search for: 100ms

...;") in new stack > AlarmReceiver: Setting read and write formats to ULAW > AlarmReceiver: Answering channel > AlarmReceiver: Waiting for connection to stabilize > AlarmReceiver: Waiting for first event from panel > AlarmReceiver: Sending 1400Hz 100ms burst (ACK) > AlarmReceiver: Sending 2300Hz 100ms burst (ACK) > AlarmReceiver: DTMF Digit Timeout on Zap/1-1 == AlarmReceiver: Incomplete string: , trying again... > AlarmReceiver: Sending 1400Hz 100ms burst (ACK) > AlarmReceiver: Sending 2300Hz 100ms burst...

...roothub.portstatus [1] = 0x00030300 PESC CSC LSDA PPS hub 2-0:1.0: port 2, status 0300, change 0003, 1.5 Mb/s usb 2-2: USB disconnect, address 87 usb 2-2: unregistering device usb 2-2: usb_disable_device nuking all URBs usb 2-2: unregistering interface 2-2:1.0 hub 2-0:1.0: debounce: port 2: total 100ms stable 100ms status 0x300 hub 1-0:1.0: state 7 ports 10 chg 0000 evt 0004 ehci_hcd 0000:00:02.1: GetStatus port 2 status 001403 POWER sig=k CSC CONNECT hub 1-0:1.0: port 2, status 0501, change 0001, 480 Mb/s hub 1-0:1.0: debounce: port 2: total 100ms stable 100ms status 0x501 ehci_hcd 0000:00:02.1...

...stecho sample application, since in testecho the tail is set to 1024 samples. But, as I explained, I used my test program for producing the result, and there I set the tail length to 200ms (1600 samples). *And*, in addition to that, I delay the frames that's fed to speex_echo_cancellation() by 100ms, to reflect the playback/recording delay when the mic.pcm file was recorded. This means, if I play the frame now, I will give it to speec_echo_cancellation() 100ms later (similar to your PLAYBACK_DELAY in mdf.c). So this is equal to shifting the blue signal in your graph to the right by 100ms. It&...

The PCI_EXP_TYPE_PCI_BRIDGE should be PCI_EXP_FLAGS_TYPE here. Also a tiny fix to the python comment. Signed-off-by: Dexuan Cui <dexuan.cui@intel.com> _______________________________________________ Xen-devel mailing list Xen-devel@lists.xensource.com http://lists.xensource.com/xen-devel

...; >> */ >> DECLARE_EWMA(pkt_len, 0, 64) >> >> +#define VF_TAKEOVER_INT (HZ / 10) >> + >> #define VIRTNET_DRIVER_VERSION "1.0.0" >> >> static const unsigned long guest_offloads[] = { > Why is this delay necessary? And why by 100ms? This is based on netvsc implementation and here is the commit that added this delay.? Not sure if this needs to be 100ms. commit 6123c66854c174e4982f98195100c1d990f9e5e6 Author: stephen hemminger <stephen at networkplumber.org> Date:?? Wed Aug 9 17:46:03 2017 -0700 ??? netvsc: delay setu...

...; >> */ >> DECLARE_EWMA(pkt_len, 0, 64) >> >> +#define VF_TAKEOVER_INT (HZ / 10) >> + >> #define VIRTNET_DRIVER_VERSION "1.0.0" >> >> static const unsigned long guest_offloads[] = { > Why is this delay necessary? And why by 100ms? This is based on netvsc implementation and here is the commit that added this delay.? Not sure if this needs to be 100ms. commit 6123c66854c174e4982f98195100c1d990f9e5e6 Author: stephen hemminger <stephen at networkplumber.org> Date:?? Wed Aug 9 17:46:03 2017 -0700 ??? netvsc: delay setu...

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second invite obviously is interpreted as second...

...the file's sampling rate, and mic signal is mic-xx. The result is result-xx-yy, where yy is either rc1 or beta3 depending on Speex version that I used for the test. For the result, I use my test program for the test [1], and during the test I set echo tail length setting to 200ms, and delay to 100ms (meaning the speaker signal that's given to speex_echo_cancellation() is 100ms older than what's currently playing). The frame length is set to 20ms. And to switch between rc1 and beta3, I simply replace mdf.c from the corresponding tarball. Nothing else is changed. The actual tail length...

Hello: I have set up a samba share between 2 linux machines, where the server updates a file (latest.txt, 200bytes) at 15 Hz. My problem is that the client does not get the latest version of this file latest.txt when it reads this file every 100ms or so. It seems to be updating the file once a second. For instance, when I check for the contents of the file latest.txt every 100ms, the same contents are printed for 10 checks although the actual file has changed almost every time I check it. It seems to be a cache problem (I really don't ne...

