audio to 8-bit text sample conversion

I have an interesting question today.

I need to convert some pokemon audio files to a list of 8-bit samples (0-255 values). I am writing an assembly routine on the MC6800 chipset that will require these sounds to be played. I plan on including an array with the 8-bit samples that the program will loop through when a function is called.

Does anyone know a way to convert audio files (wav/mp3) into a list of comma separated 8-bit text sample va开发者_运维百科lues? Or anything of this relative method?

Thank you so much in advance!

You can use the command-line "sox" tool or the Audacity audio editor to convert the file to a raw, unsigned 8-bit mono audio file.

In Audacity 1.3 or higher, open the audio then select Export, choose "Wave, AIFF, and other uncompressed types" as the format, then click Options... - then choose "Other..." for the Format, "RAW" for the Header, and Signed 8-bit PCM as the encoding. (Sorry, unsigned isn't available.)

From the command line, try sox with -c 1 for 1 channel, -t raw for no header, -u for unsigned linear, and -1 for 1 byte per sample.

Then you can use a tool like "hexdump" to dump out the bytes of the file as numbers and paste them into your code.

If sox doesn't have it, you will have to use it to generate raw (headerless) files and convert the raw files to comma-separated yourself.

EDIT: sox has "Raw textual data" as one of its formats, from the web page. You can make it convert your sound files to unsigned 8-bit linear samples in a first pass and then probably get exactly the output you want using this option for output.

For .wav it is a very simple process. You can find the .wav specification easily with a google search. It comprises a header then simply raw samples. You should read the header first, then loop through all the samples. Usually they are 16 bit samples, so you want to normalize them from the range -32768 to 32767 to your 0-255 range. I suggest simple scaling at first. If that's not successful maybe find the actual min and max amongst the samples and adjust your scale accordingly.

Well a lot depends on your audio format. The wave format, for example, consists of uncompressed interleaved PCM data.

ie for an 8-bit stereo file each sample will be arranged as follows.

[Left Sample 1][Right Sample 1][Left Sample 2][Right Sample2]...[Left Sample n][Right sample n].

ie each 8 bit stereo sample is stored in 2 bytes. 1 for the left channel and 1 for the right. This is the data format your sound hardware will most likely require.

A 16 or 24-bit audio file will work in each way but the left and right samples will be 2 or 3 bytes each, respectively.

Obviously a wave file has a load of extyra information in it. It follows the RIFF format. You can find info on it and the "chunks" wave files use at places such as www.wotsit.org.

To decompress an MP3 is more complicated. You are best off getting hold of a decompressor and running it on the MP3 encoded audio. IT will spit out PCM data as above from the other side.