/15/2011 02:59 AM, Johan Herland wrote:
2. Decode and/or upsample the input (if necessary)
into a suitable
format, typically 96kHz or 192kHz 24-bit PCM in 8 channels (for a 7.1
system).
out of curiousity, what is the benefit of upsampling? is there something
peculiar about power dacs that would make it useful or mandatory?
- A HDMI switch with audio split (SPDIF) and RS-232
control.
hdmi to spdif means that you either get only two channels, or lossy
encoding. plus you will need to think about the clocking structure.
usually this will mean that your audio card will have to slave itself to
the incoming hdmi/spdif/whatever.
- A suitable audio interface with at least 8 digital
outputs.
tough one. there are a number of cheap options with ADAT, but you will
need two at 96k due to s/muxing, and four at 192k. for the latter, the
only option i know of is the rme raydat.
iirc, rme also has a card with native aes/ebu outs, and it should be
working ootb with the normal driver, but you may well find yourself to
be the only user ever of this card under linux. so be prepared to get in
touch with adi and get your kernel compilation toolchain ready :-D
the vendor is moderately linux-friendly, at least they are providing
specs and not actively making our lives harder than necessary. plus the
gear tends to be reliable and flexible.
- Since this is still in a proof-of-concept phase,
I'd like to spend
not more than about $1000 on the interface.
So far I've been browsing interfaces like:
- MOTU 828mk3 (capable, but seems to be poorly supported by Linux)
don't go for firewire. it's great for mobile use, but a) it's on the way
out, b) it requires a very carefully tuned low-latency system, and c) it
totally does not work with frequency scaling and you will burn a quarter
of the available cpu cycles just to get the audio data in and out.
pretty deadly for an embedded and possibly passively cooled system.
- Focusrite Saffire PRO 40 (seems similar to RME
Multiface II, but
cheaper. Unsure how well it is supported by Linux)
it is well supported, but see above.
What other audio interfaces should I check out? Are
there other things
I should keep in mind when picking an audio interface?
unfortunately, the only answer for truly professional audio hardware
under linux is rme. not that their gear is bad, but it sure is
expensive, and i wonder what we're going to do if they ever turn evil...
time for those open hardware projects...
2:03 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On Sun, Dec 18, 2011 at 12:30:18PM +0100, Jörn Nettingsmeier wrote:
'Expensive' is relative in this case. RME's HDSP AES-32 has 8 x AES
in (i.e. 16 ch) and 8 x AES out. 730 Euro at Thomann. That is peanuts
compared to the price of 8 channels of digital amps. And I'm not even
considering the price of the speakers that wouldn't make using such
expensive amps a joke.
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
On 12/15/2011 02:59 AM, Johan Herland wrote:
What other
audio interfaces should I check out? Are there other things
I should keep in mind when picking an audio interface?
unfortunately, the only answer for truly professional audio hardware
under linux is rme. not that their gear is bad, but it sure is
expensive, and i wonder what we're going to do if they ever turn evil...
time for those open hardware projects...
3:28 p.m.
2011/12/18 Jörn Nettingsmeier <nettings(a)folkwang-hochschule.de>de>:
True. Ideally, I'd be able to run the HDMI directly into the PC, and
pick up the sound signal directly from the HDMI stream. However, I
don't know of _any_ consumer-available HDMI input card, and even if
they were available, I probably wouldn't be able to decode an
HDCP-encrypted stream (or maybe the audio signal isn't encrypted? I
don't really know HDMI...).
Meridian has an HDMI switch <URL:
http://www.meridian-audio.com/the-collection/accessories/hd621-hdmi-audio-p…
I've been doing some reading on 96kHz vs. 192kHz, and most people seem
to think that there is no audible difference, and that it's a lot of
marketing hype, so until I'm convinced otherwise, I'm not going to
spend extra money on 192kHz equipment. So what other equipment is
available with 2 ADAT outputs?
Thanks! That's exactly the kind of advice I was looking for. I had a
feeling that it'd be simpler with PCI(e) than with firewire, and now
I'm convinced. :-)
Indeed. I've seen some DIY projects that are interesting to hobbyists
[*]. Hopefully some of them will "grow up" to become interesting to
professionals as well.
Thanks a lot for your input! :)
...Johan
[*]: Some of the interesting DIY audio interfaces I've stumbled across:
http://www.minidsp.com/products/ministreamer
http://www.qnktc.com/ab_11/
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
On 12/15/2011 02:59 AM, Johan Herland wrote:
It's not really a requirement, but I thought it'd make sense for two reasons:
1. Making sure there's enough resolution to do room correction, volume
control, etc. without losing details.
2. Forcing everything into a common format to make thing
simpler/uniform for the remainder of the pipeline and the dacs/amps.
2. Decode and/or upsample the input (if
necessary) into a suitable
format, typically 96kHz or 192kHz 24-bit PCM in 8 channels (for a 7.1
system).
out of curiousity, what is the benefit of upsampling? is there something
peculiar about power dacs that would make it useful or mandatory? - A HDMI
switch with audio split (SPDIF) and RS-232 control.
hdmi to spdif means that you either get only two channels, or lossy
encoding. that I believe is able split of the HDMI audio without
signal loss,
but the audio is output using either MMHR (over RJ45) or SmartLink
(over 4xRCA), which are (AFAIK) both encrypted and can only be
connected to other Meridian gear. (Also, I don't have $3000 to spend
on an HDMI switch.) I don't know of any other commercially available
physical connection that can carry the full information of the HDMI
audio signal.
plus you will need to think about the clocking
structure. usually
this will mean that your audio card will have to slave itself to the
incoming hdmi/spdif/whatever.
Hmm. I don't really know much about clocking. How would you organize
the system to minimize clocking issues, and maximize fidelity?
- A suitable
audio interface with at least 8 digital outputs.
tough one. there are a number of cheap options with ADAT, but you will need
two at 96k due to s/muxing, and four at 192k. for the latter, the only
option i know of is the rme raydat. iirc, rme also has a card with native aes/ebu outs,
and it should be working
ootb with the normal driver, but you may well find yourself to be the only
user ever of this card under linux. so be prepared to get in touch with adi
and get your kernel compilation toolchain ready :-D
Yes, the RME HDSPe RayDAT and HDSPe AES are the two cards I've been
looking at the most. However, the AES seems to be ~60% more expensive
than the RayDAT, so I'm leaning towards the RayDAT. It really comes
down to how cheaply I can convert from ADAT to either AES/EBU or SPDIF
(either of which is what most digital amps will take as input).
When it comes to kernel compilation, I'm not too scared, as my
background is in Linux and software. I'm more afraid to screw up the
audio hardware part of this project... :-)
the vendor is moderately linux-friendly, at least they
are providing specs
and not actively making our lives harder than necessary. plus the gear tends
to be reliable and flexible.
That's my impression of RME too. I'll gladly spend some extra bucks on
equipment that will cause less headache in the long run.
- MOTU 828mk3
(capable, but seems to be poorly supported by Linux)
don't go for firewire. it's great for mobile use, but a) it's on the way
out, b) it requires a very carefully tuned low-latency system, and c) it
totally does not work with frequency scaling and you will burn a quarter of
the available cpu cycles just to get the audio data in and out. pretty
deadly for an embedded and possibly passively cooled system. - Focusrite
Saffire PRO 40 (seems similar to RME Multiface II, but
cheaper. Unsure how well it is supported by Linux)
it is well supported, but see above.
