search for: latencies

Alexander Chemeris wrote: > Hi, > > Here are some questions from user point of view. :) > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > >> I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, >> and there aren't any obvious artifacts. I also did a 16=>16 test, and the >> results are delayed by 10ms

As the subject line says: Is PING a good indicator of network latency? If not, how can I measure latency? Thanks, Cosmin Prund

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

Hello-- There must be a better way to do this. I have a class method in my model that finds averages and does a few calculations using find_by_sql. The problem I¹m encountering is that all computed columns from MySQL come back as type string. E.g., def self.find_averages(domain_id) if @@domain_average return @@domain_average else @@domain_average =

Hi, I see that currently physical register uses marked as "undef" can still cause dependencies. Is this intentional? SU(9): %D5<def,undef> = LDrid %R0, 0, %R10<imp-def>, %R11<imp-def> # preds left : 0 # succs left : 11 # rdefs left : 0 Latency : 1 Depth : 0 Height : 0 Successors: ...

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500

Attached is a snapshot of work-in-progress of a FFT based resampler. At the moment it works in floating point only, and only basic quality inspection has been done. Some benchmarks comparing the filter-based resampler at Q3 with the FFT resampler with overlap = in_len / 2, using 20ms chunks of data. (-O3 -ffast-math, FFTW3, gcc 4.3.0 on x86_64) 16=>48: 59us vs 19us 16=>44.1: 204us vs

Hi all, I have a question regarding the latency of the true dependency of a store followed by an aliased load in ScheduleDAGInstrs. The latency seems to depend on the store and load being volatile or not as can be seen in the post-RA-sched debug output of the attached ARM example: $ llc -O3 -debug-only=post-RA-sched store_load_latency_test.ll ... SU(2): STRi12 %R2<kill>,

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in

I'm planning on setting up some remote agents and before doing so, I did some simple PING tests to measure latency. The average latency I got was 250ms. Does anyone have experience in terms of quality of calls when there is such high latency? Can anyone comment? Thanks, Waldo

Hello I just installed a TE110P in a Dell Poweredge 750 (rackmount), which is connected via crossover T1 cable to a adit 600. Anyway, I've encountered an array of errors, which I believe I have narrowed down to the 22 IRQ misses I encounter on zttool. I've noticed that the te110p and the usb device share an IRQ. Could that be the source of my woes? Would disabling the usb device in the

On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > Alexander Chemeris wrote: > > On 5/29/08, Thorvald Natvig <thorvald at natvig.com> wrote: > > > I've done listening tests when converting wb_male.wav to 44.1, 48 and 8khz, > > > and there aren't any obvious artifacts. I also did a 16=>16 test, and the > > > results are delayed

Hi i have found a problem related with tbf and the latency that the tbf calculates.. I have used the following parameters for burst and limit burst 100Kbit limit 500Kbit lat81.8ms burst 6Kbit limit 6Kbit lat 0us burst 200Kbit limit 100Kbit lat 4294.9s As u can see in the 3rd column the latency for 100Kbit burst and 500Kbit limit is 81.8ms but for 200Kbit and limit 100Kbit is 4294.9s!!! How

Hi, I'm trying to understand why MachineScheduler does a poor job in straight line code in cases like the one in the attached debug dump. This is on AMDGPU, an in-order target, and the problem is that the IMAGE_SAMPLE instructions have very high (80 cycle) latency, but in the resulting schedule they are often placed right next to their uses like this: 1784B %140:vgpr_32 =

Hi George, Tow months ago, we talked about how to reduce the scheduling latency for a specific VM which runs a mixed workload, where the boost mechanism can not works well. I have tried some methods to reduce the scheduling latency for some assumed latency-sensitive VMs and got some progress on it. Now I hope to make it on demand. That is to say, I hope to get the scheduler to determine the

We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

Hi Andy, Thanks for the explanations. Yes AMDGPU is in-order and has MicroOpBufferSize = 1. Re "issue limited" and instruction groups: could it make sense to disable the generic scheduler's detection of issue limitation on in-order CPUs, or on CPUs that don't define instruction groups, or some similar condition? Something like: --- a/lib/CodeGen/MachineScheduler.cpp +++

Hi folks last weekend, I tried Windows Messenger first time and was stunned by the little latency it gives. Until now, I've been using softphones on Linux exclusively, like iaxcomm, linphone and sjphone, and they all give me about 1, at times even 2 secs delay. Whereas Messenger really seems to be in the millisec range. Of course, I'm now curious why there is that difference. Clearly,

Hello everybody! I have 3 gluster servers (*gluster 3.12.6, Centos 7.2*; those are actually virtual machines located on 3 separate physical XenServer7.1 servers) They are all connected via infiniband network. Iperf3 shows around *23 Gbit/s network bandwidth *between each 2 of them. Each server has 3 HDD put into a *stripe*3 thin pool (LVM2) *with logical volume created on top of it, formatted

...firefox, I experienced a latency of app. 15 secs. That's what I was expecting, but is it possible to reduce this latency further. Furthermore, must annoyingly, since the stream is playing continuously, when I send new file to be played, the latency grows as the time goes. I was able to measure latencies up to 1:30 or more after 20-30 mins of continous playing. Is that normal, or is there some parameter to set up in order to avoir this behavior. Related to this, I read that it's possible to send directly raw PCM to icecast when live streaming from the audio card input. Is it possible to send r...