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First, I think we should probably address the concept of volume. The level of a signal has nothing to do with volume. You don't have volume until it is actually put through speakers and you are a hearing it a certain distance from those speakers. Instead you have a "level".
The level is a measure of signal strength and it isn't subjective, but rather ...
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 ...
5
Sorry to disagree with Stavros, but in my humble opinion, it really depends on the room - 85 as a dolby reference is intended for rooms with a volume of 150 cubic metres or bigger.
In a typical edit suite in the near field, you wouldn't normally want to monitor any higher than -20dbFS = 79dbSPL IMHO. Anything louder and you'll probably be under-cooking your ...
5
Editorial calibrated to -18 or -20 dBFS = 85 dB SPL, Dialogue should be hitting a LM100 at -27dB LEQ(A) or -24 via a Stereo Phase Scope (but still about -27 when monitored via a Center channel Phase Scope if you're running a 5.1 mix bus - the -24 is due to a +3dB in-phase summing thing) - which when the room is calibrated to -18 or -20, should sound loud, ...
5
The difference between clip gain and volume automation is basically INPUT volume and OUTPUT volume. Clip gain is the input volume. This data is read and taken into account before insert effects, sends, volume automation, eft. Here are two examples : if you crank up the clip gain you could potentially clip your audio. If it Wasn't clipping while you were ...
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
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 ...
4
I usually have my Dial editors only do very basic level adjustments. Basically lower any really loud pfx or a line that is dramatically out of sorts with the rest of the scene. With noise reduction, I never want my editor to do any broadband NR or notch filtering. You can get into a lot of unnecessary processing that might not be needed once everything ...
4
It depends on the agreements made between the dialogue editor/supervisor and the sound supervisor. Mixers could prefer to not have volume automation, but this differs.
Why do you ask? Are you editing dialogue? Or are you mixing the movie?
Recommended reading:
Dialogue-Editing-Motion-Pictures
4
Radioshack makes a nice cheap and reliable SPL(sound pressure level) meter (the technical term for a db meter). I've seen that particular one in many a road case and I recall it only being something like $40. As a point of reference, I generally try to mix my church's services for between 90 and 96db during worship and that's plenty loud for our young (...
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
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 ...
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
You are right!
The quality of the audio will be decreased if the volume lowered by software means. As you have correctly assumed, reducing volume in software is actually similar to reducing the bit depth.
Generally, every 6 dB of attenuation is equivalent to reducing the bit depth by one.
Brief explanation: Max volume level of your sound card output, ...
3
Excellent explanation Steve. I'm going to second Brent on SPL according to room size, but it is a great response otherwise. I've got a break down on the recommended SPL level vs. room size in a post I did about monitor calibration on my site, Jellan. You can check it out here.
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
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
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
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).
3
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.....
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
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 -...
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