What other audio interfaces should I check out?
Are there other things
I should keep in mind when picking an audio interface?
unfortunately, the only answer for truly professional audio hardware under
linux is rme. not that their gear is bad, but it sure is expensive, and i
wonder what we're going to do if they ever turn evil... time for those open
hardware projects...
19 Dec
19 Dec
11:14 a.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On 12/18/2011 03:28 PM, Johan Herland wrote:
with these very simple and mostly linear operations, you will gain
nothing at all from upsampling. certainly nothing that would justify
doubling the cpu and throughput expense of your system.
preventing the full fidelity of material produced at 96k is another
issue, but such material is rare outside of studio workflows.
<snip>
all digital gear in the signal chain must run from the same clock,
unless you insert a sample-rate converter.
in a studio, there is a common wordclock which is distributed to all
players, processors, and output DACs.
consumer equipment will generally not be able to deal with external
clocking, so your source will have to be the clock master. the clock is
then distributed embedded in the signal - hdmi audio, spdif and aes/ebu
are all self-clocking.
but this also means that if you switch from blue-ray player a to
blue-ray player b, your sound card must change its clocking source from
input a to input b. which might or might not cause an audible click or
thump.
as to "maximizing fidelity", this is digital pcm. it either works
perfectly or not at all. the only way to slightly degrade the signal is
to have a lossy codec in between (such as ac3 or dts), or when you're
forced to insert a SRC. but the latter should be pretty close to perfect
if it's a good one. no longer bit-transparent, obviously.
I've been doing some reading on 96kHz vs. 192kHz, and most people seem
to think that there is no audible difference, and that it's a lot of
marketing hype, so until I'm convinced otherwise, I'm not going to
spend extra money on 192kHz equipment. So what other equipment is
available with 2 ADAT outputs?
i tend to agree on the 192k issue...
iirc, there is the rme hdsp 9636 which should fit your needs (2 adats).
or a hdsp hammerfall with the digiface break-out (3 adats). there might
be cheaper alternatives from other manufacturers, maybe others will
comment. m-audio used to be a linux-friendly choice...
<snip>
out of
curiousity, what is the benefit of upsampling? is there something
peculiar about power dacs that would make it useful or mandatory?
It's not really a requirement, but I thought it'd make sense for two reasons:
1. Making sure there's enough resolution to do room correction, volume
control, etc. without losing details. plus you will
need to think about the clocking structure. usually
this will mean that your audio card will have to slave itself to the
incoming hdmi/spdif/whatever.
Hmm. I don't really know much about clocking. How would you organize
the system to minimize clocking issues, and maximize fidelity? - A suitable audio interface with at least 8 digital
outputs.
tough one. there are a number of cheap options with ADAT, but you will need
two at 96k due to s/muxing, and four at 192k. for the latter, the only
option i know of is the rme raydat. It really comes
down to how cheaply I can convert from ADAT to either AES/EBU or SPDIF
(either of which is what most digital amps will take as input).
haven't done a proper survey, but i'd use RME gear for this job. which
means the combination of adat card plus external AES/EBU bridge will be
more expensive and less elegant than the hdsp/aes card.
When it comes to kernel compilation, I'm not too
scared, as my
background is in Linux and software.
good :)
have fun,
jörn
3:21 p.m.
2011/12/19 Jörn Nettingsmeier <nettings(a)folkwang-hochschule.de>de>:
all digital gear in the signal chain must run from the same clock, unless
you insert a sample-rate converter.
in a studio, there is a common wordclock which is distributed to all
players, processors, and output DACs.
consumer equipment will generally not be able to deal with external
clocking, so your source will have to be the clock master. the clock is then
distributed embedded in the signal - hdmi audio, spdif and aes/ebu are all
self-clocking.
I don't see how the _entire_ pipeline can run off a single external
clock. Say, I have a simple setup: A CD player connected to the audio
PC doing a simple digital volume control, and then outputting the
signal to a (Power)DAC. Obviously there will be some (small but still
non-zero) delay between the audio PCs input signal and its output
signal. How can both signals run from the same clock?
Or does the audio PC sync the output to a _later_ pulse form the clock
generator (i.e. all that matters is that the signal is synced with _a_
clock pulse, not necessarily _the_same_ clock pulse with which it was
originally transmitted)? But if so, couldn't I have one clock between
the CD player and audio PC, and a different clock between the PS and
the DAC?
And is self-clocking somehow inferior to an external clock? If an
external clock is better, how come ethernet/USB/firewire and all other
digital communication protocols run without external clocks? Sorry for
being thick, but I haven't worked with these things before...
Yeah. I guess I'll have to wait and see what digital amps I end up
with. The high-end scary expensive ones take both SPDIF coax/toslink
and AES/EBU, but there's a growing breed of cheaper DIY-inspired
digital amps (e.g. the miniDIGI/miniAMP combo <URL:
http://www.minidsp.com/products/minidspkits/minidigi >, or the SUMOH
amps <URL: http://www.sumoh.com/index-2.html >) that only take SPDIF
coax/toslink input. I don't know whether it's cheaper/easier to
convert AES/EBU or ADAT to SPDIF coax/toslink...
Then again, there are some digital amps that take USB input (e.g. the
miniSTREAMER/miniAMP combo <URL:
http://www.minidsp.com/products/ministreamer >, or the HFX RipAMP
2.1/100 <URL:
http://www.hfx.at/index.php?option=com_content&view=article&id=376&…
On 12/18/2011 03:28 PM, Johan Herland wrote:
with these very simple and mostly linear operations, you will gain nothing
at all from upsampling. certainly nothing that would justify doubling the
cpu and throughput expense of your system.
Yeah. After some thinking it's now obvious to me why I shouldn't
upsample (as long as the rest of the pipeline can do the sampling
rates I throw at it). However, I do want to use as many bits as
possible in processing, to make sure that I don't loose detail by
shifting bits into oblivion.
out of
curiousity, what is the benefit of upsampling? is there something
peculiar about power dacs that would make it useful or mandatory?
It's not really a requirement, but I thought it'd make sense for two
reasons:
1. Making sure there's enough resolution to do room correction, volume
control, etc. without losing details. preventing the full fidelity of material produced at
96k is another issue,
but such material is rare outside of studio workflows.
Actually, 24/96 and even 24/192 is becoming usual if you get your
music from sites like hdtracks.com or gubemusic.com.
plus you will need to think about the clocking
structure. usually
this will mean that your audio card will have to slave itself to the
incoming hdmi/spdif/whatever.
Hmm. I don't really know much about clocking. How would you organize
the system to minimize clocking issues, and maximize fidelity? but this also means that if you switch from blue-ray
player a to blue-ray
player b, your sound card must change its clocking source from input a to
input b. which might or might not cause an audible click or thump.
Can't you "fix" this by quickly muting the signal, then switch
sources, and unmute after you've "locked" onto the new signal? If the
switch is software-controlled (via RS-232) wouldn't that be fairly
simple to do?
as to "maximizing fidelity", this is digital
pcm. it either works perfectly
or not at all. the only way to slightly degrade the signal is to have a
lossy codec in between (such as ac3 or dts), or when you're forced to insert
a SRC. but the latter should be pretty close to perfect if it's a good one.
no longer bit-transparent, obviously.
