Re: [ecasound] Trying to normalize two channels independently

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Subject: Re: [ecasound] Trying to normalize two channels independently
From: Andy Loening (loening@ucla.edu)
Date: Mon Nov 06 2000 - 02:45:45 EET


Allright, getting close. Thanks Kai! One last little problem.... When I
combine the two channels back using the following command:

ecasound -sr:48000 \
 -a:1 -f:s16_le,1,48000 -i channel_1.wav \
 -a:2 -f:s16_le,1,48000 -i channel_2.wav -erc:1,2 -eac:0,1 \
 -a:all -f:s16_le,2,48000 -o renormalized.wav

The resultant output file only uses ~50% of the amplitude in the left and
right channels. When I use either of the following commands:

ecasound -sr:48000 \
 -a:1 -f:s16_le,1,48000 -i channel_1.wav \
 -a:all -f:s16_le,2,48000 -o renormalized.wav

ecasound -sr:48000 \
 -a:2 -f:s16_le,1,48000 -i channel_2.wav -erc:1,2 -eac:0,1 \
 -a:all -f:s16_le,2,48000 -o renormalized.wav

The resultant output file uses all 100% of the amplitude in the respective
channel (with the other channel of course being empty).

I assume this behavior is a feature, so that when you're combining
multi-track recordings into a single file you don't get clipping. Is
there anyway to disable this?

I guess I could also just amplify the two channels by 200%, but I'd want
to convert the wav files to 24bit first, so that I don't drop any bits
when going from 100%->50%->100%

Any thoughts,
        Andy

On Mon, 6 Nov 2000, Kai Vehmanen wrote:

> On Sun, 5 Nov 2000, Andy Loening wrote:
>
> > #to combine back into one channel
> > ecasound -sr:48000 \
> > -a:1 -f:s16_le,1,48000 -i channel_1.wav \
> > -a:2 -f:s16_le,1,48000 -i channel_2.wav \
> > -a:1,2 -f:s16_le,2,48000 -o try_again.wav
> [...]
> > It all seems okay up until the last command where I'm trying to convert
> > the two mono channels back into a single stereo file. When I use the
> > command as written above, I get both incoming channels on the left channel
> > of the output file.
>
> You'll need to add a "-erc:1,2 -eac:0,1" to the second chain. This will
> copy the data from ch1 to ch2, and then mute (0% amplify) ch1. So when
> writing to 'try_again.wav' we have two chains, "1" with one channel
> of audio, and "2" with ch1 muted and ch2 containing the audio from
> 'channel_2.wav'.
>
> > I also have a question about ecanormalize. Is ecanormalize going to
> > internally resample to 44100 when it's doing it's thing?
>
> Nope, it uses the sampling rate of the input file when processing.
>
>

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