Recently I have been making podcasts which consist largely of two to four people talking. I am a complete newbie to the process, and I know little or nothing about audio production.

Volume is something that has always confused me. First, when recording, I can adjust the gain of the microphone input. Then in Audacity I have a million different ways of adjusting the volume. And then ultimately when a listener downloads and plays my podcast, they can adjust the playback volume to their tastes. There are all these stages which impact the final volume, and it seems to me they're all measured relative to each other, and thus lack any kind of objective measure.

I'd like to be consistent among my podcasts, and also hopefully fall in line with some kind of standard so that my podcasts aren't wildly louder or quieter than most other things they listen to.

But it all baffles me because it seems that listening volume is so relative and subjective.

Is there a way to determine a good output volume that isn't just based on trial and error and feel?

Please bear in mind when answering that technical terminology is likely to be too advanced for me to follow.

  • 1
    possible duplicate of How do you use gain effectively?
    – JoshP
    Mar 5, 2013 at 14:03
  • 4
    The question "How do you use gain effectively", nor any of its answers mention normalization, which seems a valid answer below. Also, if I had seen that question before asking this, I would not have understood it's advanced jargon. I did search for answers before asking, and did not see that, because none of the terms I searched for appear there. In other words, this question is different enought to merit a different solution, and in any case, I feel it can be helpful for beginners like myself.
    – Dave M G
    Mar 5, 2013 at 14:27
  • 2
    Although you may not have known it, the question you are asking here is largely about "gain staging" (or gain structuring). That is the "objective measure" you're seeking. That's why I linked the other question. Normalization, while useful, is what you use to try to compensate for improper gain staging.
    – JoshP
    Mar 5, 2013 at 14:41
  • @Josh: I'm not arguing whether or not the other question potentially has useful information. I appreciate you pointing it out to me. I'm just saying that I don't think this question should be closed, because I think it approaches the issue from a direction that is different. Namely that there's no way I, and I think others, can identify that other question as useful because it assumes a lot of advanced knowledge in order to see the relevance.
    – Dave M G
    Mar 5, 2013 at 15:33

3 Answers 3


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 based on what the minimum and maximum signal level are for the system you are working with. All the way off is always negative infinity and it works it's way up to 0 which is the ideal maximum signal level for the device to maintain the best quality. However, nobody talks at exactly the same level all the time, so the signal is going to vary some. On most good equipment, it can peak a little above 0db(signal, not volume), but ideally you want to keep the gain set to be at 0 and a little below it most of the time.

Eventually, if the signal gets too powerful, it will exceed what the device can handle and you end up with the waveform getting cut off at the top and bottom. This results in a partially square wave rather than a rounded one and is called clipping.

When setting the gain for recording, your primary goal is to get the majority of the signal as close to 0 as possible (the best possible quality), but you also have to make sure that the signal does not clip, which will render that part of the recording unusable. You also don't want it too much lower than 0db because then the amount of noise that is recorded will increase in relation to the amount of actual signal.

After recording, when you get in to Audacity, it is simply scaling the signal when you adjust the levels. It makes the signal behave as if it is more or less powerful than it originally was so that you can mix it with other inputs. It is all just math being done on the signal and doesn't actually alter the waveform's basic shape, just makes it have a lower amplitude.

The difference between the highest and lowest volumes on a recording is called the dynamic range. Generally, recordings will have a larger dynamic range than will typically make an ideal listening at home. For this reason, a concept called dynamic compression is frequently used. Dynamic compression is a process by which the highest levels can be automatically gained down and the lower levels can be brought up. This results in less of a difference between the highest and lowest signal levels and decreases the dynamic range. This makes a more level volume when someone plays back the audio later.

Ideally the signal will be compressed and then the level raised to be close to 0 to minimize the noise that will be generated in the playback device and maximize the signal level. Then when the device plays it back, it can accurately process the signal to produce output to the speakers at a level that is consistent with other recordings and will have a similar volume to other things that the user was listening to.

That's a pretty quick overview of a lot of what goes in to a recording, but I hope it helps give a deeper understanding of the answer to your question.

  • 3
    Nice answer for those new to the subject. I think your definition of normalization is off however. Normalization just brings the highest level up to zero, and brings the lowest up by the same amount. The dynamic range is unchanged. What you describe here in your discussion of normalization is actually compression, which drops the level of the louder material, reducing the dynamic range, and then (if you choose to) raises the level of the material back to 0dB.
    – JoshP
    Mar 6, 2013 at 13:22
  • @Josh - thanks, my understanding of normalization was off. I have updated my post accordingly. That's what I get for never having checked the definition after having used a tool early in my recording days that had a feature mislabeled as normalization that actually both compressed and normalized the audio.
    – AJ Henderson
    Mar 6, 2013 at 14:05

One word: Normalization.

Normalization is the process of finding the loudest part of a recording and making it reach the 0 dB level so the loudest part of your recording is as loud as technically corret.

In general, the standard for the highest volume of a recording is defined as 0 dB. Everything which is not as loud will be mentioned as something with a negative value, like -6 dB, which is half the volume of 0 dB.

Aside from the details of leveling, you might want to achieve a recording which is never louder than 0 dB (altough that's technically possible but leads to distortion when you're too fare above), with the loudest part reaching exactly this volume and the lower parts should not be too far away from it, meaning, you might want to maintain a constant volume between -20 dB and 0 dB.

You might already have used gain control on the microphones to make them sound about as loud as each other. If you record each microphone separately, you can use the Normalize filter of Audacity to match the 0 dB for the loudest parts.

You might even want to look into the Audacity Compressor tutorial [1] to see how you can enhance overall volume level of each microphone recording.

If you are set with the result of each microphone, you can mix down the entire project and check if you want additional Normalization of the final project. If each microphone is leveled good, you might not even need this step.

[1] http://manual.audacityteam.org/man/Compressor

  • 1
    It sounds to me, like the OP is asking about the signal chain, not how to normalize the signal.
    – Jason Conrad
    Mar 6, 2013 at 0:27

In the production chain that you can control, your final audio level is called programme level, as far as I know. There are recommendations for television (at least in Europe) that the programme level should not exceed -9 dB, with exceptions. But for web videos there is no such regulation. As a result you often have to readjust your loud speaker's volume while switching from one YouTube video to the other. Some clips limit themselfes to -9 dB, others use the full range up to 0 dB. "The louder the better" results in what is called the loudness war.

For your podcasts you have to define your own standard, so that at least your own visitors don't have to increase/decrease their speaker's volume while switching from one podcast to the other. My personal rule that I have set for my own videos is:

  • I apply a little compression on all voices, just enough to make passages spoken in a low voice as clearly understandable as passages spoken in a louder voice.

  • Then I manually remove all peaks, or make them more silent. (Without this, the next step, the normalization, would amplify wrong.)

  • Before exporting I normalize the whole track to -3 dB. The level of -3 dB was my personal decision.

  • Why not normalize to 0 dB ? If everybody sets the value to 0, the loudness war would disappear, no ? (okay, some people would still exceeed this value...). Why set the level to a value under 0 ? I'm curious.
    – Julien N
    Aug 12, 2013 at 15:00

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