Ok, so if I save money by not using external clocking, and using
consumer-grade equipment (but stick to lossless PCM), I will either
get a perfect signal, or get no signal at all. And if I get no signal
at all, it's likely to be because my cheap devices don't agree on
where the clock pulse is. Correct?
It really
comes
down to how cheaply I can convert from ADAT to either AES/EBU or SPDIF
(either of which is what most digital amps will take as input).
haven't done a proper survey, but i'd use RME gear for this job. which means
the combination of adat card plus external AES/EBU bridge will be more
expensive and less elegant than the hdsp/aes card. ), which would bypass the entire audio interface
altogether. However,
I find it hard to believe that you could plug 4-8 of these into
a
single computer without maxing out the CPU or USB controller...
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
4:20 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On Mon, Dec 19, 2011 at 03:21:19PM +0100, Johan Herland wrote:
Yes, you could mute the amps for a short time. Anyway, when the PC
input clock is switched, the output clock will follow, your amps
will detect that and probably mute automatically until they are
resynced. That is at least what I'd expect from something costing
$6000.
Yeah. After some thinking it's now obvious to me
why I shouldn't
upsample (as long as the rest of the pipeline can do the sampling
rates I throw at it). However, I do want to use as many bits as
possible in processing, to make sure that I don't loose detail by
shifting bits into oblivion.
This is a non-problem. 24 bits gives you a S/N ratio better than
140 dB. Now assume that you adjust the volume of you power amps
such that -20 dB digital corresponds to +100 dB SPL. This is pretty
loud and you still have 20 dB headroom for peaks. That means that
the digital noise floor of your system corresponds to -20 dB SPL,
that is 20 dB below the hearing threshold and at least 40 dB
below the ambient noise level of any normal living room. So
won't hear it. Even if a low-level signal would use only 8 bits
or so, you will not hear any distortion as a result of that.
The thing that doesn't work is first reduce a signal to e.g. 8
bits and then amplify it again digitally to make it play louder.
But that won't happen unless you really use the volume controls
in the wrong way. If the amp's volume control is fixed (e.g. as
described above) this problem does not exist.
I don't see how the _entire_ pipeline can run off
a single external
clock. Say, I have a simple setup: A CD player connected to the audio
PC doing a simple digital volume control, and then outputting the
signal to a (Power)DAC. Obviously there will be some (small but still
non-zero) delay between the audio PCs input signal and its output
signal. How can both signals run from the same clock?
The will simply be a fixed number of samples delay between input
and output. The thing that matters is that the sample frequencies
are exactly the same. If they are not you have to resample.
Or does the audio PC sync the output to a _later_
pulse form the clock
generator (i.e. all that matters is that the signal is synced with _a_
clock pulse, not necessarily _the_same_ clock pulse with which it was
originally transmitted)?
Indeed, that is what happens.
But if so, couldn't I have one clock between the
CD player and audio
PC, and a different clock between the PS and the DAC?
(I assume you mean _PC_ and DAC, or PC and digital amp).
In that case the PC has to do sample rate conversion. It will also
considerably complicate your SW.
And is self-clocking somehow inferior to an external
clock? If an
external clock is better, how come ethernet/USB/firewire and all other
digital communication protocols run without external clocks? Sorry for
being thick, but I haven't worked with these things before...
In most digital formats the clock can be regenerated from the
digital signal. There's no quality loss involved in that - clock
jitter at a digital input doesn't matter as long as the clock is
good enough to recover the data without error. Professional audio
installations use a separate clock because 1) that is more reliable
when long cables are used, and 2), it makes the synchronisation
independent of the signal routing which in e.g. a studio isn't
just a simple linear chain as it would be in a domestic player
setup, and usually it isn't fixed either.
but this also
means that if you switch from blue-ray player a to blue-ray
player b, your sound card must change its clocking source from input a to
input b. which might or might not cause an audible click or thump.
Can't you "fix" this by quickly muting the signal, then switch
sources, and unmute after you've "locked" onto the new signal? If the
switch is software-controlled (via RS-232) wouldn't that be fairly
simple to do? as to
"maximizing fidelity", this is digital pcm. it either works perfectly
or not at all. the only way to slightly degrade the signal is to have a
lossy codec in between (such as ac3 or dts), or when you're forced to insert
a SRC. but the latter should be pretty close to perfect if it's a good one.
no longer bit-transparent, obviously.
Ok, so if I save money by not using external clocking, and using
consumer-grade equipment (but stick to lossless PCM), I will either
get a perfect signal, or get no signal at all. And if I get no signal
at all, it's likely to be because my cheap devices don't agree on
where the clock pulse is. Correct? I don't know whether it's cheaper/easier to
convert AES/EBU or ADAT to SPDIF coax/toslink...
AES/EBU and SPDIF are (almost) the same. The main differences
are in the electrical signal level and impedance. There are some
small coding differences as well, but in general an SPDIF input
will work with an AES signal, and an AES input *may* work with
an SPDIF signal provided the signal level is high enough, but
don't count on that.
Then again, there are some digital amps that take USB
input (e.g. the
miniSTREAMER/miniAMP combo <URL:
In that case the PC's output sample rate is set by the amplifier.
Which means you can connect only one such amplifier unless they
are synchronised with external clocks (which they probably can't
do). And even if you can get them synced the PC will still have
to sample rate conversion if it plays from an external source.
So I'd say 'don't go that way'.
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
5 p.m.
On Mon, Dec 19, 2011 at 16:20, Fons Adriaensen <fons(a)linuxaudio.org> wrote:
I see. And here using the same number of kHz on both sides of the PC
is not necessarily good enough; they actually have to be the very
_same_ clock, so that there's no chance of drifting.
Ok, I think I see now. Basically in order to prevent sample rate
conversion or extensive buffering (to compensate for clock drift), you
must make sure that the input and output runs off the very same clock
(with a whole constant number of clock cycles worth of audio being
processed at any time in the audio PC).
Understood. In my case, I'll have a fairly linear and fixed setup, and
my cable runs will not exceed 10-15 meters (which is towards the upper
end of what SPDIF can handle, IINM), so self-clocking should hopefully
be sufficient.
Yes, you could mute the amps for a short time. Anyway, when the PC
input clock is switched, the output clock will follow, your amps
will detect that and probably mute automatically until they are
resynced. That is at least what I'd expect from something costing
$6000.
Yeah, I was hoping to keep it below that. At least until I can prove
to myself that the potential audible improvement will be worth it. I'm
basically trying to keep the cost down while I experiment, and once I
get more familiar and comfortable with the system as a whole, I can
make a more qualified decision on where to put the bigger money.
Ok, so I might be able to convert from AES/EBU to SPDIF using a simple
(maybe even passive) circuit?
That seems like it'd be much cheaper than converting ADAT to SPDIF.
And it would even give me the option of upgrading to higher-end
AES/EBU amps along the way. Not to mention that AES/EBU will do longer
cable runs than SPDIF.
Thanks. That makes sense now that I have a grasp on the clocking business.
Thanks a lot for your help. I really appreciate it!
