search for: 10ms

...r the reverse? Also, is this speech or music? What tool, what options? In general, it helps a lot if you post the sample (input and output). Cheers, Jean-Marc On 06/03/2016 12:48 PM, Kevin Connor wrote: > Hi Opus list, > > I'm noticing a discontinuity in the quality between use of 10ms and > 20ms framesize for mode=AUDIO at a bitrate of about 28000. > Quality drops audibly for voice signals when encoded at 20ms > framesize, versus quality at 10ms. This effect is mode=AUDIO only. > Using mode==VOIP shows no sig. difference between 10 and 20ms framing > at this bi...

Hi Jean-Marc, Sorry for late reply, thanks for interest. It's quality good for 10ms/audio, poorer for 20ms/audio. Quality equivalent for 10,20ms for mode=voip. PESQ was the tool that alerted me to something of interest, but I don't trust PESQ to almost any degree! It's good for hearing relative differences, of course, but not absolutes. Bitrate here was 28kbps, but...

...it as a DNS server beyond what is in the attached named.conf file. Here are the results of pinging the virtual Linux servers NetBIOS name followed by the results of pinging the DNS name: C:\>ping cofr3 Pinging cofr3 [172.17.60.6] with 32 bytes of data: Reply from 172.17.60.6: bytes=32 time=10ms TTL=254 Reply from 172.17.60.6: bytes=32 time<10ms TTL=254 Reply from 172.17.60.6: bytes=32 time<10ms TTL=254 Reply from 172.17.60.6: bytes=32 time<10ms TTL=254 Ping statistics for 172.17.60.6: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-...

Tested DTX feature in Opus 1.0.2 and 1.1 for 20, 40, and 60ms frame size. They all generated only one encoded frame every 400ms during silence period as expected. However, for 10 ms frame size, the DTX period was 200ms instead. Has anyone else experienced the same? Thanks, Chris Wang -------------- next part -------------- An HTML attachment was scrubbed... URL:

...d. With -O3, the overhead of copying, applying window etc is 20%. With all options, the overhead sinks to 10%. 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 and within +/- 1 (basically, rounding errors from the FFT). -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: speex-block-snapshot.diff Url: http://lists.xiph.org/pipermail/speex-dev/attachments/20080529/6819bbbe/attachment.txt

I am trying to use the speex echo canceler and preprocessor with sipXtapi to develop a sip user agent. the sipXmedialib call flow graph uses 10ms frames and I am not sure what the implications are if I try to change this. The documentation seemed to indicate that a 20ms frame was recomended, but it didn't go into the consquences of using other frame sizes. We are going to use headsets in production but there is still a subtle echo t...

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2)

...il 100Mbit tc class add dev eth0 parent 1:1 classid 1:2 htb rate 100Mbit prio 2 ceil 100Mbit tc filter add dev eth0 parent 1:0 prio 5 u32 \ match ip src 10.10.10.1 \ match ip dst 10.10.10.10 \ flowid 1:2 tc qdisc add dev eth0 parent 1:2 handle 2: sfq perturb 10 Small pings are ok - <10ms, here is output for 65500 byte pings: Pinging test [10.10.10.1] with 65500 bytes of data: Reply from 10.10.10.1: bytes=65500 time=120ms TTL=64 Reply from 10.10.10.1: bytes=65500 time=120ms TTL=64 Reply from 10.10.10.1: bytes=65500 time=120ms TTL=64 Reply from 10.10.10.1: bytes=65500 time=120ms TT...

...e down. Connection to MySQL failed ----------------------------------------------------------------------------------- The paradox to me is that I can ping the WINS server ok: C:\>ping -a 172.16.0.121 Pinging ELY [172.16.0.121] with 32 bytes of data: Reply from 172.16.0.121: bytes=32 time<10ms TTL=128 Reply from 172.16.0.121: bytes=32 time<10ms TTL=128 Reply from 172.16.0.121: bytes=32 time<10ms TTL=128 Reply from 172.16.0.121: bytes=32 time<10ms TTL=128 Ping statistics for 172.16.0.121: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in mi...

...:) > > 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 and within +/- 1 (basically, rounding errors >> from the FFT). >> > > Do you plan to use it in VoIP environment? If so, 10ms of additional latency > is a big cost for resampling. Do you think there are any ways to reduce it? > Yes, I plan to use it in a VoIP environm...

