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24th September 2010, 11:39 | #1 | Link |
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Avi Volume far too high!
I love AGK and use it quite often. But I do have an ongoing problem.
After conversion, the volume of the avi file is far too high! Is there anyway I can reduce the volume (gain) of the converted file? Many thanks |
24th September 2010, 14:38 | #2 | Link |
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If AGK has a "Normalize" option - or particularly a "Boost" option - either disable it or set the dB level to be lower.
If you're using the original DVD audio (not re-encoded to MP3), see if you can disable DRC in your decoder. |
24th September 2010, 17:10 | #3 | Link |
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It is nothing to do with AutoGK. The normalize feature is built in and works very well, though very occasionally it can clip audio 0.5db over maximum, but not to make the audio sound like you are saying. The problem IS your source, it must be too saturated or the loud level is excessive/bass levels/dynamic range are really screwed up.
If it is too loud to start with AutoGK will only normalize it to peak levels anyway. Result: it is still loud again with 2 channels e.g if default/auto mp3 was used, or if the original audio was used it will be still too loud again! Demux your avi/mp3 track with VirtualDubMod, use Mp3Gain and analyse it, then reduce db level to a more suitable level eg 89db/default or less. Then remux quieter track back into original. Voila solved! Google these free tools! Last edited by azmoth; 24th September 2010 at 17:25. |
27th September 2010, 10:41 | #4 | Link | |
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28th September 2010, 15:57 | #5 | Link |
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Neither Shrink nor any other decrypter touches the Audio. It's exactly as it is in the DVD. What kind of audio are you talking about, the original AC3 audio or reencoded MP3? What's in the AVIs that sounds so loud? If you're not sure, check an AVI in GSpot or MediaInfo, or post the AutoGK log for us to have a look.
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29th September 2010, 05:35 | #6 | Link |
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What are you using to play back the AVIs? What sort of audio are you using? The original AC3 etc or are you converting to MP3? If you're using the original audio then the volume won't be touched.
AutoGK has never created an AVI with audio volume that's too high for me. I assume you're hearing distortion rather than suffering from an inability to simply turn it down? |
29th September 2010, 10:11 | #7 | Link | |
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4th October 2010, 02:28 | #8 | Link |
interlace this!
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FYI, broadcast is usually peak limited to -10dBFS, and averaged either to -18 or -20. if you use a normalize function on a source that's mixed for TV, then you'll immediately be 10dB louder than broadcast level (and no doubt what your TV's volume is set for).
normalizing audio by scaling to peak should be banned for this reason - peak has no relation to actual loudness. when you're backing up DVDs (that typically are somewhat equal loudness, within a couple of dBs), but tend to use theatrical mixes or TV mixes interchangeably, all you can end up with is uneven loudness. your big budget movies will be too quiet, and your TV series will be claustrophobic and unrelenting in their assault on your ears. if you need to boost audio for some reason, either use replaygain scans as a guide, or boost everything by a fixed amount and use peak limiting to prevent clipping. but your TV or stereo will no doubt have a volume control that can be adjusted as needed.
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5th October 2010, 20:34 | #9 | Link | |
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@temac
Hi! You've been asked the following question twice and not answered directly (that I can see): Quote:
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7th October 2010, 15:48 | #10 | Link | |
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If you play a DVD with the same player, I assume the volume's not too high? I've used AutoGK for years and it's volume normalising has never caused an issue, and I've played the AVIs it's created using quite a few different DVD players. Sure, some audio sounds louder than others (depending on the dynamic range), but even without normalising it's fairly rare to switch between video files and not have to adjust the volume eventually anyway. |
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9th October 2010, 15:54 | #12 | Link | |
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Assuming "average level" is a far better way of normalising, which average level do you pick? How do you know that some audio won't have enough dynamic range for it's peaks to cause clipping when normalising to the "average level" you've chosen to use? Do you normalise to your average level, then reduce it if necessary to stop peaks from clipping..... which would give you the same volume as peak normalising would have done in the first place, which of course most of the time usually results in clipping.... I'm just wondering how your "average level" method works. |
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16th October 2010, 18:03 | #13 | Link |
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Average means exactly that: average. Using peak levels often results in overmodulation in AutoGK in either left or right channel. How it would work better is normalising to say an average of 95percent of maximum levels and avoid clipping most times. Why it clips is down to the tool used. Ask the creator!
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17th October 2010, 15:59 | #14 | Link | |
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I assume you were trying to say you'd normalise to some average level which just happens to be a level which puts the peaks at around 95%, which brings me back to the question you didn't answer.... although I do appreciate you providing the definition of average instead. What's the average level you'd pick for normalising audio which would guarantee the peaks would only hit about 95%? Does all movie audio have the same dynamic range so you can pick an average level which will work for all of them? I just demuxed the MP3 audio from an AVI created by AutoGK (converted from 5.1 AC3). I know AutoGK uses tools to decode the audio to a stereo wave file, it then normalises the wave file and finally that's converted to MP3. I opened the demuxed MP3 with MP3DirectCut and told it to normalise it. It found the peak level to me -1db so wouldn't alter it. I then converted the MP3 to a wave file and opened it with a wave editor. Once again the peaks were at about -1db. So before I ask the "creator" why it clips, I think I'll need to see some evidence that it actually does. Unless of course I just happened to pick one of those AVIs which didn't contain clipped audio as it would "most of the time". By the way, how do MP3s get "overmodulated"? I can't think of a context relating to MP3s in which that makes sense. |
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18th October 2010, 17:55 | #15 | Link |
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Yes, that is the problem, not all have the same dynamic range yet DVD etc is supposed to have standard 5.1 track. 95% is a happy medium since actual loudness of a track varies and is a different issue. I have countlessly done encodings with AutoGK and found mp3 bitrates of 112, 160 clip while 128 or 192 do not most of the time. Don't ask me why/how overmodulation/distortion or clipping occurs technically to "T"?
