Buffer size and recording delay

Hi
I use a Focusrite Saffire LE audio interface, and in my core audio window the buffer size is currently set at 256 and there is a recording delay of 28ms. I don't think I've ever altered these settings since I started to use Logic. I don't know if this is related, but I do notice a slight delay when I monitor what I'm recording -- a slight delay, for example, between my voice (as I'm recording it) through the headphones and my "direct" voice.
Anyway, are these settings OK, or should they be altered for optimum performance?

256 samples Buffer size will always give you a noticable amount of latency. If you use Software Monitoring you should try setting your Buffer to 64 samples. With the recording delay slider in the preferences->Audio you can compensate for the latency (of course not in realtime) so that the Audio will be placed exactly there, where it should hae been recorded at. In your case set it to a -value. A loopback test (check the link below) will clarify the exact amount of Latency occuring on your system.
http://discussions.apple.com/thread.jspa?threadID=1883662&tstart=0

Similar Messages

Buffer Size and Latency

I knew the 2 might be connected but I recently read some information which confused me. I thought a larger buffer size was good, but I've just read that increasing the buffer size lengthens latency. Can anyone enlighten me on the relationship between these; just simple answers, please.

Audio is sent through the system in chunks. Chunks of the buffer size. Buffer size is measured in samples so a buffer setting of 256 samples rewquests and provides audio in 256 sample packets. As a real time application, this cannot be predicted so the buffer reads from the present/past as opposed to looking ahead of time. The chunk will be handed off to CoreAudio that much ( time ) later, in comparison. So...
Larger buffer increases track/plugin count since you allow the computer more time to manage it's tasks.
Decreasing allows for less total system throuhput ( = less latency ).
Many people have found a good balance for this ( if they need to dynamically alter system resources ) to be:
Low buffer size while recording, for minimal latency
Then increase buffer for mixdown, allowing more pluggies and tracks.
Your latency setting is also dependent on your audio hardware and system busses. So, a good interface and driver on a dedicated FireWire bus can allow you to decrease your buffer setting.
J

Data retrieval buffers - buffer size and sort buffer size

Any difference to tune BSO and ASO on data retrieval buffers?
From Oracle documentation, the buffer size setting is per database per Essbase user i.e. more physical memory will be used if there are lots of concurrent data access from users.
However even for 100 concurrent users, default buffer size of 10KB (BSO) or 20KB (ASO) seems very small compare to other cache setting (total buffer cache is 100*20 = 2MB). Should we increase the value to 1000KB to improve the data retrieval performance by users? The improvement impact is the same for online application e.g. Hyperion Planning and reporting application e.g. Financial Reporting?
Assume 3 Essbase plan types with 100 concurrent access:
PLAN1 - 1000KB * 100 = 100MB 1000KB * 100 = 100MB (total sort buffer size)
PLAN2 - 1000KB * 100 = 100MB 1000KB * 100 = 100MB (total sort buffer size)
PLAN3 - 1000KB * 100 = 100MB 1000KB * 100 = 100MB (total sort buffer size)
Total physical memory required is 600MB.
Thanks in advance!

256 samples Buffer size will always give you a noticable amount of latency. If you use Software Monitoring you should try setting your Buffer to 64 samples. With the recording delay slider in the preferences->Audio you can compensate for the latency (of course not in realtime) so that the Audio will be placed exactly there, where it should hae been recorded at. In your case set it to a -value. A loopback test (check the link below) will clarify the exact amount of Latency occuring on your system.
http://discussions.apple.com/thread.jspa?threadID=1883662&tstart=0

I/O Buffer Size Causes Bounce Delay

Hi All,
Came across a really frustrating problem here. I've got a pretty sizable project I'm trying to finish up. I'm about to send it to someone else to mix, so I'm bouncing out a few of the tracks that contain instruments that the mixer doesn't have. Unfortunately, everything I bounce is coming out delayed, depending on the I/O buffer size. Just to verify this, I bounced out a clip with the buffer set to 128 and one with the buffer at 1024, both with the click left on. Sure enough, the 1024 is pretty significantly delayed from the click, whereas the 128 is pretty much right on it.
I'm using some inserts across the project, but really nothing significant. All instruments, vocals, etc. are being bussed to aux tracks for grouping and then all of the group tracks are being sent to a master bus track which is going out the stereo out. I don't have any processors on the master bus or stereo out, and PDC is set to "all".
Anyone else experiencing this? More importantly, does anyone have a solution? I'd really rather not have to figure out the sample delay and shift everything.

