Can someone confirm my understanding of digital sample rate?

Hammarlund

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Model 12
This is really my first digital workflow, just making sure:

1) I know the model 12 is digital. So if I run input through a channel and out the main, there is an internal A/D/A conversion. This also shows up on the block diagram.

2) The analog fidelity of the main outs is therefore limited to the fidelity (max 48/24) of the internal A/D/A. It can't play frequencies which it didn't record.

3) So there is NOT any benefit from (for example) picking up the main outs w/ a 96/24 or higher interface. In fact there is a cost: all it does is add another useless conversion without adding any data.

4) If I want to use T12 and change to a higher sample rate before working in a DAW (for example, before working w/ effects), I can use T12 files and easily upsample/convert to a different multiple of 48/24. This won't add any data or resolution to the base recording, but it may be useful to have a higher resolution for the effect outputs, etc.

5) If I really want to capture something natively higher than 48/24 while still using the T12, I can do one of two things:

a) put the high-rate interface in front of the T12, and pass the audio through it first to capture, using the "input monitoring" outputs to then run to the T12; or

b) plug directly into the T12, and use a half-plugged-in "insert" cable to copy the output for high-bit-rate recording elsewhere.

Amirite?
 
Here's the basics:
  • 16bit = 96db dynamic range/headroom/transients= professional recording studio ambient noise level to full orchestra very loud.
  • 24bit = 104db dynamic range/headroom/transients= below hearing threshold of most humans to standing next to a jet engine at full thrust. Also allows lower recording levels that can provide greater margin of error for those pesky transients common to amplified instruments and drums.
Sampling rate = the number of analog audio wave cycle points being sampled per second, yielding statistical sampling accuracy (faithfulness) to original source material (e.g., 44.1 kHz means 44,100 samples per second. The accuracy is fixed at the defined initial sampling rate of the recording.
  • There's no audio advantage to upsampling and in fact upsampling can introduce undesired artifacts.
  • Recording at higher sample rate provides certain advantages when downsampling, particularly to lossy formats like mp3.
 
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2)/3) Yes to 2). However, if you have a better quality DA-converter, theoretically you could go from Model 12 digitally to this unit and then to analog. Every AD and DA converter has an internal rate that is much higher than the audio it is letting through (and different quality components), so there ARE quality differences, even with the same sample/bit rate.

4) True, though for the average DAW (currently most are at 64-bit resolution internally) upsampling has no real benefits. Mostly mastering engineers do upsample, also because they use processing where this matters - and they go from their mastered files tio different production outputs (they might need to deliver a 96k-24bit master anyway - next to a 320 kbps mp3).

5) I don't see why you'd want to capture both high and low SR material at the same time - why not record at the high SR and then downsample wav files to enter into the Model 12?

@Mark Richards I know you wrote 'simple answer', but IMO only qualifying bit depth to dynamic range only is too simple. A higher bit depth gives us the possibility to record at lower levels because we have higher resolution: Each sample written in 16 or in 24 'steps'.
 
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@@Mark Richards I know you wrote 'simple answer', but IMO only qualifying bit depth to dynamic range only is too simple. A higher bit depth gives us the possibility to record at lower levels because we have higher resolution: Each sample written in 16 or in 24 'steps'.

Agree Arjan. Headroom is also important for handling transient peaks. Made some edits. Thanks.
 
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