search for: 40ms

Hi folks I have a handset talking to Asterisk, which in turn puts the call through to an ITSP. The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP). The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset, it simply relays it on to the ITSP....

Is is possible to give a short hint about how the jitter buffer would "catch up" when network condition have been bad and then get better? I'm using the jitter buffer with success now, but sometimes I have a long delay that's caused by bad network conditions and then later when the conditions get better, I would think we would want the audio to gradually catch up with real-time

...due to the increased latency. The echo occurs even when I call an automated voice service or something, so I'm thinking that possibly there is a gross impedance mismatch on my side of the telco switch. I believe the following is happening (latency measures are guesses): Handset -(0ms)- PAP2 -(40ms)- Asterisk -(40ms)- SPA3000 -(0ms)- Telco Does an EC algorithm need a measurable delay to work? The EC would have to cope with an almost unmeasurably small echo delay (the delay only creeps in on the other side of the IP link. Is there another way I should be solving this problem, especially as I...

...tes how far ahead or behind the "current" timestamp it is; this is called arrival_margin. The "current" timestamp is simply the last frame successfully decoded. It maintains a list of bins for margins, this is short and longterm margin. Think of the bins like this: -60ms -40ms -20ms 0ms +20ms +40ms +60ms when a packet arrives, the margin matching it's arrivel_margin is increased, so if this packet was 40ms after the current timestamp, the 40ms bin would be increased. If this packet arrived 60ms too late (and hence is useless), the -60ms bin would increase. early...

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

Hi, We're using R in an application where asking for a probability of an event takes about 130ms. What could we do to take that down to 30ms-40ms? The query code uses randomforest, knn. -- M.

Hi all, Please guide me How to change frame_size of opus codec at run-time (20ms, 40ms, 60ms) I'm stucking in this case: 1. init codec width default config (frame_size =20ms, bandwidth=48KHz, bitrate = 48kbps...), then in runtime changing: - bitrate = 24, 16, 6kbps: sound is OK - frame_size = 40ms, 60ms: Not OK, sound is distort so bad 2. init codec with frame_size = 40ms ,...

...nsuming roughly 4-23kbps, although the average would be pretty low most of the time... > 3) What is the recomended buffer? 1 second or 2 seconds? That's way too large for an interactive conversation. I've been experimenting with different buffer sizes and my current favorite is 40ms. Sending 40ms of audio over the wire results in a delay of roughly 40ms+transmission delay+playback latency+codec latency. Some typical numbers that I'm experiencing so far would be around: 40ms packetization delay (packet rate of 25 packets per second) 30ms transmission delay (typical b...

Dear Speex developers I am going to port Speex on LPC2368 I tested Speex encoding and the mesurments shows ~40ms cpu time for one frame Do you know who ported speex to NXP or other 32bit platform? Best Regards Zohar fox -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20081105/8261e962/attachment.htm

...-framesize 20 ~/ar1.wav ar1_20_voip.bit ./opus_demo -d 16000 ar1_20_voip.bit ar1_20_voip.pcm opus_demo reports version: libopus 1.1-alpha Using recent pesq code compiled from src, +16000 option. ( same phenomenon seen with +16000 +wb option) 5ms 10ms 20ms 40ms ar1_NN_voip 4.314 4.493 4.488 4.488 ar2_NN_voip 4.346 4.442 4.436 4.474 ar3_NN_voip 3.993 4.375 4.414 4.390 ar1_NN_audio 4.292 4.485 -> 4.313 4.313 ar2_NN_audio 4.364 4.460 -> 4.350 4.350 ar3_NN_audio 3.924 4.32...

Hi,all I have tested AEC on files, it works well.I have some files,one is echo file, others are echo-added files(an origin file adding echo at different delay,such as 20ms,40ms...120ms,140ms).AEC do wonderfully on those files except echo added at 140ms-delay. But ,when i use AEC in my voip project, it does feebly. Who can give me some hints why caused this.How long can sound be picked up by mic after it plays out.This is the main problem to align echo.Give me a hint. B...