I would think it has more to do with the Fast Track Pro than your computer or Logic. Does the FTP have drivers or is it system compliant? (plug it in and it just works)
All things considered that's a very inexpensive interface.
Please don't take this next bit personally... Unless you have a pristine listening environment and top of the line equipment... recording at 96kHz is not really a wise decision. Recording at 24-bit will have a much greater impact than the sample rate.
Here's what happens at 96kHz.
The files recorded are twice the size of files recorded at 48kHz.
A 96kHz / 24-bit stereo file is approx 35 Megs a minute, add to that effects running at 96kHz and is fairly easy to over load an iMac or Laptop type machine.. a MacPro will fare a little better, depending on the audio hardware.
Effects like Space Designer and some of the top line AU-instruments s-u-c--k CPU cycles as well.
What I think you're seeing is an overload of the bus system, why it happens at higher buffer levels is anyone's guess, most would say it's the iffy M-Audio drivers.
pancenter-

Buffer size and samples per channel

I have a question regarding the allocation of the output buffer. I have
a digital I/O card PCI-6534 and I use the driver NI-DAQmx 7.4.0f0. I
would like to generate a digital output using different clock rates.
For example, I need to write 500 samples at 1000 samples per second and
other 500 samples at a rate of 10000 samples per second. the simplest
solution is to prepare two different waveforms, write the first one on
the buffer and generate the output. Then I can load the second waveform
on the buffer and generate the second output. In order to minimize the
delay between the two output sequences, I would like instead to write
the buffer once. I tried to set the buffer size to 1000, and the number
of samples per second to 500, but it doesn't work. There is a way to
set independently the buffer dimension and the number of samples which
have to be generated?

I can post the whole thing but I'll talk a little about it. It's a producer consumer loop with a continuous analog input sample from external clock program in the producer loop. Right now the consumer loop has a simple write to spreadsheet VI in it but eventually I want to average each revolution (well, two since it's a four stroke but that's neither here nor there) of pressure traces and spit out a single curve.
The wiring is simple. I have a voltage supply feeding the encoder and the quadrature A input on PFI 8 of the 6212 DAQ. I also have the Z index plugged in but nothing else to it. The analog input is a BNC right to AI 0. I can make a diagram if you want one. I've scoped the rotary encoder output and it looks great, very square with small relative rise times.
Attachments:
Yanmar Program V2.vi ‏46 KB

T61p 200gb at 7200rpm buffer size and real world performance?

Hi, I wanted to confirm what the buffer size on the 200gb 7200 option of the T61p had... there seems to be conflicting information regarding this model. I'd also appreciate any information regarding the real world performance of this particular drive...
Thanks!
Message Edited by carthikv12 on 05-19-2008 09:31 AM
Solved!
Go to Solution.

both the hitachi and seagate 200GB/7200RPM drives used in the T61p have 16MB of onboard cache. performance tests of these two drives are scattered across the internet. search google for "hitachi 200GB 7K200" and "seagate 200GB 7200.3" respectively.
ThinkStation C20
ThinkPad X1C · X220 · X60T · s30 · 600

