5
What you are asking is impossible to determine from a spectrogram. Spectrograms indicate frequencies present, not the quality of the audio. There may be loss of fine detail in one that would barely show up on a spectrogram at all or there could be noise and artifacts introduced that would make the spectrogram look more full.
Spectrograms do not ...
5
It depends what method is used.
If it's simply played back 'faster' without actually changing the data, then of course there is no lasting damage. This would be how a simple sampler would ordinarily do it, by changing the speed, ie data rate, of the playback.
If you resample so the same amount of audio plays back in half the time at the same sample rate - ...
4
First, lets start by clarifying that "volume" isn't a particularly technical concept. Most generally "volume" refers to SPL or sound pressure level which is the amount of pressure being exerted by sound waves and depends on the distance from a sound source.
Rather what we are talking about here is the impact of signal level. Within an analog system, the ...
4
If the mix is not the final production stage, you should favor to preserve all information available and avoid introducing artifacts. So if you will master or further process the signal, sampling at 96k could help. Keep in mind the difference will often be imperceptible in simple cases, and the error would be very small if you are using ideal hardware and ...
4
Audio sampled at 44.1 kHz (like normal audio CDs) can in theory contain content up to the Nyquist limit of 22.05kHz. However, you need a filter to remove all content above that limit, otherwise it folds back into the hearable range: you hear this as aliasing.
A perfect, theoretical filter would remove all content above 22.05kHz and leave everything else ...
3
Audacity will happily output 8 bit wave files. Choose export > other uncompressed then MS wav and 8 bit unsigned.
3
That depends on what you mean by audible. You can invert the phase of one and add it back to the other and it will play only what the difference between the two is, however you will hear artifacts that might not have been detectable to normal hearing in the original file since they were previously buried underneath other sounds.
It is also important to ...
2
The relevant question is what you're going to do with the mix yet. Distributing a final master at more than 48 kHz makes no sense whatsoever. The only reason it can be useful to record at higher rates is to avoid aliasing issues in any nonlinear effects plugins.
That certainly includes mastering compressors, so if your plan is to feed the digital mix to ...
2
I believe this article sums it up far better than I ever could. And I like it because it doesn't seem to lean to either side of the "sample rate war". I recommend everyone read it of you have the time.
http://www.trustmeimascientist.com/2013/02/04/the-science-of-sample-rates-when-higher-is-better-and-when-it-isnt/
From my personal experience, record at ...
2
The upper one is definitey the better one.
Spectrograms (like the ones you produced with Spek) are used to display and inspect tendencies over time. For example if some noisy signal is present all the time, it will show up as a straight horizontal line somewhere:
You can determine some general properties, like that 20 Khz cut, but an average frequency ...
2
Is it possible to do so? Absolutely. Lasers are one of the fastest and most versatile transmitters around. Fiber optics, which works by shooting a laser down an optical fiber, is the bread and butter of modern data communications and literally makes the modern Internet possible. Almost all of the fastest connections (internet backbones and residential ...
2
Generally no, using multiple inputs shouldn't lower quality as long as the USB controllers still have sufficient bandwidth for the devices to function. If you use a hub to connect everything, then you may run in to data rate issues causing latency or outright failure, but in general, the data is digital and should get from point a to point b alright.
There ...
2
The biggest problem with combining your mixer and interface is that most of the cheaper options have sub-par pre-amps, which would definitely affect your quality. I would personally recommend investing in a quality audio interface (a great option would be focusrite) with solid pre-amps, and then later you could invest in a cheaper mixer.
2
Audible differences is something different than electrically measurable differences. The techniques mentioned (by computing the difference between the two signals) allow to measure differences between two signals but don't give real details about the perceived difference.
The correct way to evaluate audible differences is to setup a panel of listener, and ...
2
Best audio quality is difficult to assess objectively.
What you can say from the given audiograms is (as you mentioned) that N° 3 includes more high frequencies, which is usually considered better.
About the N° 2 audiogram : there's usually no benefit in transcoding a lossy file into another lossy file codec because whatever information has been lost in ...
2
As far as digits are concerned, you can multiply by any "gain" you wish, as long as you do not overflow (-127 to 128 for 8 bit; -32676 to +32678 for 16bit audio; ±8.3x106 for 24bit audio, and up to ±1.7×1038 for 32bit floating point). However, there will be some rounding involved, and the fewer the bits the more the rounding error.
Note that whatever the ...
1
It may make a difference.
Your sound card (and whatever else comes after your pre-amp) will inevitably add a certain level of noise to the signal itself, as will the pre-amp itself.
If your pre-amp provides a very low level of noise, then increasing the gain will result in a better signal to noise ratio later on in the chain if there is noise being added (...
1
As the recording sounds perfectly "within acceptable parameters" on my studio monitors, maybe either your piano sound in the room is 'too fat' & you have just got used to how it sounds... or your computer speakers sound 'too thin'.
This is intentionally vague, because we can't actually hear either source in the same way you do.
If you are using a ...
1
TL;DR: It appears the iPhone 6 supports 24-bit audio and 48 kHz playback.
So I found a way to empirically test playback.
This first screenshot shows a recording of playback over iPhone USB into Quicktime. You can see from the spectrogram that there is data in the 22-24k range, which means that playback is at least 48 kHz.
This next screenshot shows ...
1
Unlike some earlier models of iOS devices, the DACs and ADCs in the iPhone 6s and 7 support native sample rates of 48000. You can experimentally test for this by writing an iOS app that configures the RemoteIO Audio Unit for a 48k sample rate, then synthesize sine waves at frequencies a bit above 22.050kHz and observing the audio output on an oscilloscope.
...
1
Bit depth is specific to certain file types, most notably Wav. Because the standard procedure of a music application is to play a compressed version of a file such as Mp3 or Aiff it throws bit depth out the window and instead outputs signal at a kbps rate instead such as 192 or 320kbps. As far as DAC goes I don't think it will improve the quality of an ...
1
According to the man page, --bps specifies the bits per sample. As far as I can tell, this parameter controls the data that will be encoded using FLAC, and can have a lossy effect. For example, here's a comparison of a short audio clip using 8, 16, and 32 bits:
You should never use 8 bits unless you're going for an effect, ...
1
Yes, you ALWAYS lose quality when open a file created with lossy compression (i.e. virtually all flavors of MP3) It may not be very much, but it is unavoidable. That is why it is called "lossy".
HOWEVER, when you RE-compress the track back through another lossy compression process, you run significantly more risk of loss. Because you are applying the ...
1
There are several major ways a signal can become distorted, these impact both digital and analog signals, but in different ways.
One of the largest sources (on long runs and at connectors) is attenuation, which is the loss of signal strength due to resistance on the line resulting in electricity converting to heat. It generally impacts the signal in a ...
1
Reading the above, I don't think that your synth, or the interface is the issue.
Running a synth directly into an interface normally just gives a pretty boring copy of the original tone. When I'm recording synths, I like to run them through an amp, any amp, and mic the output.
Depending on the desired result, running through a small bit of overdrive, ...
1
That means you are on the right track. On high volumes our ears have a flatter response. That means that on higher volumes, we start to hear more bass and more highs.
So a very simple solution to your problem might be a wrong gain/frequency structure in the bass and highs which can lead to extreme imbalances on higher volumes. Since your song sounds good on ...
1
I hear all of you on this one, especially the OP. I can also tell a difference... ...Here's the deal: I grew up nearly deaf (and inching towards deafness). Sound didn't matter until a surgery gave me back full function in one ear. The other is bad, but with special equipment, can work up to 89% efficient, which, I'm told is average human quality for my ...
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