I have searched the forum to see if what I'm about to ask now has already been discussed, but even though I found many threads on this topic, none of them discussed the matter at the level of my ignorance of these technical issues, so I'm going to
ask very simply:
I downloaded a great sounding vinyl rip (in flac format) of an album I love and it sounds much better than other versions of the album I have. But the volume level of this rip is lower than the necessary level for me to be able to listen to it via my digital audio player (zune 120) in public transportation. (I use lossless formats on the audio player, btw.) Anyway, I know that I could normalize the files to -1 db and it makes the album louder. However the dilemma is that I prefer this vinyl rip over the many versions of this album for it sounds better, but if because of normalization the sound is worsened, the cause will be lost.
So my questions are:
1. Does normalization of an audio file in wav format to -1 db result in decrease of sound quality?
2. Should I normalize tracks seperately or all of them at once?
3. There are options for limiting etc. if I choose to select all to normalize at once... I shouldn't go into that, right? Or is there limiting involved in normalization of individual tracks anyway?
Thanks in advance.
1. Does normalization of an audio file in wav format to -1 db result in decrease of sound quality?
Short answer - No! Normalization does not degrade quality. Digital volume adjustment is considered lossless... Mixing engineers, mastering engineers and other people involved in audio production do it all the time, and they don't give it a 2nd thought. (I normalize my vinyl transfers.)
I assume that you are setting the
peaks to -1dB? That's OK.
But if you want to "get picky", there are tiny rounding errors with almost any digital alteration, including normalization.
2. Should I normalize tracks seperately or all of them at once?
Some songs are supposed to be louder than others, so you should normalize the album as a whole. (All of the songs should be adjusted by the same amount.)
3. There are options for limiting etc. if I choose to select all to normalize at once... I shouldn't go into that, right? Or is there limiting involved in normalization of individual tracks anyway?
What software are you using? Normalizing should be a linear volume adjustment.
Limiting (and compression) is not-linear and CAN degrade sound quality. The only time limiting would be involved is if you boost the volume to the point where the peaks would be above 0dB.
...But if you want to "get picky", there are tiny rounding errors with almost any digital alteration, including normalization.
So. Normalization decreases sound quality.
Yes, but to a degree that is quite provably negligible for all listening scenarios.
If you normalize and output to the sound card with 24 bits of precision, the normalization error will be 20-40db under the noise floor of the DAC. If you output to 16-bit, but perform internal processing at higher precision and apply dither/noise shaping, the only sound quality loss is an addition of background noise, not nonlinear distortion. When dithering is not performed one would be extremely hard-pressed to find a quantization error spectrum which could rise significantly above that DAC noise floor... &c.
If you're concerned about such things enough to avoid normalization, then you probably have no business listening to recorded music whatsoever, as virtually all digital and analog recordings suffer from distortions far worse in magnitude.
For some strange reason people seem to think it is all or nothing. Normalize. Save the normalized result to a new name. Listen to it. If it doesn’t appeal to you, go back to the original.
It is very unlikely you could notice any result of the quantization errors in normalizing a 16 bit file, but ... .
One must have the software with the required functionally, but converting the file (new name again!) to a floating point format should be available in quite a few application, I think. Normalizing in that format will certainly make unhearable quantization errors. Do a proper dithered conversion back to your target bit depth. You will never hear the tiny amount of added noise.
If the audio is sufficiently quiet, it's most likely that the small amount of quantization noise introduced from the normalization process will be outweighed by the reduced amplification required in the analog domain (which equates to less noise) to actually be able to listen to the music.
I thank everyone for the replies...
2. Should I normalize tracks seperately or all of them at once?
Some songs are supposed to be louder than others, so you should normalize the album as a whole. (All of the songs should be adjusted by the same amount.)
3. There are options for limiting etc. if I choose to select all to normalize at once... I shouldn't go into that, right? Or is there limiting involved in normalization of individual tracks anyway?
What software are you using? Normalizing should be a linear volume adjustment. Limiting (and compression) is not-linear and CAN degrade sound quality. The only time limiting would be involved is if you boost the volume to the point where the peaks would be above 0dB.
