Frequently Asked Questions...
what are the optimum sound card features for computer that I should buy? For listening and recordiing music.?
I use the computer for converting cassettes to CD, recording music and listeningto it. I wonder what kind of sound card should I opt for. There are so many of them in market.
One of sound card specifications are as below. Is it good enough. Should I go for 96KHz(Analog-to-Digital ) or look for higher frequencies.
TECHNICAL SPECIFICATIONS:
- 24-bit Analog-to-Digital conversion of analog inputs at 96kHz sample rate
- 24-bit Digital-to-Analog conversion of digital sources at 96kHz to analog 7.1 speaker output
- 16-bit and 24-bit recording with sampling rates of 8, 11.025, 16, 22.05, 24, 32, 44.1, 48 and 96kHz
- SPDIF output up to 24-bit resolution at selectable sampling rate of 44.1, 48 or 96kHz
- 64-Voice polyphony and multi-timbral capability
- 128 GM & GS compatible instruments and 10 drum kits
Any help will highly be appreciated
Purohit
Answer:
sounds like a sound blaster audigy 2 with some extra goodies. that should be more than sufficient for your money because cd quality is 44.1khz at only 1bit dac. if you can afford it, go fer broke with the sound blaster x-fi series. check out the web site and don't pass out. don't forget to look at the signal to noise ratio when recording, though! the audigy 2 is pretty good though, 96dB. the x-fi is 116dB! s/n ratio is how many decibles higher the intentional signal is compared to any background electronic noise. the decible scale is logarithmic, which means that every increase in one is actually 10 times greater.
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Audiophile Sampling Rate Conversion
Recording studio and audiophiles demand high quality of finite sound for sampling rate conversion (SRC). SRC widely using for preparing master records publications on CD and DVD.
Music home equipment of finite users basically have 2 ranges of sampling rates (SR):
- audio CD: 44 100 - 88 200 - 176 400 Hz
- audio & video DVD: 48 000 - 88 200 - 96 000 - 176 400 - 192 000 Hz
SRC intend is not only for professional musical studios. Also it used for home audio and video.
Sampling rate conversion algorithm is:
I) Increase SR to multiple target signal's sampling frequency with interpolation.
II) Filtration of "parasitic" signals, placed above half of output sampling rate.
III) Decreasing SR ("killing" of superfluous samples). For SR increasing between "real" samples of input digital signal insert virtual samples, created by interpolator. Even accurate algorithms of interpolation create inevitable "parasitic" signals ("artefacts"). For decreasing CPU loading make inserting zero virtual samples. But such method generates additional "artefacts" (more then accurate algorithms). To reduction "artefact's" level, interpolation of input signal should possible precisely. Interpolation's accuracy is maximal degree of coincidence of initial analog (not input digital!!!) signal with "virtual" samples of interpolator. Top-quality interpolator can restore an initial analog signal precisely enough. To execute algorithm item III necessary to increase sampling rate. Rejection of samples most easier for multiple sampling rates.
"Artefacts" with frequencies above half of target SR are filtered. If don't do it, after deleting of "superfluous" samples "artefacts" will mixed with clear signal and will "dirty" sounds will added. Special attention is paid to quality of low frequencies filter during development of hi-end sampling rate converters. If this filter will not hard suppress "artefacts", it will mixed with useful signal after deleting of superfluous samples.
For hard suppress filters: more "sharp" recession to a level between passbands and suppression band of filter, have more level of "ringing".
"Ringing" is parasitic filter's output self-oscillation after sharply changing of input signal (input impulse). Output impulse will oscillate some time. "Ringing" is heard as click. For converter's developers necessary to choose the compromise between a sharpness of recession of filter's amplitude-frequency characteristic (that in the most positive influences to suppression of "artefacts") and level of "ringing". Very heavy case, when final result of conversion is sample rate 44,1 kHz. Between the maximal frequency of a useful signal (20 kHz) and half of frequency of digitization (22,05 kHz) difference only 2,05 kHz. For desirable degree of suppression of artefacts nearby 140 dB!!!
For hi-end sampling rate converters level of artefacts should less minus 150 dB. It less quantization error level for 24-bit samples (minus 144 dB). Inevitable quantization errors of output digital signal will cover "artefacts". For middle quality sampling rate converter having level of "artefacts" near minus 105-110 dB. "Artefacts" arise, both at interpolation, and at insufficient filtration of "parasitic" signals located above half of sampling rate before SR decreasing. For low quality SR converter "artefacts" achieve a level near minus 50-60 dB.
Filter should pass audio signal frequency range 0 - 20 000 Hz without useful changing of loudness. For this un-flating of the frequency characteristic (for various frequencies at passage through the filter) should not exceed difference of a level of loudness of output signal less 1-2 dB. To keep converted audio signal's waveform it is required for providing of identical time delay for all spectral components (frequencies) passed through the filter. It is provided for filter having linear (like flat inclined line) phase-frequency characteristic. Signal passes through such filter are not distorted. It is especially important for maintenance of quality sounds with sharp attack (drums, piano, guitar, etc.).
Conclusion
Quality of sampling rate conversion depend at: - quality of interpolation; - value of oversampling frequency; - quality of filtration before decreasing of sampling rate.
For modern records 24-bit samples is used. It achieve theoretical level of quantization's noise minus 144 dB. Accordingly the level of all transformation's "artefacts" should not exceed minus 144 dB. Thus "artefacts" will masked in noise of quantization.
More hard "artifacts" supression need for 32-bit and more (integer or float).
Quality of converting signal is provided with filter's linearity phase and flating of frequency characteristic.
About the Author
Audiophile Inventory corp., audiophile and professional music production software developer
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