How to detect a basic audio signal into a much bigger one (mpg123 output signal)

Question

I am new to signal processing and I don't really understand the basics (and more). Sorry in advance for any mistake into my understanding so far.

I am writing C code to detect a basic signal (18Hz simple sinusoid 2 sec duration, generating it using Audacity is pretty simple) into a much bigger mp3 file. I read the mp3 file and copy it until I match the sound signal.

The signal to match is { 1st channel: 18Hz sin. signal , 2nd channel: nothing/doesn't matter). To match the sound, I am calculating the frequency of the mp3 until I find a good percentage of 18Hz freq. during ~ 2 sec. As this frequency is not very common, I don't have to match it very precisely.

I used mpg123 to convert my file, I fill the buffers with what it returns. I initialised it to convert the mp3 to Mono RAW audio: init:

int ret;
const long *rates;
size_t rate_count, i;
mpg123_rates(&rates, &rate_count);
mpg123_handle *m = mpg123_new(NULL, &ret);
mpg123_format_none(m);
for(i=0; i<rate_count; ++i)
    mpg123_format(m, rates[i], MPG123_MONO, MPG123_ENC_SIGNED_32);

if(m == NULL)
{
    //err
} else {
    mpg123_open_feed(m);
}

(...)

unsigned char out[8*MAX_MP3_BUF_SIZE];
ret = mpg123_decode(m, buf->data, buf->size, out, 8*MAX_MP3_BUF_SIZE, &size);

`(...)

unsigned char out[8*MAX_MP3_BUF_SIZE];
ret = mpg123_decode(m, buf->data, buf->size, out, 8*MAX_MP3_BUF_SIZE, &size);

(...) `

But I have to idea how to get the resulting buffer to calculate the FFT to get the frequency.

//FREQ Calculation with libfftw3
int transform_size = MAX_MP3_BUF_SIZE * 2;
fftw_complex *fftout = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * transform_size);
fftw_complex *fftin = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * transform_size);
fftw_plan p = fftw_plan_dft_r2c_1d(transform_size, fftin, fftout, FFTW_ESTIMATE);

I can get a good RAW Audio (PCM ?) into a buffer (if I write it, it can be read and converted into wave with sox: sox --magic -r 44100 -e signed -b 32 -c 1 rps.raw rps.wav

Any help is appreciated. My knowledge of signal processing is poor, I am not even sure of what to do with the FFT to get the frequency of the signal. Code is just fyi, it is contained into a much bigger project (for which a simple grep is not an option)

Brad Brad · Accepted Answer · 2020-04-01T11:43:06

Don't use MP3 for this. There's a good chance your 18 Hz will disappear or at least become distorted. 18 Hz is will below audible. MP3 and other lossy algorithms use a variety of techniques to remove sounds that we're not going to hear.

Assuming PCM, since you only need one frequency band, consider using the Goertzel algorithm. This is more efficient than FFT/DFT for your use case.

How to detect a basic audio signal into a much bigger one (mpg123 output signal)

1 Answers