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Normalize all audio to same peak voltage?
Normalize all audio to same peak voltage?
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Old 25th October 2014, 12:03 AM   #1
SuperZanti is offline SuperZanti
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Default Normalize all audio to same peak voltage?

I'm looking to build some kind of voltage controlled amplifier that will make the peak voltage of my audio always the same.

For example if somebody plugs in an MP3 player to my system I'd like the speakers to be the same 'loudness' regardless of what the volume is set to on their MP3 player.

However, I'd like it to sample over a minute or something so that if a song is just quiet during a portion it doesn't boost the volume to create a bunch of static.
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Old 25th October 2014, 08:15 AM   #2
Ian Finch is offline Ian Finch  Australia
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Compression has been with us for a lifetime in electronic instruments, mixing and studio production but taken to the extreme of levelling out all and any valid source/programme signal will also cut the dynamic range of the music to a dull, monotonous "one level" and adds some distortion in the process, assuming you have a wide range of signal levels to deal with. DSP can be used to engineer this better but the compression issue remains.

If you want auto level setting rather than compression, you do have to sample the signal level over the whole piece, track, movement etc. to enable an appropriate single level setting. I'd say that would be impractical as an add-on for using personal devices like MP3 players.
If only I had listened....

Last edited by Ian Finch; 25th October 2014 at 08:18 AM.
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Old 25th October 2014, 10:05 AM   #3
sbrads is offline sbrads  United Kingdom
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I remember when I used to burn my old LPs to CDR there was an option to 'normalise' the A-D conversion so that the highest peak in the track didn't digitally 'clip'. This had the side effect of making music loudness fairly even between LPs without squashing the dynamics at all. So it must be possible in the digital domain but it's fraught with problems in the purely analogue domain - to hear compression at work just listen to most pop radio or internet radio to hear instruments going up and down in volume as others join in.
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Old 25th October 2014, 11:26 AM   #4
DF96 is offline DF96  England
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To sample over a minute and still be ready for peaks means that you need to delay the audio for a minute. Doing this and maintaining some semblance of audio quality means digitising, buffering, then converting back to analogue. You also have to decide whether to use an exponential law for your AGC or a 'hang' type.

The quick answer is that it can't be easily done.
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Old 25th October 2014, 12:58 PM   #5
AndrewT is offline AndrewT  Scotland
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Location: Scottish Borders
Automatic Gain Control (AGC) circuits have been around for decades.
Even very old Valve/Tube gear had AGC, especially radio tuners.
regards Andrew T.
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Old 25th October 2014, 01:47 PM   #6
DF96 is offline DF96  England
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AGC is easy.

AGC which can look ahead by up to a minute, and determine whether the signal level is a loud passage in a quiet song or a high volume setting, is hard.
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Old 25th October 2014, 02:58 PM   #7
Max Headroom is offline Max Headroom  Australia
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Location: West Australia, near to the beach, natural ambient sounds mostly.

Tired of reaching for your volume knob every time your mp3 player changes to a new song?
MP3Gain analyzes and adjusts mp3 files so that they have the same volume.

MP3Gain does not just do peak normalization, as many normalizers do. Instead, it does some statistical analysis to determine how loud the file actually sounds to the human ear.
Also, the changes MP3Gain makes are completely lossless. There is no quality lost in the change because the program adjusts the mp3 file directly, without decoding and re-encoding.
Peak normalisation is the wrong way.
Perceived loudness normalisation is the correct way.

Mp3Gain freeware is highly recommended.

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Old 25th October 2014, 11:46 PM   #8
sgrossklass is offline sgrossklass  Germany
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You could do part of a ReplayGain scanner in analog - the weighting and RMS bits at least. The rest, not so much.

Solving the OP's problem is not doable with analog electronics IMO - some AI would be handy, too.
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Old 26th October 2014, 12:19 PM   #9
owenhamburg is offline owenhamburg  Germany
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Just spent 10 mins trying to find the DSP based limiter I stumbled upon before with no luck.

It cost about the same as the "Behringer Composer Pro-XL MDX2600 Processor" which seems to be an analogue solution.

You might want to look at Fast Audio Peak Limiter By Phil Allison (Edited By Rod Elliott - ESP).

Also think about making this "anything" input balanced. It may take single ended input but you wont have the nasty humm caused if soem one touches the input connectors.

I agree with others this is better done in the digital domain especially for Audio where you can put delay in the audio chain sadly I could not find the DSP solution I stumbled upon before.
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Old 7th November 2014, 02:46 PM   #10
skippy is offline skippy  United Kingdom
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iTunes already does this. It has level normalisation option available to make all tracks have the same average loudness. Works just fine.

Interstingly the EU broadcasting industry has been forced to do this too in recent years. No more very loud ads!
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