The problem is I downloaded the vinyl rip and the files are already seperated. I'm using Adobe Audition 3.0 and when I select an individual track, under the "favorites" tab there's "Nomralize to -.1 db" option, but that's not applicable when I select all the files; if I select all and right-click and select "group waveform normalization" a more complex normalization menu appears and the limiting etc. options are there.
I gave normalizing tracks individually a shot and the result was fine for most of the tracks. But two quiet tracks resulted sounding a bit louder than they should, I guess.
So can you recommend me a software that will take separate wav files and normalize them all at once without losing the quietness of the quiet songs compared to the loud ones... ?
Regards.
You could use foobar2000's Converter to create new files with 'hard' (i.e. applied directly to the audio data, not just stored in tags) album-mode ReplayGain, although you'll probably need additional amplication to approach -1 dB, and I'm not certain if this option exists (there doesn't appear to be a page about the Converter's Processing options on the wiki).
1. Does normalization of an audio file in wav format to -1 db result in decrease of sound quality?
Generally, no. In rare borderline cases, yes.
There is a tiny theoretical loss quality that is inherent in just about every kind of processing. However, it takes a lot more than a tiny theoretical loss to cause an audible loss.
2. Should I normalize tracks seperately or all of them at once?
I presume that you are normalizing to obtain a collection of recordings that have consistent loudness. While normalizing may actually accomplish that in some cases, it is far from the best way to do it. Normalizing relates to amplitude, but due to the complex properties of human hearing, amplitude does not always mean the same thing as loudness. So, normalizing does not always improve the consistancy of perceived loudness.
3. There are options for limiting etc. if I choose to select all to normalize at once... I shouldn't go into that, right? Or is there limiting involved in normalization of individual tracks anyway?
AFAIK the best way to cause a collection of tracks to have the same perceived loudness is to be skilled at doing such things, and doing it manually. The second best way is to use a program that uses a psychoacoustic model to evaluate loudness, and follow its recommendations. IME, the second best way is actually very good, and it is the best choice for most people who don't want to make a career out of audio production.
AFAIK the best way to cause a collection of tracks to have the same perceived loudness is to be skilled at doing such things, and doing it manually. The second best way is to use a program that uses a psychoacoustic model to evaluate loudness, and follow its recommendations. IME, the second best way is actually very good, and it is the best choice for most people who don't want to make a career out of audio production.
Thanks.... But what software -which will do the job and is simple also- should I use?
AFAIK the best way to cause a collection of tracks to have the same perceived loudness is to be skilled at doing such things, and doing it manually. The second best way is to use a program that uses a psychoacoustic model to evaluate loudness, and follow its recommendations. IME, the second best way is actually very good, and it is the best choice for most people who don't want to make a career out of audio production.
Thanks.... But what software -which will do the job and is simple also- should I use?
Last time I had this situation, I used FOOBAR2000.
I double checked its results by ear and was satisfied that the FOOBAR2000 recommended level adjustments were about the same as I would develop using lengthy listening sessions and a lot of experimentation. BTW, I applied the adjustments with Audition.
As dv1989 mentioned, foobar2000 in Album Replay Gain (http://wiki.hydrogenaudio.org/index.php?title=Replaygain) mode is what you should try. It's easy and you have many options: after you have RG scanned your tracks, you can use preamp in Playback preferences if you want it louder (with processing: apply gain and prevent clipping) and then even convert the files when you are satisfied with the results (Converter also has preamp option in processing tab). Thou you don't have to alter your tracks if you play the tracks with RG aware player
...too late
But the volume level of this rip is lower than the necessary level for me to be able to listen to it via my digital audio player (zune 120) in public transportation.
I assume this is because the volume control won't go up high enough to overcome noise. What kind of headphones are you using? I see people wearing standard iPod earbuds on the subway and figure if they're able to make out the music, they must be doing damage to their ears. Best to use in-ear style headphones that seal and isolate.
Have you measured the peak levels of the tracks? Normalization won't do anything if the tracks already hit full scale.