...er buffer can be told when a packet is added that the packet contains Xms of audio, then the jitter buffer won't have a problem handling this. This is something I've encountered in trying to make a particular asterisk application handle properly IAX2 frames which contain either 20ms of 40ms of speex data. For a CBR case, where the bitrate is known, this is fairly easy to do, especially if the frames _do_ always end on byte boundaries. For a VBR case, it is more difficult, because it doesn't look like there's a way to just parse the speex bitstream and break it up into th...

Hi Kevin, Are you saying that the quality is good at 20 ms and bad at 10 ms, or 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

Hello, I've been using Opus on an STM32 M4 platform for speech coding in mono mode. I thought I'd try stereo for grins to see if I can handle the CPU load, and I'm getting a return code of -1 from opus_decode_float (using CBR and 40ms frames). I decided to try the opusenc and opusdec tools to just see how the command line apps would behave. I am getting decoded audio, but I am losing the stereo separation. My test file has speech obtained with a Tascam Recorder, and I spoke directly into each mic individually. I can clearly...

...Net stats in firebug shows each html request for the assets taking ~1s. Running same app on XP is fine and running as webrick on vista is fine too. There is also a strange pattern : each of the http requests to mongrel take marginally more than 1s, e.g. 1.04, 1.05 on webrick each request takes ~ 4-40ms. Very odd behaviour - any ideas ? weepy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://rubyforge.org/pipermail/mongrel-users/attachments/20070528/9f6a414b/attachment.html

...ical happened to the way sip traffic is handled?? I was just doing some testing and where G.711 used to use about 84kb/s of bandwith its now using between 150kb/s and 190kb/s.. The lowest I could get it to in the Snom was 160kb/s and 75 packets per second which was by setting the packet size to 40ms (was 20ms) in the Snom codec setup.. I am running a CVS version from about a month ago so I will update in the morning and see if it gets back to normal but it might be worth while to test you systems and make sure you are not hammering your bandwidth without knowing it.. Later..

...timestamp it is; this > is called arrival_margin. > The "current" timestamp is simply the last frame > successfully decoded. > > It maintains a list of bins for margins, this is short and > longterm margin. > > Think of the bins like this: > > -60ms -40ms -20ms 0ms +20ms +40ms +60ms > > when a packet arrives, the margin matching it's > arrivel_margin is increased, so if this packet was 40ms after > the current timestamp, the 40ms bin would be increased. If > this packet arrived 60ms too late (and hence is useless), the > -...

...e "current" timestamp is simply the last frame successfully decoded. Minor detail, it's the last played (whether it was successfully decoded or not). > It maintains a list of bins for margins, this is short and longterm > margin. > Think of the bins like this: > -60ms -40ms -20ms 0ms +20ms +40ms +60ms > when a packet arrives, the margin matching it's arrivel_margin is > increased, so if this packet was 40ms after the current timestamp, the > 40ms bin would be increased. If this packet arrived 60ms too late (and > hence is useless), the -60ms bin wou...

...ementation, the application doesn't even have to care about the fact that there may (or may not) be more than one frame per packet. > This is something I've encountered in trying to make a particular > asterisk application handle properly IAX2 frames which contain either > 20ms of 40ms of speex data. For a CBR case, where the bitrate is > known, this is fairly easy to do, especially if the frames _do_ always > end on byte boundaries. For a VBR case, it is more difficult, because > it doesn't look like there's a way to just parse the speex bitstream > and bre...

I''m trying out rails 3 and I''m looking at the performance statistics given by WEBrick. It says many of my database operations are taking 2ms or 7ms... but the view is taking 40ms. People often say that the database is the bottleneck in applications (which it most certainly can be, and often is)... but isn''t the rendering of ERB a little show here? My test pages really aren''t that complicated... 1 partial... 1 layout... etc. The project only has 5 model cla...