Out of Memory Error, Buffer Sizes, and Loop Rates on RT and FPGA

I'm attempting to use an FPGA to collect data and then transfer that data to a RT system via a FIFO DMA, and then stream it from my RT systeam over to my host PC. I have the basics working, but I'm having two problems...
The first is more of a nuisance. I keep receiving an Out of Memory error. This is a more recent development, but unfortunately I haven't been able to track down what is causing it and it seems to be independent of my FIFO sizes. While not my main concern, if someone was able to figure out why I would really appreciate it.
Second, I'm struggling with overflows. My FPGA is running on a 100 MHz clock and my RT system simply cannot seem to keep up. I'm really only looking at recording 4 seconds of data, but it seems that no matter what I do I can't escape the problem without making my FIFO size huge and running out of memory (this was before I always got the Out of Memory error). Is there some kind of tip or trick I'm missing? I know I can set my FPGA to a slower clock but the clock speed is an important aspect of my application.
I've attached a simplified version of my code that contains both problems. Thanks in advance for taking a look and helping me out, I appreciate any feedback!

David-A wrote:
The 7965 can't stream to a host controller faster than 800MB/s. If you need 1.6GB/s of streaming you'll need to get a 797x FPGA module.
I believe the application calls for 1.6 GB over 4s, so 400 MB/s, which should be within the capabilities of the 7965.
I was going to say something similar about streaming over ethernet. I agree that it's going to be necessary to find a way to buffer most if not all of the data between the RT Target and the FPGA Target. There may be some benefit to starting to send data over the network, but the buffer on the host is still going to need to be quite large. Making use of the BRAMS and RAM on the 7965 is an interesting possibility.
As a more out there idea, what about replacing the disk in your 8133 with an SSD? I'm not entirely sure what kind of SATA connection is in the 8133, and you'd have to experiment to be sure, but I think 400 MB/s or close to that should be quite possible. You could write all the data to disk and then send it over the network from disk once the aquisiton is complete. http://digital.ni.com/public.nsf/allkb/F3090563C2FC9DC686256CCD007451A8 has some information on using SSDs with PXI Controllers.
Sebastian

ASIO Buffer Size for Records

Hello!!!
Please, help me to set up the ASIO buffer latency to make recordings made by Audition is clean and without cracklings and etc. When I had a value 1 ms of latency, after I've recorded a sample from any source by Audition, I've got some crackling when I playback this sample (in real-time in Audition, and in the recorded file). When I set up 5 ms - it's better. Maybe there is a standard to set up the ASIO buffer latency for Audition? I have a "Creative ASIO" driver and a device "Creative X-Fi Elite Pro".
I just wanna using Audition to make records instead using the Creative audio recording software, which make records without crackling. I think Audition just have to make records without any cracklings too or might be better!!!
Thank you!!!

There's no quick answer. In reality, it's a matter of turning it up until you get reliable recording and playback. How much latency you need will depend on the spec of your computer, how well your computer is optimised for recording, how much other rubbish is running in background and the quality of the ASIO drivers provided with your sound card.
The first thing I have to say is that the Creative X-Fi series of sound cards are NOT well thought of in the recording community. They're aimed far more at gamers and people who want to watch videos than at serious sound recordists. I've also heard a lot of horror stories about the poor quality of their drivers.
Beyond that, I'd suggest that you optimise your computer for recording--exactly how to do this will depend on what operating system your have. However, a Google search on "optimising computer for sound recording" including your Windows version will find you lots of suggestions. There will likely be some pages from Presonus near the top of your results--those are very good ones.
Another point: shut down anything and everything running in background when you record. Pay special attention to networking, Wifi and any anti virus software you have--all those are bad news when recording and mixing.
Finally, depending on how/what you're recording, there may or may not be issues with cranking the ASIO latency up really high. The problem with high latency only really comes into play when you're multi tracking along to pre recorded tracks AND you are monitoring your new track via the computer rather than directly on the sound card. Proper interfaces for multitrack recording feature "direct hardware monitoring" which allows you to monitor via the interface rather than having to go via your software. If you're serious about recording, you'll eventually need to ditch the X-Fi and get a specialist audio interface.
So, optimise your computer, shut down anything unnecessary (especially wifi and anti virus) and just turn up the latency until it's reliable. I know of some people who use a relatively low latency for recording and turn it up for mixing when they use lots of real time effects.

