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....
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*
fromfread
, to read enough samples for a single channel; and calllame_encode_buffer
rather thanlame_encode_buffer_interleaved
, with the right-hand channel pointer set to eitherNULL
orpcm_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.
If you are dealing with Mono sound, remember to set the channel to 1
Also update the encoding function