#Convert 4 bit pcm to wav trial

For example, when increasing the volume, be aware that some samples my end up out of range. This is like adding a multidigit number without the carry. I've seen many questions on stack overflow where people make the mistake of dealing with 16-bit audio data byte-by-byte, even though each sample of 16-bit audio is composed of 2 bytes.

You can't simply deal with the data as a stream of bytes. You must actually use the native format of the file or the proper conversion.Int sampleCount //total number of samples = sample frames * channels For example, once you have an array of shorts representing the data, you could divide each short by two to cut the volume in half: You may be able to do all the processing you need to do in the native format of the file. Usually when we are interested in the maximum, we are looking for the maximum deviation from zero, and we don't really care if it's positive or negative - either way is going to sound equally loud. Note how we find the absolute value of each sample. Printf( 'Max left %d, Max right %d.', maxl, maxr ) Maxr = (short) MAX( maxr, abs( samples ) ) Maxl = (short) MAX( maxl, abs( samples ) ) total number of samples = sample frames * channels

Iterating through all your data and finding the max on each channel would look something like this: In the latter case, you would just need to be aware that when you index an even number it's the left channel, and when you index an odd number it's the right channel. Or you might simply have an array of shorts: To store 16-bit stereoВ interleavedВ audio in C, you might use a structure like this:

In Java, you can use readShort() from DataInputStream. If you are only dealing with C, you can do your IO directly with short arrays, or simply use casting or type punning from a raw char array. Therefore, to read raw 16 bit data, you will usually want to get the data into an array of shorts. In Java, and most C environments, a 16 bit signed integer is represented with the short datatype. It's interleaved because each left sample is followed by a corresponding right sample in the same frame.

It's 16-bit because 2 8-bit bytes make up a single sample. You can tell it's big-endian because the most significant byte (MSB) comes first. The above example shows 2 sample frames of big-endian, 16-bit interleaved audio. In case it gives any further clues I've tried various combinations of settings, increasing sample size while decreasing sample rate increases the (white-)noise, and even 8-bit is still noisy, which is why I'm thinking 4-bit.2 sample frames of big-endian, 16-bit interleaved audio. I was also going to try ffmpeg, but I can't find the appropriate format/codec to set. I've tried sox, but I'm most likely doing something wrong as it complains of an unsupported size: sox -r 44100 -e signed -b 4 -c 2 in.raw out.wav

#Convert 4 bit pcm to wav trial

raw file, so I know for sure what I'm dealing with (as I'm just going by trial and error so far). What I'm looking for is either a tool that can handle 4-bit raw formats, or even a tool that can determine (or guess at) the format of a given. However, my suspicion is that the files may in fact be encoded in 4-bits with a sample rate of 44100hz, but I'm having trouble finding a tool that can handle this. With a bit of trial and error in Audacity I've found that I can listen to a still noisy version of the audio using raw file input settings of 8-bit signed PCM in stereo with a sample rate of 22050hz. raw files from an old game (Zork Nemesis) and determined that they're audio files, however I'm having trouble converting them into something meaningful.

Convert 4-bit pcm to wav