You just need to amplify all the tracks by the same amount, that amount being the amount needed to bring the highest amplitude peak to -1dB or -0.1dB or 0dB or whatever you prefer.
You can do it manually in Audition. I only have Cool Edit Pro (much older version of Audition), but it's easy: load all the tracks into a single track (i.e. append them all), check the gain change required to peak normalise that long full-album track to the level you want (there's an option for that under "amplify", and it's also listed in the track statistics, but these can take a while to gather), close the full-album track, and then apply that same gain change to each and every track separately. Job done.
If you don't want to load all the tracks into a single track, load them all individually, check the required gain change for each track, and then use the smallest value you find.
As others have said, foobar2k will do this for you automatically, but peak normalisation-by-album requires the correct settings - which, from memory, mean ReplayGaining the tracks, and then applying the album gain with clipping prevention with a "with ReplayGain" pre-amp at maximum. It'll only work if the required gain change is less than the combination of the ReplayGain change and the pre-amp setting. It's simpler than it sounds - though it's a bit of a waste of time because it calculates the ReplayGain information and then doesn't use it (apart from the peak value).
Cheers,
David.
Normalization:
- Usually useless (Cd's, for example, tend to be normalized already)
- Generally inconsistent (If it even gets to do something, as stated above, amplitude is not the same as loudness, and two files normalized to the same value can have a much different loudness).
- So 1990's.
- For the most part the effect of the change is bigger than the side effects. (so quality should not be affected perceptibly).
Compression/limiting:
- May be mostly useless (CD's are for the most part compressed, FM Radio/TV compresses on emitting)
- Difficult to get good results (There isn't one setting for all, and most general settings either are soft, or too constant).
- They are good tools, if used properly. Here at hydrongeaudio, when this topic shows up, it is mostly for the bad use of it.
Replaygain/loudness adjustment:
- The proper method to do what most people think that normalization does.
- Like normalization, it's just a change of amplitude. Unlike normalization, it applies different amplitudes depending on how it sounds, not how it looks like.
- If you have a player that supports replaygain (there are some portable players that start to support it, but there aren't many), then applying just a tag is enough.
- If the player does not support replaygain, the change has to be applied to the file, (like normalization does).
Like suggested above, the proper way to replaygain a flac album for use on a portable that does not support replaygain is:
-Add all flacs album files in foobar2000.
-select them, context menu(right mouse click), replaygain, scan all files as an album.
-select them all again, context menu, convert.
-select the output format (you can create it wav, mp3, flac...).
-select processing tab, enable replaygain, and select album gain mode. Inside the replaygain configuration, you can increase the final gain, if 89dB is not enough (probably your case with the portable. 92 or 94 is a good suggestion. if you neeed higher than 96, there is probably a problem with your player).
- and let it convert.
There is a tiny theoretical loss quality that is inherent in just about every kind of processing. However, it takes a lot more than a tiny theoretical loss to cause an audible loss.
This is the right answer. I recently applied 120 gain changes to a 32-bit Wave file containing a 400 Hz sine wave, and after all of 120 processes the distortion products were all below the -150 dB limit of my FFT program. If you do the same 120 volume changes with a 16-bit Wave file, the distortion is still below 0.1 percent. So for one gain change the distortion added is surely insignificant.
--Ethan
If you want the experience intended by the performance, you want the same amplification on each track, preserving the differences between tracks. In Audition you should find Open Append on the File menu. If the files are all alone in one folder, select all to get them all into one file, or select the individual tracks while holding down the Ctrl key (select in reverse sequence: 10, 9, 8, ... to get the original sequence: 1, 2, 3 ...). All will end up in one file with que ranges delimitating the individual tracks. Normalize to the value you want under Transform/Amplify/Normalize.
If you want the result back to individual files, open the Que List, select all, Batch. You will have to rename the output files if you want the original names. That is quick and easy with copy-paste.
If you want the experience intended by the performance, you want the same amplification on each track, preserving the differences between tracks.
One could hope that the origional mastering engineer actually thought about setting artistically relevant levels for the various tracks. However, in these days of hypercompression, that is not always a safe assumption.
