What would cause a recording to have a completely symmetric frequency plot?

Question

I'm developing software that receives PCM audio data and converts it to a WAVE file. The audio stream has the following characteristics:

1. 16 bit signed integers
2. 6000 frames sent every 1/8th seconds, ie 48000 samples per second
3. Mono-channel

I can save to the file just fine, but the audio sounds distorted upon playback.

After doing some experimentation, I've determined that the audio is symmetric across the frequency domain, causing the strange distortions.

My question is: what would cause this?

Could it be reading the data using an incorrect sampling rate? Or some strange normalization? I don't have enough experience with FFT to understand the relationship between time and frequency domains.

If I move the audio file into Audacity and use the "change pitch" tool with +100% (effectively doubling the pitch), it sounds roughly normal. It seems like the entire frequency spectrum has been squished into 12kHz and the mirrored across the other 12kHz range.

Any help is appreciated, and more info can be provided if necessary!

Answer 1

According to this spectral output you are not dealing with 48kHz sampled audio. For some reason the audio has been decimated and the actual sampling rate is 24kHz. Consequently the 'nyquist' frequency is 12kHz.

One of the key things to remember about digital audio, is that during the process of sampling analogue audio the spectral content is mirrored around the nyquist frequency (Fs/2). For 48kHz sampled audio, this is 24kHz, and for 24kHz sampled audio this is 12kHz. So basically the mirroring you are seeing here is typical of this. Were you to be able to analyse this to much higher frequencies, you would see mirroring around multiples of the Nyquist frequency all the way up (theoretically) to infinity.

Now, having said all this, you're still going to be pretty sure that your audio is 48kHz, and consequently, no further forward.

You should further note, that this effect can be simulated by 'nulling' every alternate sample (i.e. taking every odd sample and simply making the value zero). If you were to do this with an existing WAV file, this is exactly the sort of spectrum you would see.

Consequently, if you take all of this into account, and you are still positive your audio is 48kHz sampled, it points to an error in your software which is probably to do with the way you are exporting the signed integers in your PCM file. If I was you, I wouldn't try to reinvent the wheel here. There are any number of usable libraries out there that will write rock solid WAV files for you, without having to start from scratch.

Be careful of the WAV/RIFF header, and also be aware of 'endianness' when dealing with this data.