Ben,

I did figure it all, but only after I had the RIFF specs well in front of
me... :)

Thank you

Duilio



At 11.41 18/10/99 -0700, you wrote:
>Duilio,
>
>Sure it can. If it's a simple PCM .WAV file, with no compression, then all
>you have to do is figure out where the header ends. The rest is raw audio
>data, and all you need to do is figure out whether it's hi or lo endian (in
>the case of 16-bit data), and what the midpoint is.
>
>Cheers,
>Ben
>
>----- Original Message -----
>From: Duilio Foschi <foschid@SHUTTLE.ETRURIA.NET>
>To: <PICLIST@MITVMA.MIT.EDU>
>Sent: Monday, October 18, 1999 11:17 AM
>Subject: Re: [OT] WAV files
>
>
>> Daniel,
>>
>> I tried that.
>>
>> I can assure you that the RIFF structure of a WAV file cannot be guessed
>by
>> a binary dump.
>>
>> You'd have a better chance with the state lottery :)
>>
>> Thank you
>>
>> Duilio
>>
>> At 20.23 18/10/99 +1300, you wrote:
>> >Duilio, why don't you open a 16 bit *.wav for random binary input and
>> examine the
>> >data ? try a silent sample and then a low frequency sine wave, the
>results
>> should
>> >be fairly apparent.
>> >
>> >regards,
>> >Graham Daniel.
>> >
>> >Duilio Foschi wrote:
>> >
>> >> Brent,
>> >>
>> >> my files are sampled at 16 bits.
>> >>
>> >> How do this change things ?
>> >>
>> >> Also, when does the header exactly end and data start ? (I need to take
>a
>> >> regular WAV file for input and output a regular WAV file to be played
>later).
>> >>
>> >> Thank you
>> >>
>> >> Duilio Foschi
>> >>
>> >> At 22.44 16/10/99 +1300, you wrote:
>> >> >> I need to know how the sound information is stored in a WAV file and
>how I
>> >> >> can programmatically raise the volume of the sound in a WAV file.
>> >> >>
>> >> >> Any hint welcome
>> >> >>
>> >> >> Thank you
>> >> >>
>> >> >> Duilio Foschi
>> >> >>
>> >> >
>> >> >Not so [OT] because you can do this on a PIC.  Here's some ideas.
>> >> >
>> >> >Assuming a simple mono 8 bit per sample recording, there is a
>> >> >about 20 bytes or so of header information at the start of the file
>> >> >and then the audio samples are stored as 8 bit values.  The data is
>> >> >"biased" with a 7Fh DC offset so that silence is recorded as 7Fh.
>> >> >Kind of like a single rail audio amplifier would be biased at half the
>> >> >supply voltage.  For simple play back from a micro with a PWM
>> >> >output just send each byte to the PWM register sequentially at the
>> >> >original sample rate.  Put an RC filter on the output and
>> >> >capacitively couple it to an audio amp.  A little rough by hey - it
>> >> >works!  Even play back the header bytes, they go by so fast you
>> >> >hardly hear a scratch.
>> >> >
>> >> >Back to the subject, to digitally amplify a .wav file you would need
>> >> >to do something like this for each 8 bit sample:
>> >> >- Remove the offset by subtracting 7Fh for data above 7Fh or
>> >> >subtracting from 7Fh for data below 7Fh
>> >> >- Multiply by your amplification factor
>> >> >- Put the offset back in by adding 7Fh or subtracting from 7Fh as
>> >> >necessary
>> >> >- Store it or play it
>> >> >
>> >> >You are limited to what you can do in an 8 bit file before clipping.
>> >> >Using 16 bit intermediate maths helps by letting you do stuff like
>> >> >this: for 10% increase in volume, multiply by 11 then divide by 10.
>> >> >
>> >> >Brent Brown
>> >> >Electronic Design Solutions
>> >> >16 English Street
>> >> >Hamilton, New Zealand
>> >> >Ph/fax: +64 7 849 0069
>> >> >Mobile: 025 334 069
>> >> >eMail:  brent.brown@clear.net.nz
>> >> >
>> >
>