Reading the OP again, it appears to me that there was problem with a digital player/headphone combination that could not play every track loud enough in noisy listening environments, which is rally a different set of priorities.
I agree that the best solution would be to correct the problem where it started, and obtain a digital player/headphone amp/headphone setup that had enough reserve gain to be satisfactory as it is being used. Fiio has some under $20 headphone amps that may be highly useful even in noisy mobile environments.
There is a tiny theoretical loss quality that is inherent in just about every kind of processing. However, it takes a lot more than a tiny theoretical loss to cause an audible loss.
This is the right answer. I recently applied 120 gain changes to a 32-bit Wave file containing a 400 Hz sine wave, and after all of 120 processes the distortion products were all below the -150 dB limit of my FFT program. If you do the same 120 volume changes with a 16-bit Wave file, the distortion is still below 0.1 percent. So for one gain change the distortion added is surely insignificant.--Ethan
In this regard, there is an interesting experiment, see postings #98 to #102 here:
http://www.gearslutz.com/board/remote-poss...malizing-4.html (http://www.gearslutz.com/board/remote-possibilities-acoustic-music-location-recording/345015-do-you-normalize-there-any-benefit-normalizing-4.html)
Floating point is a different animal than the integer math Ethan used. You should consider Ethan's results worst case and be happy that many modern DAWs use floating point math.
I thank all for the answers.
On foobar2000, when applying the replay gain to the tracks during conversion,
Source: Album
Processing: apply gain and prevent cllipping
with rg info: left at 0
without rg info: THIS IS WHERE I DON'T UNDERSTAND.
I read that it should be slided to -8db but I don't understand what this is... could you please help me with this...
Thanks in advance.
With that slider, if your files are not tagged with replayGain data, you can tell fb2k to make a volume adjustment.
With modern popular music recordings, the replayGain adjustment usually lowers the volume to match the reference level of 89dB. So if most of your music is averaging out to having replayGain values of -5, then it would be reasonable to assume that, on average, any music you might wish to play that does not yet have replayGain tags should be adjusted -5dB--it's basically a guess.
If you are busy listening to your RPG'd files at your desired volume and you try new CD or something that is rather loud and doesn't have RPG tags, it can cause you to reach for your volume knob rather quickly!
With that slider, if your files are not tagged with replayGain data, you can tell fb2k to make a volume adjustment.
With modern popular music recordings, the replayGain adjustment usually lowers the volume to match the reference level of 89dB. So if most of your music is averaging out to having replayGain values of -5, then it would be reasonable to assume that, on average, any music you might wish to play that does not yet have replayGain tags should be adjusted -5dB--it's basically a guess.
If you are busy listening to your RPG'd files at your desired volume and you try new CD or something that is rather loud and doesn't have RPG tags, it can cause you to reach for your volume knob rather quickly!
90 % of the stuff that I listen to were recorded during the 70s. That's the very reason I'm looking for a way to boost the volume of some of my albums, for in the 70s most albums weren't recorded at high levels compared to the modern recordings.
With what settings would I get the loudest result without sacrificing quality (ie. clipping)?
What happens if I set the "with replay gain" setting higher than 89db?
What happens if I set the "without replay gain" setting at -1db or even slide it to +db levels?
Here's one example:
(http://img248.imageshack.us/img248/9190/rgcapture.jpg) (http://img248.imageshack.us/i/rgcapture.jpg/)
EDIT:
I converted the above album with settings at: 89db, with gain 0, without gain -3db.
The resulting files turned out even queiter than the orginal files!
So with what settings should I convert this album?
Thanks in advance for your replies.
Your older recordings are probably less (dynamically*) compressed...
Modern recordings are compressed to make them sound louder. Compression is used to increase the overall
average volume without boosting the peaks or or clipping the peaks. (Sometimes it's over-done, and some modern CDs are clipped.)
Does the make sense? Your modern recordings
sound louder even though they are normalized to the same peak level as your older recordings.
90 % of the stuff that I listen to were recorded during the 70s. That's the very reason I'm looking for a way to boost the volume of some of my albums, for in the 70s most albums weren't recorded at high levels compared to the modern recordings.