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
On Mon, Dec 19, 2011 at 03:21:19PM +0100, Johan
Herland wrote:
Thanks for quantifying this for me. Between the crazy audiophile
forums and my lack of experience in this field, it's not always easy
to keep track of how much is good enough, and what is just crazy
over-engineering. :)
Yeah. After some thinking it's now obvious to
me why I shouldn't
upsample (as long as the rest of the pipeline can do the sampling
rates I throw at it). However, I do want to use as many bits as
possible in processing, to make sure that I don't loose detail by
shifting bits into oblivion.
This is a non-problem. 24 bits gives you a S/N ratio better than
140 dB. Now assume that you adjust the volume of you power amps
such that -20 dB digital corresponds to +100 dB SPL. This is pretty
loud and you still have 20 dB headroom for peaks. That means that
the digital noise floor of your system corresponds to -20 dB SPL,
that is 20 dB below the hearing threshold and at least 40 dB
below the ambient noise level of any normal living room. So
won't hear it. Even if a low-level signal would use only 8 bits
or so, you will not hear any distortion as a result of that. I don't
see how the _entire_ pipeline can run off a single external
clock. Say, I have a simple setup: A CD player connected to the audio
PC doing a simple digital volume control, and then outputting the
signal to a (Power)DAC. Obviously there will be some (small but still
non-zero) delay between the audio PCs input signal and its output
signal. How can both signals run from the same clock?
The will simply be a fixed number of samples delay between input
and output. The thing that matters is that the sample frequencies
are exactly the same. If they are not you have to resample. Or does the
audio PC sync the output to a _later_ pulse form the clock
generator (i.e. all that matters is that the signal is synced with _a_
clock pulse, not necessarily _the_same_ clock pulse with which it was
originally transmitted)?
Indeed, that is what happens.
But if so, couldn't I have one clock between
the CD player and audio
PC, and a different clock between the PS and the DAC?
(I assume you mean _PC_ and DAC, or PC and digital amp).
In that case the PC has to do sample rate conversion. It will also
considerably complicate your SW. And is
self-clocking somehow inferior to an external clock? If an
external clock is better, how come ethernet/USB/firewire and all other
digital communication protocols run without external clocks? Sorry for
being thick, but I haven't worked with these things before...
In most digital formats the clock can be regenerated from the
digital signal. There's no quality loss involved in that - clock
jitter at a digital input doesn't matter as long as the clock is
good enough to recover the data without error. Professional audio
installations use a separate clock because 1) that is more reliable
when long cables are used, and 2), it makes the synchronisation
independent of the signal routing which in e.g. a studio isn't
just a simple linear chain as it would be in a domestic player
setup, and usually it isn't fixed either. but this also means that if you switch from blue-ray
player a to blue-ray
player b, your sound card must change its clocking source from input a to
input b. which might or might not cause an audible click or thump.
Can't you "fix" this by quickly muting the signal, then switch
sources, and unmute after you've "locked" onto the new signal? If the
switch is software-controlled (via RS-232) wouldn't that be fairly
simple to do? I don't
know whether it's cheaper/easier to
convert AES/EBU or ADAT to SPDIF coax/toslink...
AES/EBU and SPDIF are (almost) the same. The main differences
are in the electrical signal level and impedance. There are some
small coding differences as well, but in general an SPDIF input
will work with an AES signal, and an AES input *may* work with
an SPDIF signal provided the signal level is high enough, but
don't count on that. Then again,
there are some digital amps that take USB input (e.g. the
miniSTREAMER/miniAMP combo <URL:
In that case the PC's output sample rate is set by the amplifier.
Which means you can connect only one such amplifier unless they
are synchronised with external clocks (which they probably can't
do). And even if you can get them synced the PC will still have
to sample rate conversion if it plays from an external source.
So I'd say 'don't go that way'.
5:57 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On Mon, Dec 19, 2011 at 05:00:31PM +0100, Johan Herland wrote:
Thanks for quantifying this for me. Between the crazy
audiophile
forums and my lack of experience in this field, it's not always easy
to keep track of how much is good enough, and what is just crazy
over-engineering. :)
Be very careful with what you read on 'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days.
Ok, I think I see now. Basically in order to prevent
sample rate
conversion or extensive buffering (to compensate for clock drift), you
must make sure that the input and output runs off the very same clock
(with a whole constant number of clock cycles worth of audio being
processed at any time in the audio PC).
Almost all audio cards use the *same* clock for input and output
anyway. Either one generated internally, or from a separate clock
input, or derived from one of the inputs. The latter is usually fixed,
for example the RME AES card can take its clock from AES input #1, but
not from the other AES inputs.
Ok, so I might be able to convert from AES/EBU to
SPDIF using a simple
(maybe even passive) circuit?
In the worst case it needs some resistors or a small transformer, but
in many cases (RME included) all you need is a correctly wired cable.
That is because the RME outputs are transformer balanced and can be
switched to a lower voltage.
The delicate point for AES->SPDIF is not the signal level or impedance
(as the AES signal is much stronger than needed anyway), but if the
SPDIF input accepts the AES format. The audio data is identical for
both, but there are also some subcode bits. One of these identifies
the signal as either 'professional' or 'consumer' - this affects the
meaning of some other subcode bits. Some SPDIF inputs will refuse the
stream if the 'pro' bit is set. The RME driver allows to select this
bit (at least in the Windows version, but I suspect also on Linux),
so you're probably safe in that case.
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
20 Dec
20 Dec
12:19 a.m.
On Monday, December 19, 2011 06:16:06 PM Fons Adriaensen did opine:
On Mon, Dec 19, 2011 at 05:00:31PM +0100, Johan
Herland wrote:
Chuckle. I have to disagree a wee bit Fons, you under-estimate the
percentage that is all smoke and mirrors, its at least 99.9% these days. :)
Lots of card claim 24 bit, but a monotonicity check will often fail before
the 14th bit has been fully tested.
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
Nothing will dispel enthusiasm like a small admission fee.
-- Kim Hubbard
Thanks for quantifying this for me. Between the
crazy audiophile
forums and my lack of experience in this field, it's not always easy
to keep track of how much is good enough, and what is just crazy
over-engineering. :)
Be very careful with what you read on 'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days.
1:41 a.m.
On Tue, Dec 20, 2011 at 00:19, gene heskett <gheskett(a)wdtv.com> wrote:
On Monday, December 19, 2011 06:16:06 PM Fons
Adriaensen did opine:
Is this on pure digital devices (where you should expect bit-perfect results)?
Are you claiming that manufacturers in practice don't meet the specs
listed for their devices?
Does this seem like a trend, or are there just some interfaces with
bugs out there?
Are some brands notoriously worse than others?
(just curious, since one wouldn't want to buy crappy equipment)
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
Be very careful with what you read on
'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days.
Chuckle. I have to disagree a wee bit Fons, you under-estimate the
percentage that is all smoke and mirrors, its at least 99.9% these days. :)
Lots of card claim 24 bit, but a monotonicity check will often fail before
the 14th bit has been fully tested.
2:21 a.m.
On Monday, December 19, 2011 07:50:53 PM Johan Herland did opine:
List only reply..
On Tue, Dec 20, 2011 at 00:19, gene heskett
<gheskett(a)wdtv.com> wrote:
TBT, I am a bit hardware oriented, and I base that statement on reading the
data sheets on what look to be interesting bits of smart sand I have found
links to.