Adobe AIR dynamic sound and recording delay issue

I'm writing mobile application with Adobe AIR. The application use AIR Microphone API to record sound to file and later replay it.
The problem manifest only on mobile devices, not simulator. Specifically only on iOS devices, android devices seem to work OK.
Sometimes the recorded sound is missing samples. I know this because I use iFunBox to copy the recorded file to another application that replay it. The dropped frame manifest during playback as very fast audio because only part of the samples were recorded.
Sometime the playback is to slow which manifest as very slow audio. I know this because when the recording is fine and the other application play the sound right or when I take a file I recorded in the simulator (which run on my MacBook) and it only play slow on the mobile device.
How can I make sure the sound is good even when the application is a bit busy?

A little bump.
I still haven't found any solution to this.

Based on my computer what I/O buffer size and process buffer should I have to stop getting the system too slow error message?

I need some sugestions on my settings. I have a 13 in macbook pro runnig 10.8.2 with a 2.7 Ghz i7 with 4 GB of ram.
I am using Logic pro X 10.0.3 and when I get to a really load heavy section of a project then I get a too slow error message.
I know this will happen from time to time because of my computer but is there anyway to make it better?

Im curious how you are running LPX on an OS X 10.8.2 system?
As to your question.. there is no specific answer... Try playing with the settings (making a note of your current ones) and see what improvements, if any, are made under different combinations of the settings.
Personally I'd add more RAM if you can... as that alone may help matters by some degree...
Finally,
This article may also help.. (Its written for LP9 but most of it is applicable for LPX too)
http://support.apple.com/kb/TA24535

DBMS_LOB.WRITEAPPEND Max buffer size exceeded

Hello,
I'm following this guide to create an index using Oracle Text:
http://download.oracle.com/docs/cd/B19306_01/text.102/b14218/cdatadic.htm#i1006810
So I wrote something like this:
CREATE OR REPLACE PROCEDURE CREATE_INDEX(rid IN ROWID, tlob IN OUT NOCOPY CLOB)
IS
BEGIN
 DBMS_LOB.CREATETEMPORARY(tlob, TRUE);
 FOR c1 IN (SELECT ID_DOCUMENT FROM DOCUMENT WHERE rowid = rid)
 LOOP
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('<DOCUMENT>'), '<DOCUMENT>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('<DOCUMENT_TITLE>'), '<DOCUMENT_TITLE>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH(NVL(c1.TITLE, ' ')), NVL(c1.TITLE, ' '));
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('</DOCUMENT_TITLE>'), '</DOCUMENT_TITLE>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('</DOCUMENT>'), '</DOCUMENT>');
 FOR c2 IN (SELECT TITRE,TEXTE FROM PAGE WHERE ID_DOCUMENT = c1.ID_DOCUMENT)
 LOOP
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('<PAGE>'), '<PAGE>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('<PAGE_TEXT>'), '<PAGE_TEXT>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH(NVL(c2.TEXTE, ' ')), NVL(c2.TEXTE, ' '));
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('</PAGE_TEXT>'), '</PAGE_TEXT>');
 DBMS_LOB.WRITEAPPEND(tlob, LENGTH('</PAGE>'), '</PAGE>');
 END LOOP;
 END LOOP;
END;
Issue is that some page text are bigger than 32767 bytes ! So I've got an INVALID_ARGVAL...
I can't figure out how can I increase this buffer size and how to manage this issue ??
Can you please help me :)
Thank you,
Ben
Edited by: user10900283 on 9 févr. 2009 00:05

