2009-04-22 12:22:35 AD73311 bandwidth issues and Actual Sampling Rate
Rob Maris (GERMANY)
Message: 73071
When using the AD73311 in standard 8 kHz sampling rate mode, there is some lack of brilliance in any speech signal which is recorded resp. played back.
According to the data sheet, it is recommended to use 64 kHz actual sampling rate in order to get more raw bandwidth, which - for the recording section - should be followed by a digital filter implemented in DSP prior to converting to 8 kHz sampling rate.
For the sake of an experiment, I have modified the AD73311 drivers so as to operate at 16 kHz. While doing subsequent arecord and aplay commands I found that recording with the default 8 kHz rate indeed yields a brighter speech upon playback. However, it can also be noted that hiss noise tends to get audibly transposed to a lower frequency. This shows that the automatic ALSA rate conversion does no filtering (or at least not sufficient filtering).
I'd like to obtain hints and/or info pointers about how to establish bandwidth&anti-alias improvements according to the proposed method, so as that the converted 8 kHz rate available to the user program is already filtered at driver/ALSA level. Or: are any resources readily available?
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2009-04-22 12:48:26 Re: AD73311 bandwidth issues and Actual Sampling Rate
Robin Getz (UNITED STATES)
Message: 73074
Rob:
For the ALSA details - the best place to check is the ALSA mailing list. (just rememebr to tell them you are doing things on a fixed point processor).
-Robin