If your 70's recordings are normalized (maximized),
you can't increase the volume (linearly**) any more without clipping. So... if you want to match the volume and you can't
increase the volume on the older dynamic recordings,
your only choice is to reduce the volume on the louder tracks. This is why Replay Gain reduces the gain on most tracks.
What happens if I set the "with replay gain" setting higher than 89db?
You are not leaving RG as much "room to work". If you've set-up RG to prevent clipping, it will end-up having no effect on many tracks. If you allow it to clip, the tracks will be louder and many will be clipped.
What happens if I set the "without replay gain" setting at -1db or even slide it to +db levels?
The tracks without RG will play louder. Since RG is reducing the volume on most tracks, these non-RG tracks will generally be louder than the RG tracks. If you go above 0dB, many of these tracks will clip.
Normalization works on the
peaks. The peak volume does not correlate well with perceived volume.
Replay Gain tries to adjust the perceived volume. (BTW - The 89dB value is an
acoustic loudness reference level, whereas the 0dB reference is the maximum peak
digital value in a computer file or fed into a DAC, etc.)
*
Dynamic compression is not related to
file compression like FLAC or MP3.
** Compression is non-linear. You can use an audio editor to compress your older recordings and make them louder... But, reducing dynamic contrast will make them sound boring, just like modern recordings!
With what settings would I get the loudest result without sacrificing quality (ie. clipping)?
What happens if I set the "with replay gain" setting higher than 89db?
What happens if I set the "without replay gain" setting at -1db or even slide it to +db levels?
You actually want to set the reference level lower than 89 to accommodate a library with a wider range of loudness. Anything that gets a positive Replay gain reading may potentially clip when played back with volume leveling enabled. The Replay gain specification suggests using 83. I personally use 85. 89 seems to work well for material only back to the 1990s.
MediaMonkey has a "clipping prevention" feature associated with its Replay gain implementation. It's not real clear how this works but I believe it lowers playback gain below the Replay gain suggestion if the Replay gain value would results in clipping.
With what settings would I get the loudest result without sacrificing quality (ie. clipping)?
What happens if I set the "with replay gain" setting higher than 89db?
What happens if I set the "without replay gain" setting at -1db or even slide it to +db levels?
You actually want to set the reference level lower than 89 to accommodate a library with a wider range of loudness. Anything that gets a positive Replay gain reading may potentially clip when played back with volume leveling enabled. The Replay gain specification suggests using 83. I personally use 85. 89 seems to work well for material only back to the 1990s.
MediaMonkey has a "clipping prevention" feature associated with its Replay gain implementation. It's not real clear how this works but I believe it lowers playback gain below the Replay gain suggestion if the Replay gain value would results in clipping.
In fact I don't listen to stuff back to back, one after the other. I listen to indivudal albums one at a time. In fact I don't much listen to musiuc through my computer, and I don't really care for having to adjust the volume continuously in between tracks.
What I want to be able to do is increase the overall volume level of indivudal albums for putting into my mp3 player. Without clipping and sacrificing quality, I want to be able to increase an albums overall sound. I mean I want to embed these replaygain values in the tracks forever. That's it.
So if you consider my questions against this background, what would you recommend me?
There's one other thing I don't understand. In Foobar2000, after scanning and updating the gain levels, I'm asked to set a value for "without gain" before the conversion process. I don't get it. These tracks have already been scanned and updated and became tracks "with gain" anyway, isn't it so? So the setting for "without gain" should be ineffective even if I set it to +1000db, since these tracks belong to the "with gain" category, ain't that true?
Different editors may use terms like "normalize" differently. In Audacity, the normalize function removes any DC bias, and re-balances the channels. This could be useful in ripping vinyl as sometimes one channel is consistently a little louder than the other.
Amplify would be the function to scale both channels by the same amount.
Another thing to check with vinyl rips is whether the loudest part of the file is a click. Remove that before scaling.
In fact I don't listen to stuff back to back, one after the other. I listen to indivudal albums one at a time. In fact I don't much listen to musiuc through my computer, and I don't really care for having to adjust the volume continuously in between tracks.