Hi ben,
I'm afraid, that doesn't help much, since you have obviously rewritten your procedure based on the advise given here.
Coluld you please post your new procedure, as formatted SQL*Plus, embedded in {noformat}{noformat} tags, like this:SQL> CREATE OR REPLACE PROCEDURE create_index(rid IN ROWID,
2 IS
3 BEGIN
4 dbms_lob.createtemporary(tlob, TRUE);
5
6 FOR c1 IN (SELECT id_document
7 FROM document
8 WHERE ROWID = rid)
9 LOOP
10 dbms_lob.writeappend(tlob, LENGTH('<DOCUMENT>'), '<DOCUMENT>');
11 dbms_lob.writeappend(tlob, LENGTH('<DOCUMENT_TITLE>')
12 ,'<DOCUMENT_TITLE>');
13 dbms_lob.writeappend(tlob, LENGTH(nvl(c1.title, ' '))
14 ,nvl(c1.title, ' '));
15 dbms_lob.writeappend(tlob
16 ,LENGTH('</DOCUMENT_TITLE>')
17 ,'</DOCUMENT_TITLE>');
18 dbms_lob.writeappend(tlob, LENGTH('</DOCUMENT>'), '</DOCUMENT>');
19
20 FOR c2 IN (SELECT titre, texte
21 FROM page
22 WHERE id_document = c1.id_document)
23 LOOP
24 dbms_lob.writeappend(tlob, LENGTH('<PAGE>'), '<PAGE>');
25 dbms_lob.writeappend(tlob, LENGTH('<PAGE_TEXT>'), '<PAGE_TEXT>');
26 dbms_lob.writeappend(tlob
27 ,LENGTH(nvl(c2.texte, ' '))
28 ,nvl(c2.texte, ' '));
29 dbms_lob.writeappend(tlob, LENGTH('</PAGE_TEXT>'), '</PAGE_TEXT>')
30 dbms_lob.writeappend(tlob, LENGTH('</PAGE>'), '</PAGE>');
31 END LOOP;
32 END LOOP;
33 END;
34 /
Advarsel: Procedure er oprettet med kompileringsfejl.
SQL>
SQL> DECLARE
2 rid ROWID;
3 tlob CLOB;
4 BEGIN
5 rid := 'AAAy1wAAbAAANwsABZ';
6 tlob := NULL;
7 create_index(rid => rid, tlob => tlob);
8 dbms_output.put_line('TLOB = ' || tlob); -- Not sure, you can do this?
9 END;
10 /
create_index(rid => rid, tlob => tlob);
FEJL i linie 7:
ORA-06550: line 7, column 4:
PLS-00905: object BRUGER.CREATE_INDEX is invalid
ORA-06550: line 7, column 4:
PL/SQL: Statement ignored
SQL>

Getting recv buffer size error even after tuning

I am on AIX 5.3, IBM J9 VM (build 2.3, J2RE 1.5.0 IBM J9 2.3...), Coherence 3.1.1/341
I've set the following parameters as root:
no -o sb_max=4194304
no -o udp_recvspace=4194304
no -o udp_sendspace=65536
I still get the following error:
UnicastUdpSocket failed to set receive buffer size to 1428 packets (2096304 bytes); actual size is 44 packets (65536 bytes)....
The following commands/responses confirm that the settings are in place:
$ no -o sb_max
sb_max = 4194304
$ no -o udp_recvspace
udp_recvspace = 4194304
$ no -o udp_sendspace
udp_sendspace = 65536
Why am I still getting the error? Do I need to bounce the machine or is there a different tunable I need to touch?
Thanks
Ghanshyam

Can you try running the attached utility, and send us the output. It will simply try to allocate a variety of socket buffer sizes and report which succeed and which fail. Based on the Coherence log message I expect this program will also fail to allocate a buffer larger then 65536, but it will allow you verify the issue externally from Coherence.
There was an issue with IBM's 1.4 AIX JVM which would not allow allocation of buffers larger then 1MB. This program should allow you to identify if 1.5 has a similar issue. If so you may wish to contact IBM support regarding obtaining a patch.
thanks,
Mark Attachment: so.java (*To use this attachment you will need to rename 399.bin to so.java after the download is complete.) Attachment: so.class (*To use this attachment you will need to rename 400.bin to so.class after the download is complete.)

Doing Buffered Event count by using Count Buffered Edges.vi, what is the max buffer size allowed?

