search for: passband

...hing I found speex, but before I try to engineer the solution, I'd like to get any advice on use of speex in with the expected source material, which is likely to be noisy (static and stuff mixed in with the source audio). The source audio (monaural) will be pre-filtered to fit with 300-3000 Hz passband (can be slightly narrower if need be), and may not always be a single voice (that is, may be>1 voices interfering with the audio passband, or even non-voice such as tones and other stuff that would appear in the passband of the receiver.) So based on this, would I want to avoid speex or proceed...

...to engineer >> the solution, I'd like to get any advice on use of speex in with the >> expected source material, which is likely to be noisy (static and >> stuff mixed in with the source audio). The source audio (monaural) >> will be pre-filtered to fit with 300-3000 Hz passband (can be >> slightly narrower if need be), and may not always be a single voice >> (that is, may be>1 voices interfering with the audio passband, or >> even non-voice such as tones and other stuff that would appear in the >> passband of the receiver.) So based on this, wou...

...ow can anyone briefly describe what type of filter I should set up (FIR, IIR, all-pole), why one is better than the other, and if you have filter coefficient files lying around (lowpass, 19 or 20 kHz cuttoff) PLS forward them. I'd like a 19kHz 'brick wall' cuttoff with no ripple in the passband. : ) >From my undergrad engineering work I vaguely remember about 'too steep a cutoff gives you a lot of ripple in the passband', and other such design homilies about RC filters. What I'm looking for here are some similar guides/rules of thumb for current digital filtering apps (AF...

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

...iving the speaker) is analog and contiguous in nature. On other hand, all codecs (and everything in DSP) works on discrete signals (samples). When you convert from discrete samples to analog signal, you need to apply an ideal filter, which has a frequency response of 1.0 (0 dB attenuation) in it?s passband and 0.0 (?infinite dB) in its stopband. The most important thing to understand here is that such an ideal filter does not exist in reality. What we do (most DACs do this) is to approximate such a filter using IIR?s or an FIR with some delay (sufficient delay that the truncation of impulse response...

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to