As I am not privy to what chip is on audio card x, at any one week n, I
think to actually name names & model numbers (something else I have learned
to miss-trust, dealing with ATI's video cards, which can get changed at
about the same rate we change underwear, can be exceptionally educational)
is generally pretty frustrating. So other than the card I presently use, I
won't give any name, rank or serial numbers. Why? Ears are at least as
variable as eyeballs. My eyes for instance, have had glass in front of
them for the last 71 years. My color vision OTOH, is very very good.
I don't claim to have ears that matched Larry Klien's when he was doing all
his pundrity in the audiofile rags 30 years ago, but at my age, I think
they are generally not lying to me when I listen to something for several
hours, with a rather eclectic mix of music, and do not get tired of it.
Toss in a bit of digital aliasing, a bit of monotonicity errors, then add a
3rd order intercept of 40db or less for the two tone so the net total
distortion approaches .5%, and these old ears are screaming for some
silence in 5 to 10 minutes.
However, I have been amazed at the number of folks out there who can listen
to 5% THD and a 20 db intercept on a two tone signal, with a bit of
aliasing thrown in, and absolutely cannot hear it, the difference to them
is a total shrug. It boggles my mind to think of what they are missing.
OTOH, given the class of folks who seem to inhabit this list, I think there
are some altogether decent ears here. Your tastes in music might assault
me, and mine would yours I'm equally sure, but the technical ability I hear
in most of the samples posted so far is far better done than the average
garage band console can pass. You are being 'picky' and for that I salute
you all. Technically, the state of the art in making music seems to
continually improve. And its people like you that are part of that driving
force behind that.
Thanks for reading this far. I'll get me coat now.
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
We will have solar energy as soon as the utility companies solve one
technical
problem -- how to run a sunbeam through a meter.
On Monday, December 19, 2011 06:16:06 PM Fons
Adriaensen did opine:
Is this on pure digital devices (where you should expect bit-perfect
results)?
Are you claiming that manufacturers in practice don't meet the specs
listed for their devices?
Does this seem like a trend, or are there just some interfaces with
bugs out there?
Are some brands notoriously worse than others?
(just curious, since one wouldn't want to buy crappy equipment)
Have fun! :)
...Johan Be very careful with what you read on
'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days.
Chuckle. آ I have to disagree a wee bit Fons, you under-estimate the
percentage that is all smoke and mirrors, its at least 99.9% these
days. :)
Lots of card claim 24 bit, but a monotonicity check will often fail
before the 14th bit has been fully tested.
12:29 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
On Mon, Dec 19, 2011 at 08:21:12PM -0500, gene heskett wrote:
Toss in a bit of digital aliasing, a bit of
monotonicity errors, then add a
3rd order intercept of 40db or less for the two tone so the net total
distortion approaches .5%, and these old ears are screaming for some
silence in 5 to 10 minutes.
Mmm. An IP3 of 40dB means that at e.g. -10dB (already high as
an average level), IM is at -110dB. That is nowhere near to
0.5%...
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
21 Dec
21 Dec
12:24 a.m.
On Tuesday, December 20, 2011 06:22:02 PM Fons Adriaensen did opine:
On Mon, Dec 19, 2011 at 08:21:12PM -0500, gene heskett
wrote:
Picky Fons :), but I swear I can hear it, or maybe I stated the
terminology wrong? Or have I been listening to SSB radios too long?
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
The state of innocence contains the germs of all future sin.
-- Alexandre Arnoux, "Etudes et caprices"
Toss in a bit of digital aliasing, a bit of
monotonicity errors, then
add a 3rd order intercept of 40db or less for the two tone so the net
total distortion approaches .5%, and these old ears are screaming for
some silence in 5 to 10 minutes.
Mmm. An IP3 of 40dB means that at e.g. -10dB (already high as
an average level), IM is at -110dB. That is nowhere near to
0.5%...
Ciao,
12:44 a.m.
New subject: What audio interface to use for a Linux-powered surround?preamp?
On Tue, Dec 20, 2011 at 06:24:13PM -0500, gene heskett wrote:
On Tuesday, December 20, 2011 06:22:02 PM Fons
Adriaensen did opine:
:-) Have you ever listened to AM radio on headphones with
LSB in the left channel and USB in the right one (and the
local oscillator synced to the AM carrier) ? It's absolutely
wonderful. All the interference is stereo, and the AM station
is exactly in the center which makes it very easy to ignore
the interference.
Regarding IM distortion, I've found that in many cases the
cause is not the active analog electronics but non-linear
contact resistance in switches, pots and faders. A few years
ago I did a routine check on my analog mixer. It had horrible
IM on one of the channels wired to the soundcard. And the
reason was simply that the mic/line switch on that channel
hadn't been used for years. Just toggling it a few times
made the distortion go away...
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
Mmm. An IP3 of 40dB means that at e.g. -10dB
(already high as
an average level), IM is at -110dB. That is nowhere near to
0.5%...
Picky Fons :), but I swear I can hear it, or maybe I stated the
terminology wrong? Or have I been listening to SSB radios too long?
4:52 a.m.
New subject: What audio interface to use for a Linux-powered surround?preamp?
On Tuesday, December 20, 2011 10:15:07 PM Fons Adriaensen did opine:
On Tue, Dec 20, 2011 at 06:24:13PM -0500, gene heskett
wrote:
Its been yonks, as in decades, and not in stereo, but a locked oscillator,
and yes, even then I was surprised at the quality. Which I felt at the time
could be largely attributed to the fact that the usual diode detector was
being pumped by the locked oscillator, getting rid of the very non-linear
detection characteristics from only having 3 or 4 volts of drive at dead
carrier levels in the average radio. I think much of the same advantages
can be had by a 4 quadrant multiplier type of detector, which separates the
signal from the LO but gives what may be mathematically provable to be
essentially identical except for the noise being in stereo in your example.
Something built around the MC-1495 would be a good starter point, although
I don't think its as linear as can be designed in discrete parts running in
an oven.
I'd consult with Bob Pease at National Semi., but his penchant for driving
an old VW bug eventually wrote his obit about a year back.
On Tuesday, December 20, 2011 06:22:02 PM Fons
Adriaensen did opine:
:
:-) Have you ever listened to AM radio on headphones with
LSB in the left channel and USB in the right one (and the
local oscillator synced to the AM carrier) ? It's absolutely
wonderful. All the interference is stereo, and the AM station
is exactly in the center which makes it very easy to ignore
the interference. Mmm. An IP3 of 40dB means that at e.g. -10dB
(already high as
an average level), IM is at -110dB. That is nowhere near to
0.5%...
Picky Fons :), but I swear I can hear it, or maybe I stated the
terminology wrong? Or have I been listening to SSB radios too long? Regarding IM distortion, I've found that in many
cases the
cause is not the active analog electronics but non-linear
contact resistance in switches, pots and faders. A few years
ago I did a routine check on my analog mixer. It had horrible
IM on one of the channels wired to the soundcard. And the
reason was simply that the mic/line switch on that channel
hadn't been used for years. Just toggling it a few times
made the distortion go away...
