search for: 7khz

Thanks! Btw, have you tried using SBR-technology or similar with speech codecs? That might be a good idea I thought.. But I don't know if it produces as good quality with speech codecs as it does for music codecs. Do you know if there is any open source variant of SBR? /Pontus -----Ursprungligt meddelande----- Från: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]För Jean-Marc

...link.ca/~amstereo/amstereo.htm> for more information. > Telephone is nominally 8 kHz mono > (i.e. really bad) though I think the use of digital voice codecs in the > last 20 years may have improved on this a bit. Telephone lines (POTS) have a frequency range of 300-3400Hz. That means 7kHz mono should be enough, although 8kHz is generous towards the transition bandwidth/roll-off. - Andrew

...if content above about 10kHz is properly harmonically related to the lower pitched fundamentals which usually give rise to them. I don't know of any voice specific coder that even attempts to capture energy above 10kHz. SBR just isn't relevent. Most wideband speech coding captures only 7kHz to 8kHz bandwidth. The key improvement that gives over the 3kHz to 4kHz most mainstream voice coders capture is to clean up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear almost the same at telephone bandwidth. At 7kHz bandwidth they have enough character to make them more dist...

...s asterisk translating this (btw. how do faxes work from europe to north america - the telcos have the same problem)? - which signalling protocoll will be used on the T1 side? is asterisk translating this correctly? - btw. where is the different bitrate coming from? is it 7bit T1 and 8bit E1 or 7kHz and 8kHz sample rate? regards frank sautter

...ile as possible, and a high audio quality demonstration of the difference in pronounciation of a "D" in English, Spanish, Chinese, and German, using as much space as needed to make the difference sound clear. My current thought is to filter the input down to a bandwidth of 7KHz or 4KHz (traditional values for high and low quality speech), decimate the samples so that the sound is sampled at, say, 44/3=14.6KHz or 44/5=8.8KHz, then run it through the standard Vorbis encoder. Vorbis then sees an ordinary 20KHz bandwidth stream that sounds like a tape recording running at 3...

...0kHz is properly harmonically related to the lower > pitched fundamentals which usually give rise to them. > > I don't know of any voice specific coder that even attempts to capture > energy above 10kHz. SBR just isn't relevent. Most wideband speech coding > captures only 7kHz to 8kHz bandwidth. The key improvement that gives > over the 3kHz to 4kHz most mainstream voice coders capture is to clean > up unvoiced sounds. fffff, sssss, and other unvoiced sounds appear > almost the same at telephone bandwidth. At 7kHz bandwidth they have > enough character to...

Hello all, This is slightly off-topic, but should be of interest to some people on this list. I just released version 0.3.2 of the CELT ultra low-delay audio codec (http://www.celt-codec.org/). CELT is designed to encode high quality speech and music with less than 10 ms delay and at rates starting from around 32 kbit/s. This version is "special" in that it is the basis for some

...Now, when trying to visualize the weights to see a bit of what was going on, I also computed the phase for each frequency bin. When looking just at the phase, I can see a very clear and distinct pattern of going from -pi to +pi in the areas where I know there is echo (specifically, the lower 7khz of j==M-1), and what looks like random noise for the rest. Do you have any idea where this pattern originates from, and more importantly, could it be used as additional conditioning of W? (ie: if the phase doesn't match the pattern, reduce the amplitude as it's a false match).

Hi Jean Marc, I was very impressed by the comparative results. Could you give a bit more information about what really are 7kHz and 3.5kHz in the 48kHz table? I am looking forward that this will bring a bigger separation between the basic tools and speex in a sense that the basic tools could be used with more than one codec. I even understand that there is interest in the tools alone (Echo supression, ressampler, Jitter...

[ > Telephone is nominally 8 kHz mono (i.e. really bad) though I [ > think the use of digital voice codecs in the last 20 years may [ > have improved on this a bit. [ [ Telephone lines (POTS) have a frequency range of 300-3400Hz. That [ means 7kHz mono should be enough, although 8kHz is generous towards [ the transition bandwidth/roll-off. 7 kHz would require a tighter brick-wall filter than even CD, are you even sure it's possible to go from flat to silent in 100 Hz (between 3400 Hz and 3500 Hz)? Almost all telephone connections...

...to visualize the weights to see a bit of what was going > on, I also computed the phase for each frequency bin. When looking just at > the phase, I can see a very clear and distinct pattern of going from -pi > to +pi in the areas where I know there is echo (specifically, the lower > 7khz of j==M-1), What you see is a "linear phase", which is the frequency equivalent of a delay in the time domain. So basically, the phase you see is just the representation of where the "main impulse" is in the time domain version of W (i.e. the time offset between the two signal...

The G.729.1 "wideband" codec is starting to show a slight bit of traction. There is a possibility that Asterisk could support G.729.1 - would you use it or buy it if it was available? More importantly, does any equipment with which your systems currently exchange traffic support G.729.1? Currently, the number of devices supporting G.729.1 seems to be fairly limited and it

Is there a list somewhere of "standard" encoding rates? I know, for example, CDs are encoded at 44100, as is a lot of digital sound, but I've seen programs that specify different levels of quality (like radio, phone, tape, CD) and I'd like to know if there are some encoding rates that are accepted as standardized for recording at different levels of quality. If so, is there

Hi, I'm still working on visualizing the echo canceller, but I discovered something that might be interresting. During testing, i did this: Generate a test signal (10+x sine waves per frame), where x increases by one for each iteration, and wraps around at 100. Set the speaker signal for the frame to the test signal. Add 0.5*test signal to the mic signal. When watching the power graph

...the weights to see a bit of what was going >> on, I also computed the phase for each frequency bin. When looking just at >> the phase, I can see a very clear and distinct pattern of going from -pi >> to +pi in the areas where I know there is echo (specifically, the lower >> 7khz of j==M-1), > > What you see is a "linear phase", which is the frequency equivalent of a > delay in the time domain. So basically, the phase you see is just the > representation of where the "main impulse" is in the time domain version > of W (i.e. the time offset...

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to