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LOW SNR FFT

Started by Richard August 7, 2019
Hello,
I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude.
Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB.
Why not get a dynamic range of 100-110dB as is logical in a 24bit process?
Any ideas?
Thanks.
On 7.8.19 15:18, Richard wrote:
> Hello, > I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. > Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB. > Why not get a dynamic range of 100-110dB as is logical in a 24bit process? > Any ideas? > Thanks. >
The number of codec bits says nothing of the purity of the analog audio signal coming to your codec. Are you sure that the no-signal is quiet enough? -- -TV
Hi, thanks for the answer.
Yes, I'm sure, because I hear the "clicks" the signal makes when it saturates 
and the codec says that there is overflow. I use the output of a frofessional mixer.
On Wednesday, August 7, 2019 at 11:03:33 AM UTC-4, Richard wrote:
> Hi, thanks for the answer. > Yes, I'm sure, because I hear the "clicks" the signal makes when it saturates > and the codec says that there is overflow. I use the output of a frofessional mixer.
If I understand it may not be noise in the front end/mixer, but rather noise interjected by the processing equipment. What quality is the 24 bit converter circuit? What do you get for noise with no input at all, i.e. a shorted input? Are you using an eval board for the CODEC or your own board? -- Rick C. - Get 1,000 miles of free Supercharging - Tesla referral code - https://ts.la/richard11209
Am 07.08.19 um 14:18 schrieb Richard:
> I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. > Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB. > Why not get a dynamic range of 100-110dB as is logical in a 24bit process?
Calculate the RMS energy from the raw ADC output and do the same for the FFT of the signal. Normally they should be nearly equal (as long as your FFT is normalized). If the energy significantly increases by the FFT you have calculation problems in the FFT algorithm. E.g. quantisation noise. The latter happens if the FFT uses no more bits than 24 bits internally. If the RMS level of the ADC output is already that high you have problems in the analog part. Marcel
On 08/07/2019 02:18 PM, Richard wrote:
> Hello, > I am processing an audio signal with CS4244 codec at 24 bits and 48kHz. Every 2048 samples the FFT is calculated and its amplitude. > Without input signal I get FFT values ​​close to 30dB. With maximum input values ​​(overflow) I get values ​​of 90dB, with a scale of 0 to 120dB. > Why not get a dynamic range of 100-110dB as is logical in a 24bit process? > Any ideas? > Thanks. >
maybe you are missing a reconstruction filter somewhere...
On Wednesday, August 7, 2019 at 10:31:25 PM UTC+8, Tauno Voipio wrote:
> > The number of codec bits says nothing of the purity of > the analog audio signal coming to your codec. > > Are you sure that the no-signal is quiet enough? >
If you are talking about audio (20 kHz) then you will only have 90 dB SNR in the recording. If you go to seismic (128 Hz filter) then you would not get 120 dB, due to various noise from transducer and pre-amp.