Oh my yes, lost track of the times over the last 60 years. And yet its
rare enough to catch me up sometimes till I've slept on it.
One other instance comes to mind, one of my employees had built an audio
board for the tv station, using 'raysistors' for the switching elements
ostensibly because the switching they could do was silent, unlike the
relays & switches of the day (1984). But at nominally +4dbm through them,
the intermod was pretty obvious. I pleaded for & bought a 12 + 4x a
channel input multiplier (48 in, 12 of them mics) board from Logitech that
used lamps & cds cells, much quieter intermod specs. But it was loaded
with enough 5532's that a cop of coffee sitting on it was nothing but 1/4"
of mud in the bottom in 2 hours. That heat killed the cds cells by the
grocery bag full. A couple 5" rotrons to suck out the hot air cut that
down to 1 or 2 a year, and some clip-on heat sinks helped the 5532's last
for years.
We ran that board till the digital switchover in mid 2008, 5.5 years after
my watch that officially started in Oct '84 was officially over. I think
Russ (the owner) grabbed it and sent it to one of his radio stations that
needed a board. It was too big for our satellite uplinking production
truck so it still has a 4 channel toy board in it IIRC.
Ciao,
Cheers Fons, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
Brain fried -- Core dumped
10:08 a.m.
New subject: What audio interface to use for a Linux-powered surround?preamp?
Showing my old age too...
I remember those things Gene and Fons were just talking about.
And I remember using 1495s in analog synth stuff a long time ago.
Thanks for the memories...
12:04 p.m.
New subject: What audio interface to use for a Linux-powered surround?preamp?
On Wednesday, December 21, 2011 06:02:01 AM Tracey Hytry did opine:
Showing my old age too...
I remember those things Gene and Fons were just talking about.
And I remember using 1495s in analog synth stuff a long time ago.
Thanks for the memories...
Recent ones at that, does any one here remember a vacuum tube, type 6BN6?
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
So I'm ugly. So what? I never saw anyone hit with his face.
-- Yogi Berra
1:21 p.m.
New subject: What audio interface to use for a Linux-powered?surround?preamp?
On Tue, Dec 20, 2011 at 10:51:59PM -0500, gene heskett wrote:
Its been yonks, as in decades, and not in stereo, but
a locked oscillator,
and yes, even then I was surprised at the quality.
It's quite easy to do in SW these days. All you need is one of those
japanese boxes that were mentioned on this list (IIRC) a few days
ago. They provide downconverted I,Q. Input via you soundcard, add
a PLL, a few multiplies and some filters.
I'd consult with Bob Pease at National Semi., but
his penchant for driving
an old VW bug eventually wrote his obit about a year back.
I didn't know he died... He has certainly been a source of
inspiration for an entire generation of analog designers.
I still keep the old NS databooks he wrote...
One other instance comes to mind, one of my employees
had built an audio
board for the tv station, using 'raysistors' for the switching elements
ostensibly because the switching they could do was silent, unlike the
relays & switches of the day (1984). But at nominally +4dbm through them,
the intermod was pretty obvious. I pleaded for & bought a 12 + 4x a
channel input multiplier (48 in, 12 of them mics) board from Logitech that
used lamps & cds cells, much quieter intermod specs. But it was loaded
with enough 5532's that a cop of coffee sitting on it was nothing but 1/4"
of mud in the bottom in 2 hours. That heat killed the cds cells by the
grocery bag full. A couple 5" rotrons to suck out the hot air cut that
down to 1 or 2 a year, and some clip-on heat sinks helped the 5532's last
for years.
Hehe. For me one of the main advantages of digital audio gear is that
signals don't pass through zillions of switches, pots, etc. as it would
in e.g. an analog mixer. That alone makes a very big difference.
Ciao,
--
FA
Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.
4:24 p.m.
New subject: What audio interface to use for a Linux-powered?surround?preamp?
On Wednesday, December 21, 2011 10:02:55 AM Fons Adriaensen did opine:
And I've been keeping every issue of whatever rag he was in at the time,
for around 35 years, just in case I could pick his brain for good ideas.
His dub did the usual dub trick of tucking the outside rear tire under
going around a corner a bit too pushy, the resultant roll hit a good sized
tree. Instantly according to the officers on the scene. Sad news indeed.
[...]
But it puts you at the mercy of digital multipliers for gain control.
There is a prop delay there that doesn't help. Using hardware helps, but
way too many bean counters force the use of software cuz its cheaper than
good 32 bit multiplier chips. Bean counters will be the death of us yet
you know. :)
On Tue, Dec 20, 2011 at 10:51:59PM -0500, gene heskett
wrote:
They've done a software radio issue occasionally in Circuit Cellar that I
try to grok, doing a lot without 20 lbs of hardware always amazes me. But
I like Steve's attitude about all this, he claims his favorite programming
language is solder. :) For me, I've gotten to the point where I use
solder only if I can't do it in 6x09 machine code. Yeah, I know its slow,
but its also quite easy to write good, dead stable software for it.
Its been yonks, as in decades, and not in stereo,
but a locked
oscillator, and yes, even then I was surprised at the quality.
It's quite easy to do in SW these days. All you need is one of those
japanese boxes that were mentioned on this list (IIRC) a few days
ago. They provide downconverted I,Q. Input via you soundcard, add
a PLL, a few multiplies and some filters. I'd
consult with Bob Pease at National Semi., but his penchant for
driving an old VW bug eventually wrote his obit about a year back.
I didn't know he died... He has certainly been a source of
inspiration for an entire generation of analog designers.
I still keep the old NS databooks he wrote... couple 5"
rotrons to suck out the hot air cut that down to 1 or 2 a
year, and some clip-on heat sinks helped the 5532's last for years.
Hehe. For me one of the main advantages of digital audio gear is that
signals don't pass through zillions of switches, pots, etc. as it would
in e.g. an analog mixer. That alone makes a very big difference. Ciao,
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
No directory.
19 Feb
19 Feb
12:27 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
Sorry to reply to such an old topic, going through my Linux Audio folder for the first
time in a few months (been to busy to keep up lately.)
I hope things have progressed nicely :-)
For audio I would also have suggested RME. It looks like you have chosen to go one of the
fully digital routes, to get all the AES you need straight out of the card. If you had
really wanted some analogue options alongside it then RME do ADAT<->AES converters,
so could have been used with one of the more general purpose cards with ADAT.
http://www.rme-audio.de/en_products_adi_4_dd.php
As to video. There are some HDMI capture cards available, EG Black Magic if on a tight
budget, but from a quick reading it seems they all only support the baseline of 2 channel
audio on HDMI, rather than the possible 8 channels.
A solution could be to use a HDMI->HDSDI converter, although being broadcast equipment
generally not cheap, but these can be found supporting the full 8 channels and HDSDI
ingest cards will generally support all 8 channels too.
Converter (they also do cards but couldn't find one suitable for you.)
http://www.dveo.com/broadcast-systems/HDMI-to-HD-SDI-converter.shtml
For professional video cards DVS and Deltacast seem to have good Linux support, Black
Magic to a lesser degree (and they're really prosumer at best.)
http://www.deltacast.tv
http://www.dvs.de
Neither has an off the shelf product that will help with HDMI ingesting but still may be
worth looking at/talking to.