I'm currently using Count Buffered Edges.vi to do Buffered Event count with the following settings,
Source : Internal timebase, 100Khz, 10usec for each count
gate : use the function generator to send in a 50Hz signal(for testing purpose only). Period of 0.02sec
the max internal buffer size that i can allocate is only about 100~300. Whenever i change both the internal buffer size and counts to read to a higher value, this vi don't seem to function well. I need to have a buffer size of at least 2000.
1. is it possible to have a buffer size of 2000? what is the problem causing the wrong counter value?
2. also note that the size of max internal buffer varies w
ith the frequency of signal sent to the gate, why is this so? eg: buffer size get smaller as frequency decrease.
3. i'll get funny response and counter value when both the internal buffer size and counts to read are not set to the same. Why is this so? is it a must to set both value the same?
thks and best regards
lyn

Hi,
I have tried the same example, and used a 100Hz signal on the gate. I increased the buffer size to 2000 and I did not get any errors. The buffer size does not get smaller when increasing the frequency of the gate signal; simply, the number of counts gets smaller when the gate frequency becomes larger. The buffer size must be able to contain the number of counts you want to read, otherwise, the VI might not function correctly.
Regards,
RamziH.

About Labview tool kit for Ti's DSP (RTDX buffer size)

Now I am using Labview tool kit for Ti's DSP to
do the RTDX communication between Labview and Ti's 6713DSK. Now I am
encounting a problem that the my Human interface (Labview) works fine in
the first 10 minutes. However after that data looks freeze. The RTDX
become very slow. It's hard to input data or receive data to DSP through
Labview. I am sending 1 Integer 16 bit data and 1 floating data to DSP
using RTDCXwrite vi and the frequency is 33.3Hz. And at the same time I am
transfer 5 integer 16 bit data from DSP to Labview. The frequency is 25Hz.
What's wrong with this? Is that because the buffer for RTDX has been used
up? And how to set up the RTDX buffer in Labview? because I check the vi
about RTDX and can't find a definition about the buffer. This is very
important in my project. Please tell me how to set up the buffer size and
the RTDX work state (continuous mode or discontinuous). Thank you very
much and I am looking forward to hearing from you.

Now I am using winxp, 512Mb RAM, LV version is 7.1. My labview coding and DSP code main function is attached below. Please help me out. I want to transfer 8 16bit integer data. Is it better that I used an array to group these data then transfer this array using only one channel. Compared with using 8 independent channels to transfer them.
Attachments:
HMI_pannel.zip ‏85 KB

Onboard buffer size

Hi, can anyone tell me the difference between the "buffer size" and "onboard buffer size" under the DAQmx buffer>>output property?
Thanks,
David
www.controlsoftwaresolutions.com
Solved!
Go to Solution.

OnbrdBufSize - is the fast memory every DAQ card has to store the acquired data. It's like RAM in the PC. For example here you find that the NI PCI 6120 has 128MB of this memory which will equal to some amount of samples the buffer can store (this number you will read). This said it also explains why you can read and not set this property.
Output.BufSize - Before you start data acquisition software buffer is defined based on the task you perform. I think it just allocates part of the fast memory of the DAQ card. The reason why the buffer is set (either by the user or automatically) is to have defined memory space for each task. Imagine you have more tasks using the card and each of them needs some memory of the card - if I am wrong correct me somebody.
If your acquisition is finite (sample mode on the Timing function/VI set to Finite Samples), NI-DAQmx allocates a buffer equal in size to the value of the samples per channel attribute/property. For example, if you specify samples per channel of 1,000 samples and your application uses two channels, the buffer size would be 2,000 samples. Thus, the buffer is exactly big enough to hold all the samples you want to acquire.
If the acquisition is continuous (sample mode on the Timing function/VI set to Continuous Samples), NI-DAQmx allocates a buffer equal in size to the value of the samples per channel attribute/property, unless that value is less than the value listed in the following table. If the value of the samples per channel attribute/property is less than the value in the table, NI-DAQmx uses the value in the table.
Sample Rate Buffer Size
No rate specified 10 kS
0–100 S/s 1 kS
100–10,000 S/s 10 kS
10,000–1,000,000 S/s 100 kS
>1,000,000 S/s 1 MS
LV 2011, Win7

Buffer size and recording delay

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