Hello everyone, Can any body please tell me the compiler options to optimise the speex code for ARM TE architecture. Fixed point is enabled. It currently takes about 700ms to encode 5 sec worth of voice. I need to get it down to under 10ms. Is the option --enable-arm5te-asm supported? I am using speex 1.1.12 version. The configure.ac file does not seem to support --enable-arm5te-asm option. I tried --enable-arm5e-asm but the result is'nt much different. Real time is not possible at that rate. Processor is Xscale 533MHz....

From: Stephen Hemminger <stephen at networkplumber.org> Date: Tue, 19 Dec 2017 09:55:48 -0800 > could be 10ms, just enough to let udev do its renaming Please, move to some kind of notification or event based handling of this problem. No delay is safe, what if userspace gets swapped out or whatever else might make userspace stall unexpectedly?

From: Stephen Hemminger <stephen at networkplumber.org> Date: Tue, 19 Dec 2017 09:55:48 -0800 > could be 10ms, just enough to let udev do its renaming Please, move to some kind of notification or event based handling of this problem. No delay is safe, what if userspace gets swapped out or whatever else might make userspace stall unexpectedly?

...his is clear in ping results - I don't know if it's just a ping problem or if it manifests elsewhere as well. It does also affect mtr and traceroute times. I guess it's more a kernel issue than a CentOS problem, but is anyone else seeing the same? Ping results have a resolution of 10ms - so a ping <10ms shows as 0.000ms... --- On an Opteron machine running CentOS-3.4 --- # uname -a Linux omega.xxx 2.4.21-27.0.2.ELsmp #1 SMP Wed Jan 19 11:28:26 CST 2005 x86_64 x86_64 x86_64 GNU/Linux # ping 192.168.1.200 PING 192.168.1.200 (192.168.1.200) 56(84) bytes of data. 64 bytes from...

...ros, assertions, and checked arithmetic and then encoded 2 hours of mixed speech/audio with these parameters: Sample rate = 48000 Channels = 1 Application = OPUS_APPLICATION_AUDIO Bitrate = 24 KB/s Force Mode = MODE_SILK_ONLY Signal Type = OPUS_SIGNAL_AUTO Complexity = 10 Frame size = 480 samples (10ms) No errors came up in encoding. Chandrakala, are these the encoding parameters that you believe should trigger the error? - Logan Hi, > > We are looking at the VAD implementation used in opus. We are looking at > the code where speech probability is calculated based on which SNR is >...

...> > the quality. Higher quality => higher latency. You could set the overlap to > > > 0, but that would give you some nasty artifacts. > > > You can also resample with smaller block sizes. In the example I used > 20ms > > > blocks and 50% overlap. If you use 10ms blocks and 50% overlap, latency > > > sinks to 5ms. > > > > How quality and CPU usage depends from block size? > > Looking at edge case - 1 sample blocks - I should have <1 sample latency, > > but something should be wrong here, e.g. required CPU power may be &g...

Hi Opus list, I'm noticing a discontinuity in the quality between use of 10ms and 20ms framesize for mode=AUDIO at a bitrate of about 28000. Quality drops audibly for voice signals when encoded at 20ms framesize, versus quality at 10ms. This effect is mode=AUDIO only. Using mode==VOIP shows no sig. difference between 10 and 20ms framing at this bitrate. Pesq tot...

...gt; 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 and within +/- 1 (basically, rounding errors > > > from the FFT). > > > > > > > > > > Do you plan to use it in VoIP environment? If so, 10ms of additional > latency > > is a big cost for resampling. Do you think there are any ways to reduce > it? &...

Thank you Jean-Marc. My understanding is that G.729 is a telephone codec, so there must have been some reason why its developers went to 10ms/frame. Do you know why that might be? From a recent post on this list I saw somebody talking about your decoded sample rate being 8KHZ/sec. and then he mentioned that being 160 bytes at 20ms/frame. That said, I take it that your decoded samples are 1 byte wide ... is that correct? We're...

Hi all, I''m running some tests with netem and I noticed some strange behaviour that looks like a bug: I''m pinging another machine and adding delay with netem. When I tell netem to give me a 10ms delay, it works fine. The problem is that when I ask for a 11ms delay, it gives me 20ms! It happens for any value between 11ms an 20ms, and it repeats for values over 20ms, now rounding up to 30ms. Here is an example: # ping x.x.x.x PING x.x.x.x (x.x.x.x) 56(84) bytes of data. 64 bytes from x....