Re: Is ALSA the right tool for my app?

"sonofzev@xxxxxxxxxxxx" <sonofzev@xxxxxxxxxxxx> · Fri, 14 Mar 2008 16:11:02 +0900

Hi John 

Following from Sean, the appropriate system to use would be netjack. http://netjack.sourceforge.net/

I know Geoff Beasley showed me his design for a studio here in Melbourne that utilises it, (however I've been slack and 2 years later still haven't visited)

One thing that would be important to consider is latency, I don't believe it is possible to have a low latency running off a network. So it depends on what you are doing. If it is a simple control room and recording room setup this should be okay, however if you want to isolate musicians to say different rooms in your house and then do a "live" take you may have some issues resulting from the latency. 

I've just moved house and will be setting up a new home studio with a mixing room that doubles as a control room for recordings in the living room. Unfortunately I have to sit an exam soon and won't be able to fully test it out for at least a month. 

cheers

Allan 

On Thu Mar 13 10:54 , Sean Edwards  sent:

If this is a network application, you may be on your way with ALSA for sound.  However, for network communication and synchronization, you may want to look into Jack (the Jack Audio Connection Kit):

http://jackaudio.org/

I am sure you will get other helpful suggestions from this list.

-=Sean Edwards=-

----- Original Message ----

From: John Sigler 

To: alsa-user@xxxxxxxxxxxxxxxxxxxxx

Sent: Thursday, March 13, 2008 12:16:13 PM

Subject:  Is ALSA the right tool for my app?

Hello,

I've 
been 
struggling 
with 
alsa-lib 
for 
about 
a 
week, 
mainly 
trying 
to 

come 
up 
with 
the 
appropriate 
configuration 
file.

I'd 
like 
to 
describe 
what 
I'm 
trying 
to 
do, 
so 
that 
ALSA 
users 
might 

tell 
me 
whether 
the 
ALSA 
infrastructure 
is 
the 
correct 
tool 
for 
the 

app 
I'm 
writing.

Consider 
two 
systems 
A 
and 
B 
in 
separate 
physical 
locations, 
both 

connected 
to 
some 
kind 
of 
network.

A 
is 
considered 
the 
"sender" 
because 
it 
captures 
audio, 
and 
sends 
it 

to 
B, 
while 
B 
is 
considered 
the 
"receiver" 
because 
it 
receives 
the 

data 
sent 
by 
A, 
and 
plays 
it 
back.

The 
code 
running 
on 
A 
is 
conceptually 
simple:

while 
( 
1 
)

{

capture 
a 
packet's 
worth 
of 
data 
(e.g. 
1024 
bytes)

add 
some 
information 
in 
a 
header

send 
the 
datagram 
to 
B

}

The 
input 
is 
an 
AES 
signal. 
I've 
been 
told 
that 
it 
is 
very 
rare 
for 

capture 
boards 
to 
provide 
the 
raw 
AES 
frames. 
Is 
this 
correct? 
If 
it 

is 
true, 
the 
board 
would 
capture 
the 
AES 
payload 
(either 
PCM 
or 

compressed 
Dolby 
E).

Now, 
to 
add 
some 
complexity, 
I'm 
considering 
a 
board 
with 
multiple 

inputs, 
to 
be 
able 
to 
run 
many 
processes 
in 
parallel.

(I 
have 
an 
RME 
AES-32 
board 
which 
provides 
8 
AES 
inputs.)

On 
B, 
the 
code 
is 
more 
complex, 
because 
the 
issue 
of 
synchronization 

rears 
its 
ugly 
head.

B 
is 
trying 
to 
playback 
the 
stream 
sent 
by 
A, 
with 
a 
small 
time 
shift 

(e.g. 
200 
ms) 
at 
the 
exact 
same 
rate 
that 
it 
was 
captured 
by 
A. 
This 

means 
the 
code 
needs 
to 
buffer 
a 
number 
of 
audio 
frames, 
then 
start 

the 
playback, 
and 
monitor 
the 
level 
of 
the 
hardware 
buffer 
to 

determine 
whether 
that 
buffer 
is 
slowly 
emptying 
or 
filling 
up. 
Every 

time 
B 
receives 
data 
from 
A, 
it 
pushes 
the 
data 
to 
the 
board's 

hardware 
buffer. 
To 
keep 
the 
level 
constant, 
it 
might 
need 
to 
add 
or 

remove 
a 
few 
frames 
once 
in 
a 
while.

The 
AES-32 
board 
also 
provides 
8 
AES 
outputs, 
so 
I'd 
use 
another 
such 

board 
on 
B. 
There 
might 
be 
several 
streams 
played 
back 
at 
the 
same 
time.

What 
is 
your 
opinion 
/ 
thoughts 
on 
this 
architecture?

-- 

Regards.

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