...gt; > > >> +#define VF_TAKEOVER_INT (HZ / 10) > > >> + > > >> #define VIRTNET_DRIVER_VERSION "1.0.0" > > >> > > >> static const unsigned long guest_offloads[] = { > > > Why is this delay necessary? And why by 100ms? > > > > This is based on netvsc implementation and here is the commit that > > added this delay.? Not sure if this needs to be 100ms. > > > > commit 6123c66854c174e4982f98195100c1d990f9e5e6 > > Author: stephen hemminger <stephen at networkplumber.org>...

> FYI: The below is just my interpretation of the code, I might be wrong. Most of it is right. Actually, would you mind if I use part of your email for documenting the jitter buffer in the manual? > Each time a new packet arrives, the jitter buffer calculates how far ahead > or behind the "current" timestamp it is; this is called arrival_margin. > The "current"

...gt; > > >> +#define VF_TAKEOVER_INT (HZ / 10) > > >> + > > >> #define VIRTNET_DRIVER_VERSION "1.0.0" > > >> > > >> static const unsigned long guest_offloads[] = { > > > Why is this delay necessary? And why by 100ms? > > > > This is based on netvsc implementation and here is the commit that > > added this delay.? Not sure if this needs to be 100ms. > > > > commit 6123c66854c174e4982f98195100c1d990f9e5e6 > > Author: stephen hemminger <stephen at networkplumber.org>...

...lence suppression, >this needs to be full duplex, and I doubt you don't get that from a >modem. If we could add silence suppression, we could do with half >duplex, effectively saving half the bandwidth. in addition, if we could >increase the packetization to something like 50 or 100ms for extreme >use (like dial-up), we could end up with a lot less. Say, using 100ms >slicing with IAX and speex at 6.3kbps, we'll end up with effective >bandwidth use of 8860bps plus ethernet. now that's something you can >brag about.... >roy Miguel Ruiz Velasco ___...

My understanding is that: 1. rails is single threaded and serves one request at a time 2. mongrel is single threaded and can process one rails request at a time 3. mongrel_cluster starts a few mongrel servers on the same machine Based on this, the single most important factor becomes the time it takes to load a single page (to avoid blocking). If this is all true, what''s considered a

...whenever I skip or interpolate a packet. This means that > if the packets are on average 20 ms in advance when we drop a frame, > then they will be considered all "on time" (0 ms) after that. Yes, but assume that after a long steady period, your network latency suddenly drops with 100ms. (100ms is excessive, but I see 60ms quite frequently from users on DSL/Cable connections who also do a bit of P2P on the same line) What happens now is that the +100ms bin starts increasing steadily, and suddenly it's enough to skip a frame. A frame is skipped, and the histogram gets shift...

Hi! there are some problems with echo cancellation filter from speex. problem 1: some noises, echo is removed, but sometimes you can hear some noises instead of echo. I was trying with many different parameters for buffer length (40ms and 20ms), filter length (from 100ms to 4s) and echo tail (2 to 5 buffers), but could not find the right setting. problem 2: it happens that the filter suddenly stops working and returns silence, so you don't hear anything. any ideas how to cope with this problems? regards hs

...ssume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN->UK + UK->China (around 1000ms !) or only from Japan to China??? Please let me know. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/...

...nternet connection has no CIR, only "maximum dl/ul speeds" given by provider * most important: our outgoing and incoming traffic must be shaped to some rate that will provide possibly low latency. For users that do not have active connections I''d like to ensure no more than 100ms latency for ping or any other low-traffic connections For several years of my experiments with traffic shaping over Linux I found no tool for creating such system. For example, HTB require given, constant ''ceil'' parameter. I would like to have some qdisc that can automatic...

...n, >>this needs to be full duplex, and I doubt you don't get that from a >>modem. If we could add silence suppression, we could do with half >>duplex, effectively saving half the bandwidth. in addition, if we could >>increase the packetization to something like 50 or 100ms for extreme >>use (like dial-up), we could end up with a lot less. Say, using 100ms >>slicing with IAX and speex at 6.3kbps, we'll end up with effective >>bandwidth use of 8860bps plus ethernet. now that's something you can >>brag about.... > > >>ro...

...in zapata.conf, when I press the "R" key nothing happens. The other side of the communication hears a "click" but everything stays the same. I have found by trial and error (and in some docs after that) that the standard flash hook timing for my country (Spain) phone systems is 100ms. So, if I put rxflash=100 in Zap/1 channel configuration in zapata.conf (actually any value below 104) Asterisk detects when I press the "R" key, but instead of interpreting it as a flash hook it's interpreted as if I hung up the phone. I get a new dialtone but the other side gets dis...