Or maybe all Black Magic need to do is a firmware update for the Intensity or Deklink
Studio HDMI input cards . http://www.blackmagic-design.com/products/intensity/ &
http://www.blackmagic-design.com/products/decklink/
Or has the HDMI switch done away with the need for a lot of this? I
would of used a basic switch, which are available cheaply, then into the
computer using one of the above methods...
Good luck with it all anyway.
Dale.
From: gheskett(a)wdtv.com
To: linux-audio-user(a)lists.linuxaudio.org
Date: Mon, 19 Dec 2011 20:21:12 -0500
Subject: Re: [LAU] What audio interface to use for a Linux-powered surround preamp?
On Monday, December 19, 2011 07:50:53 PM Johan Herland did opine:
List only reply..
On Tue, Dec 20, 2011 at 00:19, gene heskett
<gheskett(a)wdtv.com> wrote:
TBT, I am a bit hardware oriented, and I base that statement on reading the
data sheets on what look to be interesting bits of smart sand I have found
links to.
As I am not privy to what chip is on audio card x, at any one week n, I
think to actually name names & model numbers (something else I have learned
to miss-trust, dealing with ATI's video cards, which can get changed at
about the same rate we change underwear, can be exceptionally educational)
is generally pretty frustrating. So other than the card I presently use, I
won't give any name, rank or serial numbers. Why? Ears are at least as
variable as eyeballs. My eyes for instance, have had glass in front of
them for the last 71 years. My color vision OTOH, is very very good.
I don't claim to have ears that matched Larry Klien's when he was doing all
his pundrity in the audiofile rags 30 years ago, but at my age, I think
they are generally not lying to me when I listen to something for several
hours, with a rather eclectic mix of music, and do not get tired of it.
Toss in a bit of digital aliasing, a bit of monotonicity errors, then add a
3rd order intercept of 40db or less for the two tone so the net total
distortion approaches .5%, and these old ears are screaming for some
silence in 5 to 10 minutes.
However, I have been amazed at the number of folks out there who can listen
to 5% THD and a 20 db intercept on a two tone signal, with a bit of
aliasing thrown in, and absolutely cannot hear it, the difference to them
is a total shrug. It boggles my mind to think of what they are missing.
OTOH, given the class of folks who seem to inhabit this list, I think there
are some altogether decent ears here. Your tastes in music might assault
me, and mine would yours I'm equally sure, but the technical ability I hear
in most of the samples posted so far is far better done than the average
garage band console can pass. You are being 'picky' and for that I salute
you all. Technically, the state of the art in making music seems to
continually improve. And its people like you that are part of that driving
force behind that.
Thanks for reading this far. I'll get me coat now.
Cheers, Gene
--
"There are four boxes to be used in defense of liberty:
soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
My web page: <http://coyoteden.dyndns-free.com:85/gene>
We will have solar energy as soon as the utility companies solve one
technical
problem -- how to run a sunbeam through a meter.
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On Monday, December 19, 2011 06:16:06 PM Fons
Adriaensen did opine:
Is this on pure digital devices (where you should expect bit-perfect
results)?
Are you claiming that manufacturers in practice don't meet the specs
listed for their devices?
Does this seem like a trend, or are there just some interfaces with
bugs out there?
Are some brands notoriously worse than others?
(just curious, since one wouldn't want to buy crappy equipment)
Have fun! :)
...Johan
Be very careful with what you read on
'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days.
Chuckle. آ I have to disagree a wee bit Fons, you under-estimate the
percentage that is all smoke and mirrors, its at least 99.9% these
days. :)
Lots of card claim 24 bit, but a monotonicity check will often fail
before the 14th bit has been fully tested.
2:11 p.m.
Thanks for your reply; some comments below.
2012/2/19 Kaza Kore <dj_kaza(a)hotmail.com>om>:
As to video. There are some HDMI capture cards
available, EG Black Magic if
on a tight budget, but from a quick reading it seems they all only support
the baseline of 2 channel audio on HDMI, rather than the possible 8
channels.
[...]
An added complication here, is that I want something that will work
with HDCP (AFAICS, the prosumer video equipment you mention will not
accept HDCP signals). Also, I'm not really in the market for lots of
video equipment. I simply want something that can switch between
mutiple HDMI sources (including consumer equipment such as BluRay/DVD
players), and then split the chosen HDMI signal, so that video goes
directly to the screen, and audio goes to my computer for further
processing.
Hence, I'm not really in the market for prosumer/professional video
equipment. I don't actually want the video signal into my computer at
all. However, there is currently no (consumer-level) standard (except
HDMI) for transferring multi-channel uncompressed digital audio
between devices. Ideally, I'd want a device with HDMI input and
HDMI(video) output, along with outputs for transmitting 8ch LPCM
(decoded from the HDMI audio signal). The audio output could be
something like 1 x ADAT, or even 4 x SPDIF (or AES/EBU) - I don't
really care - as long as it could be straightforwardly connected to my
computer.
Or has the HDMI switch done away with the need for a
lot of this? I would of
used a basic switch, which are available cheaply, then into the computer
using one of the above methods...
For now, I am using a cheap HDMI switch with an audio de-multiplexer
that redirects the 2ch. bitstream to an SPDIF output. This is
obviously not ideal, since it cannot handle >2ch LPCM (including the
Dolby TrueHD or DTS-HD Master Audio formats), but it _is_ able to do
the Dolby Digital and DTS formats that are compressed into the 2ch.
stream. This is "good enough" for now, since most of my video material
is DVDs (although the BluRay collection is growing slowly), and most
of the high-quality audio is in computer files that I can play
directly from the computer and thus does not have to traverse the HDMI
switch.
Considering the direction technology is moving, I might have most
video and audio accessible directly from the computer in the future,
thus making the HDMI switch less important (only needed for things
like console games, etc.)
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
7:29 p.m.
New subject: What audio interface to use for a Linux-powered surround preamp?
Content Protection sucks!
Going via an analogue hole would be one way, but a pain in the arse. I wouldn't expect
any prosumer, and even less professional, to allow you to bypass encryption though.
Software solutions may show up...
Seems there is one HD switch/multiplexer available which does though.
http://www.moomecard.com/index.php?option=com_content&view=article&…
Output the separate audio and video you want (although audio is on 2x TOSLink.)
But obviously there is the worry that a new version of HDCP will be release tomorrow so
bypassing in this manner is never ideal!
Dale.
Date: Sun, 19 Feb 2012 14:11:14 +0100
Subject: Re: [LAU] What audio interface to use for a Linux-powered surround preamp?
From: jherland(a)gmail.com
To: dj_kaza(a)hotmail.com
CC: gheskett(a)wdtv.com; linux-audio-user(a)lists.linuxaudio.org
Thanks for your reply; some comments below.
2012/2/19 Kaza Kore <dj_kaza(a)hotmail.com>om>:
As to video. There are some HDMI capture cards
available, EG Black Magic if
on a tight budget, but from a quick reading it seems they all only support
the baseline of 2 channel audio on HDMI, rather than the possible 8
channels.
[...]
An added complication here, is that I want something that will work
with HDCP (AFAICS, the prosumer video equipment you mention will not
accept HDCP signals). Also, I'm not really in the market for lots of
video equipment. I simply want something that can switch between
mutiple HDMI sources (including consumer equipment such as BluRay/DVD
players), and then split the chosen HDMI signal, so that video goes
directly to the screen, and audio goes to my computer for further
processing.
