Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Stuart Longland
On 20/09/15 03:43, Tomas Härdin wrote: > Couldn't you do further low-pass filtering in software, then decimate to > 8 kHz? (caveat: I haven't checked if the code actually does this) That's exactly what it does. It samples at 16kHz from the ADC, filters it to produce an 8kHz stream. Actually a

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Brady O'Brien
I don't think a linear interpolator would work here. There's quite a bit of attenuation in the 0-Fs/4 band (~8dB by Fs/8) and only around -28dB on the aliases. -28dB might be enough for the aliases, but I don't think it's flat enough in the Fs/4 band to be useful.

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-19 Thread Tomas Härdin
Couldn't you do further low-pass filtering in software, then decimate to 8 kHz? (caveat: I haven't checked if the code actually does this) /Tomas On Fri, 2015-09-18 at 16:13 +1000, glen english wrote: > Hi Stuart > > An elaboration on David's reply. > > We want our audio BW to go to at least

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread glen english
Hi Stuart An elaboration on David's reply. We want our audio BW to go to at least say 3.4 kHz . That is what fixed line telephones offer. If we were to sample at 8kHz, our nyquist frequency is 4kHz of course. Any spectral information above the

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread Stuart Longland
Hi David & Glen, On 18/09/15 16:13, glen english wrote: > If we were to sample at 8kHz, our nyquist frequency is 4kHz of course. > Any spectral information above the nyquist rate will be aliased bay into > the baseband, IE below the nyquist rate. > > IF we assume we don't care about aliases

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread David Rowe
Also there is very little speech energy out past 12 kHz, i.e. its already 50dB down, and this is communications quality speech so a little aliasing is lost in the codec artefacts. On the radios interface side the audio is band limited by the radio's frequency response, ie the xtal filter BW.

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-18 Thread glen english
yeah you can pretty much get away with NO aliasing filter considering the source ( a band limited microphone) and the required voiceband SN. still, good reason to have at least a single RC- from say switchmode power supply whistles etc getting back into the input, and aliasing down into the

Re: [Freetel-codec2] Silly question regarding SM-1000 sample rate

2015-09-17 Thread David Rowe
Hi Stuart, Running the ADC/DAC at 16 kHz largely removes the need for analog anti-aliasing/reconstruction filters, significantly simplifying the hardware. - David On 17/09/15 19:23, Stuart Longland wrote: > Hi all, > > Just looking at the source code, it hit me. When in analogue mode, we >