Hi Katja,
Regarding all this with...
Digital clocks need to be synchronized to work together or else you will
get clicks. As you have said it can be done in a driver or it can be
done on-chip if your hardware has that feature. It does indeed add to
the latency. On pro sound cards you would find word clock to sync cards
together or possibly time code input/output for synchronising with video
cameras. Consumer hardware does not have these features since most
people wouldn't have any use for this. Some equipment can hold a very
stable clock even after being disconnected from the others. This feature
comes with added costs.
Super expensive digital radio mics have a noticeable latency even though
they are highly specialised equipment built to minimise this very
latency. It's about 3ms for a good setup.
Anyhow the lowest latency you will find is using analog wireless packs.
The only point of digital radio mics is Hollywood can't probably cope
with non-encrypted dialogues of your favourite TV series during the shoot.
Don't go for the cheapest, and you might not need the high end. You can
rent that stuff if it's not needed for such a long time.
The Sennheiser e100 (gen3) is pretty decent for the price. That would be
about 450~500€ for a kit.
There is a pro level audio-over-ethernet distribution system called
ethersound. It's proprietary, but worth looking into. As with other
digital format, there is a master clock somewhere in the system to which
every one syncs. Once again digital converters are the main culprits
when it comes to latency.
http://www.ethersound.com/technology/overview.php
Cheers
Pierre-Olivier
On 06/03/2013 12:41, katja wrote:
Netjack docs look promising. It binds Jack clients on the transmitter
computer (called 'slave') to the sample clock on the receiver computer
('master'). Netjack supports transmission over wifi, and over internet
(using the CELT codec). It is primarily designed for 'distributed
music'. Here is a paper introducing Netjack:
http://www.google.nl/url?q=http://lac.linuxaudio.org/2009/cdm/Saturday/22_Hohn/22.pdf&sa=U&ei=0AM3UajlJIfAO5qDgIgF&ved=0CDgQFjAJ&usg=AFQjCNF47yIrp3T6cuIXCKhl0AUIaeiA2w
For anyone using Pd on a mobile device as sound generator, Netjack
could be a great way to deliver audio over wifi. For my purpose
though, I still need access to the sound card at the transmitter side
to capture audio from a mic. An audio rate conversion routine called
AlsaInOut is provided with Netjack so you can use the slave's sound
card for playback. However, this can not be used for capturing or
monitoring, due to large processing delay. So, clock drift is the show
stopper here.
Clock drift is not related to network transmission but to the slight
differences in clock speeds of sound cards even if they are set to the
same nominal sample rate. It also happens with two sound cards used
together on one computer, like when using an USB mic for input and the
system sound card for output. Core Audio can handle such a setup, but
at the expense of increased and fluctuating latency. Jack can combine
soundcards, but buffer under- or overruns are audible.
What is so hard about compensating clock drift? If it must be done in
a theoretically correct way, pitch should not change so it will be a
time stretching routine which will indeed introduce considerable
latency. However, clock speeds often differ by a very small amount,
like a factor 1/1000 or less. What if we just accept the pitch
difference? You would get playback speeds like 1.001 or 0.999,
perfectly acceptable for my purpose at least. This would greatly
simplify the task of resampling.
Katja
On 3/5/13, Charles Goyard <[email protected]> wrote:
Hi,
see if netjack fills the bill :
Netjack is a Realtime Audio Transport over a generic IP Network. It is
fully integrated into JACK.
When you transport Audio Signals over a Network, you will always have
the problem, that two Computers are bound to two different Sample
Clocks. The other Network Transports face the Problem of compensating
for the drift between these Sample Clocks. This is very hard to do.
NetJack works around this problem, by syncing all Clients to one
Soundcard. So there will be no resampling or glitches in the whole
network. Packet loss is now also handled gracefully. By using the celt
codec, its even possible, that single packet losses get masked by the
Packet Loss Concealment Code.
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack2
Cheers,
--
Charles
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