dev request

dev request

[Brad Holland | Muzik Liberated]

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Hey Guys, 
Since you are sox developers we thought that you may well be the best people to talk to.
We are looking for a small script (perhaps using sox) that scans a raw audio file and applies some modulation to the audio file which amplifies the content at specific points (when the signal is lower than -6db) and  continues to do this every time the sound does this throughout its decay.

This is to combat 8-bit sample artifacts when creating custom roms for a  drum machine. (Alesis HR-16). There is a special ASIC chip in this machine that detects a 6db spike in a waveform and the chip then turns down the gain -6db in essence, by keeping the amplitude of the waveform above the noise floor inherent in an 8-bit audio data you are able to achieve much higher fidelity in short decaying 'drum shot' type sounds.

We are a community of enthusiasts for the machine, although none of us could code anything like this. We'd be willing to pay/donate to the project or an individual to knock us up a bit of code to convert this..

A much more in-depth explanation of the process we need to accomplish is written in this post:http://www.burnkit2600.com/diy-sound-roms/  

If someone is interested in helping us with this please get back to me! 

Cheers
Brad

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dev request

[Brad Holland | Muzik Liberated]

[-- Attachment #1.1: Type: text/plain, Size: 1269 bytes --]

Hey Guys, 
Since you are sox developers we thought that you may well be the best people to talk to.
We are looking for a small script (perhaps using sox) that scans a raw audio file and applies some modulation to the audio file which amplifies the content at specific points (when the signal is lower than -6db) and  continues to do this every time the sound does this throughout its decay.

This is to combat 8-bit sample artifacts when creating custom roms for a  drum machine. (Alesis HR-16). There is a special ASIC chip in this machine that detects a 6db spike in a waveform and the chip then turns down the gain -6db in essence, by keeping the amplitude of the waveform above the noise floor inherent in an 8-bit audio data you are able to achieve much higher fidelity in short decaying 'drum shot' type sounds.

We are a community of enthusiasts for the machine, although none of us could code anything like this. We'd be willing to pay/donate to the project or an individual to knock us up a bit of code to convert this..

A much more in-depth explanation of the process we need to accomplish is written in this post:http://www.burnkit2600.com/diy-sound-roms/  

If someone is interested in helping us with this please get back to me! 

Cheers
Brad

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Re: dev request

[Eric Wong]

Brad Holland | Muzik Liberated <brad@muzikliberated.com> wrote:
> We are looking for a small script (perhaps using sox) that scans a raw
> audio file and applies some modulation to the audio file which
> amplifies the content at specific points (when the signal is lower
> than -6db) and  continues to do this every time the sound does this
> throughout its decay.

I think using the "compand" effect should get you there.  Tuning it for
your audio is the trickier part.  Here's some great posts on using it
(not written by me):

http://sourceforge.net/mailarchive/message.php?msg_id=23427259
http://mid.gmane.org/20090829011522.cf27e6ee.fmiser@gmail.com
http://mid.gmane.org/20091231022430.aaa510b3.fmiser@gmail.com

(disclaimer: I'm no expert in audio nor sox, just a Unix plumber)

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Re: dev request

[Cedric Roux]

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On 09/08/2013 11:08 PM, Brad Holland | Muzik Liberated wrote:
> Hey Guys,
> Since you are sox developers we thought that you may well be the best people to talk to.
> We are looking for a small script (perhaps using sox) that scans a raw audio file and applies some modulation to the audio file which amplifies the content at specific points (when the signal is lower than -6db) and  continues to do this every time the sound does this throughout its decay.
>
> This is to combat 8-bit sample artifacts when creating custom roms for a  drum machine. (Alesis HR-16). There is a special ASIC chip in this machine that detects a 6db spike in a waveform and the chip then turns down the gain -6db in essence, by keeping the amplitude of the waveform above the noise floor inherent in an 8-bit audio data you are able to achieve much higher fidelity in short decaying 'drum shot' type sounds.
>
> We are a community of enthusiasts for the machine, although none of us could code anything like this. We'd be willing to pay/donate to the project or an individual to knock us up a bit of code to convert this..
>
> A much more in-depth explanation of the process we need to accomplish is written in this post:http://www.burnkit2600.com/diy-sound-roms/
>
> If someone is interested in helping us with this please get back to me!
>
> Cheers
> Brad

Hi Brad,

attached hack might do the trick.
If not, tell me, I'll have a closer look,
there might be bugs in there (just a quick
and dirty hack).

You can compile it, right? If you're using
windows you may need to change stuff in there
to handle binary files correctly.

Anyway, just tell me if something is wrong, whatever it is.

Regards,
Cédric.

[-- Attachment #2: christmas.c --]
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#include <stdio.h>
#include <sys/stat.h>
#include <stdlib.h>
#include <math.h>

int main(int n, char **v)
{
  char *infile;
  char *outfile;
  FILE *f;
  struct stat s;
  float *data;
  float *max;
  ssize_t size;
  ssize_t i;
  float gain;
  int up;
  float last;

  if (n != 3) {
    printf("gimme <input float mono file> <output float mono file>\n");
    return 1;
  }

  infile = v[1];
  outfile = v[2];

  if (stat(infile, &s)) { perror(infile); return 1; }
  size = s.st_size;

  data = malloc(size * sizeof(float)); if (data == NULL) abort();
  max = malloc(size * sizeof(float)); if (max == NULL) abort();

  f = fopen(infile, "rb"); if (f == NULL) { perror(infile); return 1; }
  fread(data, size, 4, f);
  fclose(f);

  max[size-1] = fabsf(data[size-1]);

  /* populate max */
  for (i = size-2; i >= 0; i--) {
    float m = fabsf(data[i]);
    if (m > max[i+1]) max[i] = m; else max[i] = max[i+1];
  }

  /* apply gain */
  gain = 1;
  up = 1;
  last = data[0];
  for (i = 0; i < size; i++) {
    data[i] *= gain;
    max[i] *= gain;
    if (up == 0 && data[i] > last) {
      /* local minimum */
      if (max[i] <= 0.5) gain *= 2;
    }
    if (data[i] > last) up = 1;
    else if (data[i] < last) up = 0;
    last = data[i];
  }

  f = fopen(outfile, "wb"); if (f == NULL) { perror(outfile); return 1; }
  fwrite(data, size, 4, f);
  fclose(f);

  return 0;
}

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