8
It's important to remember that a compressor does not make a sound louder, it reduces dynamic range. By bringing down the peak values, you give yourself more headroom to bring up the overall signal level (so the quieter parts can be louder).
It is that "make up gain" which makes a sound louder, not the act of compression. The choice between using ...
6
If you want to be technically accurate, a sound file does not have a volume, it has a level. Volume can only exist when there is an actual speaker producing sound as that has a fixed SPL(sound pressure level). A signal can be played back at any volume provided the speaker is able to produce it at that volume.
Instead, it is referred to as signal level ...
5
how far away? is your macbook air connected to speakers?
Your file has a "loudness" (RMS level) -> player volume (gain) -> system volume (gain) -> audio interface voltage specs (gain/level) -> amp (gain) -> speakers (sensitivity). So it's impractical to try and "compute" it with that many variables. Grab an SPL meter.
There are SPL meter smartphone apps ...
5
Practical version
dB → gain-multiplier:
g = 2d / 6
gain-multiplier → dB:
d = 6 · log2(g)
I find these definitions far more handy than the ones below: changing the amplitude by a factor of two is quite an intuitively relevant change. But, alas, in pre-computer times people couldn't seem to like logarithms of bases other than ten, so.....
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
I doubt this is really an error – as this interface is USB-powered, the designers probably spec'd the headphone output with rather low maximum voltage, so as to avoid supply problems; I know at least the old Tascam US-x2y series suffered from that problem. For consumer headphones, which are targeted at low-voltage battery-supplied devices and hence ...
4
As you don't mention which platform/OS you're running, I would suggest loudness-scanner.
Something like :
loudness dump -m 0.1 *.mp3 > data.txt
should provide you with a text file containing raw data (momentary loudness every 0.1 second).
4
Personally, I prefer tutorials without music. I also think a bit of background noise is not an awful thing. Gated dialogue or over-baked noise reduction is far more distracting. Choosing a mic you can get close to, thereby increasing the signal to noise ratio will help a lot. Also consider building a makeshift sound booth to block some of the noise out.
4
Phase cancellation is a common cause of confusion for many audio professionals, sometimes this is caused by a channel becoming inverted for some reason. Maybe this answer can help you, and other people reading this, to identify, or eliminate this type of phase cancellation as the cause of a problem.
I've analysed the audio from the video and have come to a ...
4
To move the whole automation, you have to select the entire area of the automation first, then move it.
BUT..
This is a common problem, that pretty much everyone solves the same way.
Do not under any circumstances automate the final volume of a track...EVER.
In Ableton you drop a utility effect on the track and use it to do volume automation. This way ...
4
ReplayGain tags aren't standard in WAV files, so you have to alter the PCM data with the required gain.
As per my reading of the Replaygain specs, a correctly implemented Replaygain scanner will print out the gain required to attain 89 dB SPL (as defined in the specs).
FFmpeg has a filter to detect replaygain. You can run
ffmpeg -i in.wav -af replaygain -...
3
Audacity does normalising but it sounds like what you want is a peak limiter. Try the Yohng W1 Limiter, which is modelled a a well-known and (possibly overused) plug-in. It doesn't look all that, but it does the job nicely.
You should also try a compressor, but you'll need to use it really carefully on classical material.
3
Normalisation only raises the overall volume uniformly (the soft parts will be louder, but so will the loud parts). What you're looking for is dynamic-range compression, and I think there's a plug-in for that in Audacity. I'd advise you to learn a bit about the different controls on a compressor before you get stuck in.
3
Probably, yes. "Headphone power" depends basically on two things, output voltage and headphone impedance (I discussed it in some detail here). Many devices don't bother providing a very great voltage, but for dedicated headphone amps it's pretty much expected that they can make any off-the-shelf headphones loud enough for all typical use cases.
Still, if ...
3
Integrated circuit power amplifiers reach a point of negligible incremental return quite low on the cost curve. From a reputable source you are not going to see a measurable improvement beyond $20. It's an extremely mature technology, about the oldest and most active of any electronic technology. Correlating the quality of a headphone amp with price is ...
3
Change is typically "disturbing". A sharp change of amplitude (a crash, a scream, an explosive sound, etc.) Or even sudden quiet after moderate but steady background noise will disturb some people. So a circuit that detects any sharp change (up or down) in sound amplitude. That would be my primary focus for detection of "disturbing sound".
Of course, ...
3
It really depends on the material; are these sounds that you want to remove truly peaking at a lower level than every bit of the speech? How about in loudness or 'energy'?
You need to look into the kind of sounds you want to remove and see how they differ from the ones you want to keep. This is key to being able to separate them.
Are they of a shorter ...
3
Gain-staging without meters…
Turn your output device [in this case your phone] up gradually, turning down your listening levels as you do this to keep the sound comfortable.
If you start to hear distortion from the loudest signals, turn the output device back down by maybe 20%.
Repeat for each stage in your signal chain.
This will drive every stage at '...
2
Well, to be perfectly fair, you get what you pay for with these things. I think $20 is not a reasonable expectation for a headphone amp at all. Headphone amps are definitely in the professional sound market, and as such, they command both a certain build quality and price range. For any decent headphone amp I would expect to pay at least $100-200 minimum, ...
2
The dB level depends on what you are playing and how far away you are from the speakers. The only way to get an accurate estimate of the SPL is to use a SPL meter.
2
You could use a gate to try and remove as much of the background noise as possible. If the background noise is in the low frequencies, try low-cutting the commentary with an EQ or filter. Audacity also has some good tools but it depends on the timbre of the background noise as to whether they will work. As for the levels, you should have the soundtrack quite ...
2
Thanks for your answers. This is what I ended up doing:
I found out you can buy a splitter for a mic (called an "isolation transformer", $40) which allows you to use two preamps (in other words it allows you to send the mic signal to two places, one hard wired and one that's "isolated", basically "spying on" magnetic noise from the other line). I split the ...
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 ...
2
I'm honestly not sure what you are asking here. Compression does not increase the "volume" of a signal, it decreases it. Compression makes a quiet portion of the sounds louder relative to a louder portion by reducing the signal strength when the signal strength is high.
Often a gain is applied after compression to keep the signal strength up, but this ...
2
This might be due to a phase problem,
With your headphones and speakers the audio source is still really near so you fully enjoy the space emulation without loosing level.
When you are far from your speakers a lots of frequencies might be cancelled and impossible for you to hear.
The 3rd voice (man) is Ok other voices have a bad phase.
Try to switch the ...
2
There is a “Velocity” MIDI effect in Ableton Live, with a patch called “Add some random.” Place this MIDI effect onto the channel that you want to randomize.
2
Basically, you're right with all your points.
The level between different albums can indeed be dramatically different, depending on how agressively it was mastered. Records from the early days of CDs are often very quiet (engineers celebrated not having to worry about vinyl's noise floor anymore, and also, early CD players didn't really work reliable with ...
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