Hence, I'm not really in the market for prosumer/professional video
equipment. I don't actually want the video signal into my computer at
all. However, there is currently no (consumer-level) standard (except
HDMI) for transferring multi-channel uncompressed digital audio
between devices. Ideally, I'd want a device with HDMI input and
HDMI(video) output, along with outputs for transmitting 8ch LPCM
(decoded from the HDMI audio signal). The audio output could be
something like 1 x ADAT, or even 4 x SPDIF (or AES/EBU) - I don't
really care - as long as it could be straightforwardly connected to my
computer.
Or has the HDMI switch done away with the need
for a lot of this? I would of
used a basic switch, which are available cheaply, then into the computer
using one of the above methods...
For now, I am using a cheap HDMI switch with an audio de-multiplexer
that redirects the 2ch. bitstream to an SPDIF output. This is
obviously not ideal, since it cannot handle >2ch LPCM (including the
Dolby TrueHD or DTS-HD Master Audio formats), but it _is_ able to do
the Dolby Digital and DTS formats that are compressed into the 2ch.
stream. This is "good enough" for now, since most of my video material
is DVDs (although the BluRay collection is growing slowly), and most
of the high-quality audio is in computer files that I can play
directly from the computer and thus does not have to traverse the HDMI
switch.
Considering the direction technology is moving, I might have most
video and audio accessible directly from the computer in the future,
thus making the HDMI switch less important (only needed for things
like console games, etc.)
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
20 Feb
20 Feb
12:26 a.m.
On Sun, Feb 19, 2012 at 19:29, Kaza Kore <dj_kaza(a)hotmail.com> wrote:
Content Protection sucks!
Indeed. I'm only trying to access my legally owned content at the best
possible quality, but HDCP and industry collusion makes it almost
impossible. Talk about shooting yourself in the foot...
Going via an analogue hole would be one way, but a
pain in the arse. I
wouldn't expect any prosumer, and even less professional, to allow you to
bypass encryption though. Software solutions may show up...
Yeah, doing an analog hole would sort of defeat the entire purpose of
my exercise (which is to keep the audio digital for as long as
possible - ideally all the way to the speaker).
Seems there is one HD switch/multiplexer available
which does though.
http://www.moomecard.com/index.php?option=com_content&view=article&…
Output the separate audio and video you want (although audio is on 2x
TOSLink.)
Hmm. I don't think it actually supports any more than 2ch LPCM (and
the compressed Dolby/DTS formats that fit into the 2ch bandwidth). The
2 TOSLink outputs are probably not connected to send separate parts of
a multi-channel signal (in any case, they'd only manage to send 2x2
channels, which isn't enough for 7.1 surround). Instead, they probably
supply the 2ch audio for each of the DVI outputs.
But obviously there is the worry that a new version of
HDCP will be release
tomorrow so bypassing in this manner is never ideal!
Yeah, as long as we're dependent on proprietary formats and physical
media, I guess we'll have to deal with the industry's misguided
attempts to control the consumer.
Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
20 Dec
20 Dec
1:29 a.m.
On Mon, Dec 19, 2011 at 17:57, Fons Adriaensen <fons(a)linuxaudio.org> wrote:
This looks promising. It seems I'm closing in on a system looking
something like this:
- Various HDMI sources connected to an RS-232 controlled HDMI switch
with SPDIF output (e.g.
http://www.marmitek.com/en/product-details/audio-video-at-home/connect/conn…
). Ideally, I wanted a better method of extracting audio from the HDMI
switch, but it seems none are available.
- I also have a networked media drive that will be available to the
audio PC for direct playback.
- An audio PC with the RME HDSPe AES interface, doing surround
decoding, volume control, room correction, and crossover in software,
and outputting 2-channels-per-speaker mixed PCM data on the available
AES/EBU outputs.
- Running one AES XLR cable to each speaker position, where each
(2-way) speaker has a dedicated 2-channel digi-amp/powerDAC that does
the final conversion to the analog speaker inputs.
- I'm currently looking at digi-amps from SUMOH (
http://sumoh.com/index-2.html ), which seem to offer a nice tradeoff
between price and performance, although my search is far from over.
I currently have a 5.1 speaker system, so I'll need 5 amps (running
the subwoofer through its own analog amp for now). Given that I
already have the audio PC (minus audio interface), and the HDMI switch
is on its way, a quick calculation puts me somewhere near €2500, which
is in line with what I'd originally estimated for this project.
That should give me a good starting point for playing around with the
software, especially the room correction and crossover configurations.
I'll also be quite flexible hardware-wise:
- Expanding to 7.1 (requires a total of 7x2 (+ 1) channels, which the
RME card will handle; then simply add 2 speakers and 2 amps).
- Upgrading one amp at a time, and experimenting with different amps
for different speakers (e.g. putting some more money in the front
speaker amps, since those are more "important" than the surround
speakers).
Thanks again for all your help. Have fun! :)
...Johan
--
Johan Herland, <jherland(a)gmail.com>
www.herland.net
On Mon, Dec 19, 2011 at 05:00:31PM +0100, Johan
Herland wrote:
Thanks. Useful advice. I like to think that I only spend money on
stuff for which I have a clear and scientific rationale for why it's
worth the money, but it does require a lot of time researching
available options and understanding the underlying technologies (don't
get me wrong, the learning process is fun as well). However, when you
get into the specific implementations and "secret sauce" of chip
technologies, it's sometimes impossible to make 100% informed
decisions. For example, I'm currently comparing digital amplifier
technologies, especially TI Equibit/PurePath HD vs. Diodes/Zetex DDFA
(they seem to be the main commercially available contenders in the
all-digital HiFi-amp space), but so far there are so many stats and
other variables that muddy the waters. At some point I'll just have to
listen to the available options, and simply let my ear decide.
Thanks for quantifying this for me. Between the
crazy audiophile
forums and my lack of experience in this field, it's not always easy
to keep track of how much is good enough, and what is just crazy
over-engineering. :)
Be very careful with what you read on 'audiophile' forums. 99% of the
high-end audio business is based on ignorance and hype these days. Ok, so I might
be able to convert from AES/EBU to SPDIF using a simple
(maybe even passive) circuit?
In the worst case it needs some resistors or a small transformer, but
in many cases (RME included) all you need is a correctly wired cable.
That is because the RME outputs are transformer balanced and can be
switched to a lower voltage.
The delicate point for AES->SPDIF is not the signal level or impedance
(as the AES signal is much stronger than needed anyway), but if the
SPDIF input accepts the AES format. The audio data is identical for
both, but there are also some subcode bits. One of these identifies
the signal as either 'professional' or 'consumer' - this affects the
meaning of some other subcode bits. Some SPDIF inputs will refuse the
stream if the 'pro' bit is set. The RME driver allows to select this
bit (at least in the Windows version, but I suspect also on Linux),
so you're probably safe in that case.
4667
days inactive
4733
days old
linux-audio-user@lists.linuxaudio.org
24 comments
6 participants
participants (6)
-
Fons Adriaensen -
gene heskett -
Johan Herland -
Jörn Nettingsmeier -
Kaza Kore -
Tracey Hytry