Audio speed changes on converting WAV to MP3

I create a WAV (PCM) to MP3 converter. But the output is too fast.

This is the code, that converts the encoding.

FILE *pcm = fopen(in_path, "rb");
FILE *mp3 = fopen(out_path, "wb");

int read, write;    

const int PCM_SIZE = 8192;
const int MP3_SIZE = 8192;

short int pcm_buffer[PCM_SIZE*2];
unsigned char mp3_buffer[MP3_SIZE];

lame_t lame = lame_init();
lame_set_in_samplerate(lame, sampleRate);
lame_set_brate(lame, byteRate);
lame_set_num_channels(lame, channels);
lame_set_mode(lame, MONO);
lame_set_VBR(lame, vbr_default);
lame_init_params(lame);

do {
    read = fread(pcm_buffer, 2*sizeof(short int), PCM_SIZE, pcm);
    if (read == 0)
    {
        write = lame_encode_flush(lame, mp3_buffer, MP3_SIZE);
    }
    else
    {
        write = lame_encode_buffer_interleaved(lame, pcm_buffer, read, mp3_buffer, MP3_SIZE);
    }
    fwrite(mp3_buffer, write, 1, mp3);
} while (read != 0);

lame_close(lame);
fclose(mp3);
fclose(pcm);

The parameters sampleRate, byteRate and channels are read from the WAV header.

I believe something is missing in the code....

Solution

You are setting it up to encode a mono stream (lame_set_mode(lame, MONO);) but providing the data as if it were interleaved stereo.

If it's a mono stream, then remove the 2* from fread, to read enough samples for a single channel; and call lame_encode_buffer rather than lame_encode_buffer_interleaved, with the right-hand channel pointer set to either NULL or pcm_buffer, to encode just one channel.

If it's a stereo stream, then don't set the mode to mono. You probably shouldn't do this anyway; I think it detects the mode based on the number of channels.

Also, as I mentioned when I wrote that code, you should check for and handle errors if you're using it in a real application. It's a very basic example.