Very muffled audio from concert recording; can it be fixed?

I took a video at a concert about 4 years ago using a point-and-shoot camera with fairly limited video/audio capabilities – the sound in the video is very muffled and I’d like to improve it and clear it up if at all possible. I’ve been trying to fix it with Audacity 1.3 but I haven’t had much luck so far…however, I don’t have much experience with this sort of thing so I don’t know if the lack of improvement is because this audio is beyond repair and already as good as it’s going to get or because I just don’t know what I’m doing and haven’t applied the right adjustment, effect or filter yet.

Here’s the audio file, extracted from the video using “Direct stream copy” in VirtualDub:

If anyone could tell me what I could do (if anything) to clear this track up and improve the quality/clarity or have a go at it themselves it would be much appreciated. Thanks!

There’s not a great deal that you can do to improve that. In fact I don’t think that there’s anything you can do to make a substantial improvement.
Point-and-shoot cameras just don’t have the capability of handling the extreme high sound levels from live rock concerts.
The way that professional record live rock concerts is to take a direct feed from the mixing desk.
You may be able to make some subjective improvement by using the Equalization effect, but it’s not going to make any real improvement.

When television news people have to shoot stuff like this and they can’t get a desk feed, they use special microphones and attenuators to keep everything from overloading and distorting.

Many rock bands don’t care if people try to record their concerts because a vast majority of the recordings get what you got.


The sample rate on that is only 8000Hz, so sounds very muddy, (no frequencies above 4KHz).
If that’s the rate on the original video recording not a lot can be done except boost what high frequencies are there ….

If the sound when you play the video sounds much better, with more high frequency content, you should double check that the poor 8000Hz rate is not being imposed by the conversion software which is extracting the audio.

Audacity 1.3 will rip the audio from video if the FFmpeg libraries are installed.

Thanks for the responses!

I’ve been playing around with the Equalization effect and have been able to make some slight improvements but it’s been mostly guesswork on my part (dragging the line around randomly to find what sounds best) since I really don’t know much about this sort of thing. Are there any plug-ins or additional EQ curves that I could import that may be useful for this?

I extracted the audio file with VirtualDub using “Direct Stream Copy” mode so I figured it would be the same as in the video, but I’m still a beginner to all this so I could be wrong.

I used MediaInfo to check the audio info from the original raw video:

ID : 2
Format : PCM
Format settings, Endianness : Little
Format settings, Sign : Unsigned
Codec ID : raw
Duration : 11mn 47s
Bit rate mode : Constant
Bit rate : 64.0 Kbps
Channel(s) : 1 channel
Sampling rate : 8 000 Hz
Bit depth : 8 bits
Stream size : 5.40 MiB (2%)

And here’s the info from the extracted WAV audio file:

ID : 0
Format : PCM
Format settings, Endianness : Little
Codec ID : 1
Codec ID/Hint : Microsoft
Duration : 11mn 47s
Bit rate : 64.0 Kbps
Channel(s) : 1 channel
Sampling rate : 8 000 Hz
Bit depth : 8 bits
Stream size : 5.40 MiB (100%)

So it looks like the sample rate isn’t a product of the extraction but rather just the way the audio was originally recorded.
In VirtualDub there’s the option of “Full Processing Mode” instead of “Direct Stream Copy” which provides these Audio Conversion settings when exporting the audio track:

I tried exporting it using a sampling rate of 44100Hz and checked the boxes for 16-bit, Stereo and High quality but couldn’t tell any difference when listening to that file versus the original…the only noticeable difference is that the original audio file is 5.4MB and the audio file with the conversion settings applied is 119MB, so I assume that applying those settings offers no benefit over the original 8000Hz audio file since that’s how the audio was originally recorded.

About your attachment: what exactly did you do to achieve that result? Any specific plug-ins or EQ curves used or did you just manually boost the treble using Equalization?

If you look at the frequency analysis* ( comparing before and after treble boost) you can see what frequencies have been boosted …

[ * see “Analyze”, “plot spectrum” ]

As you’ve found out resampling at a higher rate doesn’t improve sound quality, it just makes the file bigger.

{ Update: listening back I think I’ve used a bit of chorus on the treble boosted version, gives a stereo effect .
You could try Steve’s pseudostereo effect plugin instead of chorus }.