However it is distortion and any tool like mp3Gain/MagixDeluxe will prove that this occurs after analysing some mp3tracks done with AutoGK so you are very wrong thinking otherwise. Why don't you lend us your extensive experience and explain why it does not occur! |
19th October 2010, 04:12 | #16 | Link | |||
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Firstly, MP3s can only be changed in volume in 1.5db steps when doing so without re-encoding. It's a limitation I don't understand fully but it likely means the values MP3Gain displays are only accurate to 1.5db anyway (I'm not saying the program is bad as I use it religiously on my music MP3s). Secondly, MP3Gain's volume analysis and reported clipping have absolutely zero to do with whether or not the actual audio contained in the MP3 has been clipped. It's reporting whether the MP3 is likely to have peaks which will clip the preamp of the playback device after it's been ReplayGained, not whether the MP3 itself contains peaks which have been clipped. Plus MP3s can store values above 0db so they don't actually cause clipping when encoding. It's all easy enough to confirm. Load one of your "clipping" MP3s into MP3Gain and note how it says "clip track". Now change the target volume from 89db (the default) to 79db. The clip track warning will likely disappear. Why? Mp3Gain hasn't changed the volume of the MP3 yet, but when it does all it'll be doing is reducing the volume to a point where the peaks aren't likely to clip the preamp. It hasn't magically fixed any clipped peaks in the MP3 audio itself..... it can't. Once the peaks are clipped they're clipped. All Mp3Gain can do is turn those clipped peaks up and down. To prove it I demuxed the MP3 audio from 3 AVIs I'd recently created as they were still on the hard drive. I just picked the first 3 episodes of a series. I ran three through MP3Gain and it indicated "clip track" for all three. I checked the tags MP3Gain added to one of the AVI MP3s after running the analysis. MP3Gain was showing a clip track warning but the tag it created revealed the actual peak to be -0.988031. No clipping. Next I loaded each one into MPDirectCut and told it to normalise them. It couldn't do so as they were too close to 0db (1.5db steps for MP3) but it reported the peaks at being -0.9db, -0.6db and -0.7db. Not one of the MP3s contained audio which exceeded 0db despite all three producing clip track warnings in MP3Gain. Quote:
When converting to MP3 filtering is first applied (high pass and low pass filters)... the frequencies which are filtered are probably dependant on the bitrate being used. This filtering creates a whole new waveform to be sampled. Then dither comes into play. Dither is a kind of intentionally introduced noise which is used to randomise quantisation errors. Basically every time you alter (filter) a waveform and then resample it, because each sample can only be a certain number of values there must be "rounding". Dither randomises the way the rounding is done (for want of a better description) as random type rounding sounds natural to the human ear. So..... It is possible the filtering and consequent dither will result in a sample which exceeds 0db but the MP3 is able to store that value faithfully so it's still not clipping the actual waveform, it's only when decoding the MP3 that the stored value greater than 0db may be clipped. If it's decoded to wave file it will, if it's simply decoded and sent to the amplifier it may not. I say all this is nothing to worry about because it's a tiny amount of possible clipping of the waveform you're never likely to be able to hear, especially in relation to a movie soundtrack where over a two hour period the normalised wave file AutoGK created for converting to MP3 probably only hits 0db a couple of times for a fraction of a second. Probably when someone fires a gun and you're not likely to hear 0.3db of clipping there. CD players which oversample use dithering and they can cause peaks to exceed 0db slightly for the same reason, but nobody seems to complain about it. I'd open some of those MP3s you've run through MP3Gain and assumed were clipping, reverse any volume changes MP3Gain may have applied and then open the tags to see if any of the peak levels really exceed 0db. When you do, I think you'll probably re-evaluate your thoughts on AutoGK's normalising causing clipping. |
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19th October 2010, 12:29 | #17 | Link |
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Originally Posted by yetanotherid
To prove it I demuxed the MP3 audio from 3 AVIs I'd recently created as they were still on the hard drive. I just picked the first 3 episodes of a series. I ran three through MP3Gain and it indicated "clip track" for all three. I checked the tags MP3Gain added to one of the AVI MP3s after running the analysis. MP3Gain was showing a clip track warning but the tag it created revealed the actual peak to be -0.988031. No clipping. Actually it proves nothing other than that the initial analysis may be wrong. That is why you must check with other programs too and if same e.g clipping shows, then it is correct. I always find mp3Gain is fairly accurate. No, I have not re-evaluated my initial thoughts about the normalizing procedure as many tools I have used corrobate clipping effects in some tracks. I do not rely on just one tool for analysis. I agree like you write the actual distortion is nothing to worry about, but to some ears it will sound "scratchy." |
19th October 2010, 12:36 | #18 | Link | |
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It all depends on how well does work the analog chain (from the DAC to the speakers) in "impuls mode" (typical example cymbals and guitars). The red dots are the digital sample, the curve is the analog output of the DAC. |
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19th October 2010, 16:10 | #20 | Link | |||
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Err.... MP3Gain was one of the programs I used. Quote:
Have you ever opened an MP3 tag and looked at the peak value MP3Gain writes to the MP3 or are you still assuming "clip track" means the waveform was clipped somewhere during the process of normalising and converted to MP3? I don't know what "many tools" might involve or whether you understand them properly either. Well I'd be keen to understand the point. If you run more than one program and they don't agree on how loud the peaks are then you need to work out which one is rubbish and stop using it. Aside from ReplayGain, what other types of analysis do you run? Quote:
Yet when audio in a video file could have a few tiny instances of clipping over a duration of hours, suddenly everyone's got bionic ears. |
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