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Old 11th December 2015, 19:26   #1  |  Link
alexVS
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Looking for ac3filter analog

Hi!

I know there is a great tools ac3filter, that allows to do many things, e.g. increase volume in central channel, decrease volume of "surround" sounds, increase "voice" sound etc.

I could convert ac3 to stereo wav using my settings and graphedit.

Is there a program with the same functionality? May be something standalone, cause I don't want to interfere with codec pack already installed. And looks like ac3filter not updated for 3 years already.

Practically I want to take ac3 file, increase voice sound (or central channel) make dynamic compression, listen the result, and mux the whole file in stereo.
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Old 11th December 2015, 20:33   #2  |  Link
AlexKane
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What you need is the ability to decrease the dynamic range of the program material.

In essence, you need to:
  1. Compress the loud parts
  2. Increase the volume
This way you will perceive quieter sounds, without becoming overwhelmed by the louder aspects of the mix (since you applied compression in step 1).

You can either extract the sound and compress it using compressor plugins in a DAW or, use the Dynamic Audio Normalizer
in any of the provided flavors (CLI, VST, etc).

PS. AC3filter does not operate in the way you described. It is effectively a wide-band downwards compressor/limiter.

Last edited by AlexKane; 11th December 2015 at 20:33. Reason: cosmetic surgery
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Old 13th December 2015, 11:05   #3  |  Link
Ghitulescu
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If the functionality doesn't need to be in real time, like ac3filter does, any audio software supporting AC3 would do.
I do not know no software that can do this in RT.
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Old 13th December 2015, 13:54   #4  |  Link
hello_hello
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I have ffdshow installed with all it's decoding disabled except for uncompressed audio. That way something else can do the decoding but ffdshow will process the decoded audio and therefore it's filters can still be used. I'm pretty sure AC3Filter can be configured the same way (PCM only). The only downside is, if you want to enable any AC3 dynamic range compression, the decoder would need to be able to do it.

I compress via a WinAmp plugin. ffdshow will load them. Here's a couple I'd recommend, and obviously you can compress any audio type, and not just AC3 with dynamic range compression data.
LoudMax (dsp_LoudMax.dll is in a zip file under "Additional Releases")
RockSteady

Another alternative, if you want to downmix and convert would be to use foobar2000. It's AC3 decoder (it's a separate plugin) will apply AC3 dynamic range compression, but I don't bother as I apply compression via RockSteady. foobar2000's DSPs can be used with it's converter.
There's Winamp DSP Bridge for loading Winamp plugins, VST adapter for loading VST plugins (The LoudMax DSP I mentioned earlier also comes as a VST plugin), a couple of matrix mixer DSPs for downmixing however you like (I've only used the first one) Matrix Mixer & Channel Mixer and there's another compressor DSP called R128Norm.

Foobar2000 might be a bit of work to set up if you've not used it before, but it has an advantage of being able to save conversion presets and you can just load files into it's playlist and batch convert. As opposed to importing the audio into a program, downmixing, applying compression, exporting it again, then re-encoding etc.

There's a zip file with some compression comparisons attached to this post if you're interested (the uncompressed source is called MeGUI Downmix). I probably could have configured LoudMax better but I'd not used it before. From memory I compressed with each compressor then ran a ReplayGain scan and used the result to adjust each sample to the same volume to make comparing them easier.

I've found when applying compression there's not much point increasing the volume of the centre channel much, because it just gets compressed.

PS The Dynamic Audio Normalizer mentioned a couple of posts ago works much the same way as RockSteady and LoudMax and it's quite good.

Last edited by hello_hello; 13th December 2015 at 17:02.
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Old 13th December 2015, 17:06   #5  |  Link
pandy
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For offline processing (but also to play in real time trough ffplay) ffmpeg can be used, you may apply dynamic compression to only center channel (usually dialogs) and left remain part not affected by dynamic compression, also you can use DynamicAudioNormalizer written by LoRd_MuldeR https://www.ffmpeg.org/ffmpeg-filters.html#dynaudnorm .
Some examples how to manipulate audio channels in ffmpeg are here: https://trac.ffmpeg.org/wiki/AudioChannelManipulation
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