What I want to be able to do is increase the overall volume level of indivudal albums for putting into my mp3 player. Without clipping and sacrificing quality, I want to be able to increase an albums overall sound. I mean I want to embed these replaygain values in the tracks forever. That's it.
So if you consider my questions against this background, what would you recommend me?
In that case, you don't want Replay Gain at all! As I said before, RG tries to make all tracks (or albums) play with the
same loudness... It's NOT trying to
maximize loudness! And, since most tracks can't be increased without clipping, it works by
reducing most tracks.
Just normalize (maximize) all of your albums to 0dB (some people recommend slightly below 0dB). In other words,
increase all of the tracks by the same amount. This can be a little tricky if your album is separated into tracks. The most straightforward way to do this is to re-combine all of the songs into a single WAV (or FLAC) file, normalize, and re-split the tracks. (If you normalize the album as a whole, some tracks won't be normalized to 0dB... Maybe only one track will be normalized to 0dB, but this is how it should be!)
Again, if these older recordings are more dynamic than your modern recordings,
they can be normalized/maximized and they will never be as loud as the modern recordings. In fact, your FLAC files may already be normalized and re-normalizing may do nothing. (The only way to make them "sound louder" is to apply dynamic compression, which I
don't recommend... There's a reason you like that 70's sound, and the lack of over-compression is probably part of it!)
If the album is mastered properly, you shouldn't have to adjust the volume between songs. Some songs are supposed to sound louder than others, some songs (even quiet-sounding tracks) will have higher peaks than others, etc. If you normalize the tracks separately, you mess-up the relationship.
Replay Gain has two modes/values. If you choose "album gain", it will apply the same gain adjustment to all tracks. This will retain the relative gain between tracks as originally mastered (assuming you haven't messed that up by normalizing the tracks individually).
During playback in the "album" mode, Replay Gain will adjust the level once every time you start a different album. The goal is to make all
albums equally loud without adjusting the volume in the middle of an album and messing-up the way the album was produced. When playing back in "track" mode, the level will be adjusted for each song.
There's one other thing I don't understand. In Foobar2000, after scanning and updating the gain levels, I'm asked to set a value for "without gain" before the conversion process. I don't get it.
I don't use foobar, but I if I understand correctly, I believe this is for playing tracks without replay gain. If all of your tracks have replay gain, this setting won't do anything. If you want to maximize your non-replay gained tracks and you want to avoid clipping, set this to 0dB (no change).
So if you consider my questions against this background, what would you recommend me?
I went back to the beginning and see that you want to listen on public transport. The likely reason you cannot hear well is because the train/plane/bus is too loud. The best fix is to quiet the ambient noise it by using in-ear or noise cancelling headphones.
You can turn it up louder but you risk hearing damage with combined high ambient and headphone levels. Some devices just don't go up loud enough to overcome high level ambient noise and/or damage your ears.
If for some reason, you want to try to get around this, you'll want to apply some of the same dynamic range compression processing that the professionals use to make today's CDs sound so much louder than yesterday's. Technically speaking, this processing does not cause clipping but it definitely does decrease sound quality.
Are there any decent multi-band DRC VST plugins available?
While it can decrease sound quality, when used carefully, DRC will not necessarily cause your stuff to sound bad. I have a feeling some of these comments are the typical overly-polarized, knee-jerk reactions to the loudness war that we commonly see on this forum.
I believe many audio editors have dynamic range processing facilities. I’ve used those of CoolEdit to remake a few CDs for in-automobile listening. Not knowing much about what I should do, I just played with some samples of wide range orchestral music until I had something that seemed halfway reasonable. It isn’t perfect, someone who knew where they were going could probably do much better, but it made a quite significant listening improvement under those noisy conditions.
While it can decrease sound quality, when used carefully, DRC will not necessarily cause your stuff to sound bad. I have a feeling some of these comments are the typical overly-polarized, knee-jerk reactions to the loudness war that we commonly see on this forum.
DRC is a type of distortion of the original recording. Information in the original recording is lost. I think it is OK to say that it "decreases sound quality". I agree that it doesn't necessarily "sound bad".
There are listening situations (e.g. transportation, exercise) where you need to reduce dynamic range to be able to enjoy the music. There needs to be a DRC dial on playback equipment. Why has this not happened?
There needs to be a DRC dial on playback equipment. Why has this not happened?
I agree with the need. The most probable reason why a DRC dial has not been included on other than a minority of all playback gear is probably price/perforamnce.
DRC adjustments have been fairly common features of surround receivers, usually labelled something like "Midnight Listening" or some such.
I don't think it is about price/performance. It's a simple DSP algorithm. Might be a usability issue. How do you explain to someone what DRC does. Can you get reasonable results with one or two preset modes?
The reason there is DRC on AVRs is because movie soundtracks actually have dynamics. The DRC rarely works in stereo mode. Popular music recording have all the dynamics squeezed out of a during mastering. DRC during playback is not going to do anything. Sadly I think I just answered my original question.
I don't think it is about price/performance. It's a simple DSP algorithm. Might be a usability issue. How do you explain to someone what DRC does. Can you get reasonable results with one or two preset modes?
I don't know how complex of a box you have to build before the DSP resources to build a good DRC can have zero incremental cost. I think it would still be more than a basic MP3 player.
The reason there is DRC on AVRs is because movie soundtracks actually have dynamics. The DRC rarely works in stereo mode. Popular music recording have all the dynamics squeezed out of a during mastering. DRC during playback is not going to do anything. Sadly I think I just answered my original question.
Perhaps we should put ourselves in the position of people who actually have to sell product. I think we end up with a chicken and egg problem. The manufacturer of the digital player questions whether adding a DRC facility will sell more players or allow him to get more for each player that is sold. The producers of recordings question whether they can trust the DRC hardware that is out there to optimize the sales of his recordings should he produce them uncompressed.
I think it is OK to say that it "decreases sound quality".
...and it is every bit as OK to say that it can be used to increase sound quality. Personally, I would avoid making assertions about sound quality, especially considering the topic being discussed.
I think it is OK to say that it "decreases sound quality".
...and it is every bit as OK to say that it can be used to increase sound quality. Personally, I would avoid making assertions about sound quality, especially considering the topic being discussed.
Point taken. I was trying to adhere to the OP's language. Perhaps the original question needs to be rephrased.
On Audacity if you use "Amplify" it'll raise the levels of the track to the maximum level WITHOUT clipping. Should be a quick and easy way to normalize, I suppose.
Did the OP miss this...
Normalization:
- Usually useless (Cd's, for example, tend to be normalized already)
...because it's true. The example the OP posted had a peak at 0.92. Raising this to 1.00 isn't going to make much audible difference (if any).
DRC the old tracks, or get a louder mp3 player, or more sensitive earphones, or reduce the background noise.
DRC is the cheapest option, but if you're picky, it probably won't satisfy. If it was possible to take a 1970s style master, run it through a one-click free batch process, and make it as loud as a 2010 master, then people wouldn't pay tens or hundreds of thousands of dollars for equipment to make 2010-style masters. ;)
You might be able to get close enough. There are loads of DRC plug-ins out there. On the more complex ones, you could spend half a life-time getting the settings right. e.g. try http://www.stereotool.com/ (http://www.stereotool.com/)
Cheers,
David.
There is a plugin called vlevel for foobar which works quite nicely as DRC. The settings I use are: strength 1.0, multiplier 2, buffer 5 seconds. (not sure if vlevel works with the latest version of foobar, I use 9.6.4) This DSP can be used for file conversion as well.
Is vlevel a multi-band compressor? Depending on how much compression you need, anything less might not be able to get the job done without creating a sonic mess.
I tend to agree with greynol, and can suggest Stereo Tool (http://www.stereotool.com/) as a free multiband compressor which I have at least toyed around with, and seems to work.
vlevel (http://vlevel.sourceforge.net/about/)
Basically, vlevel is just a volume normalizer, it doesn't color the sound in any way other then volume changes.