Audio conversion (LAME) C++ WAV -> MP3

Question

Good afternoon. Sorry, I do not speak very good English.

For some days, I have tried to convert WAV to MP3. For that I use LAME. I can convert the file through the command prompt with the command:

lame.exe-B128 - resample 8-mj \ "C: \ \ test.wav \" \ "C: \ \ test.mp3 \"

but I'd like to not go through a command line (for portability across OSes), but I can’t do it in C++ (via the LAME library).

Here is my code:

int main()
{
//...
//...
// extraction of PCM audio data file .wav in file.pcm
short read, write;

FILE *pcm = fopen("file.pcm", "rb");
FILE *mp3 = fopen("file.mp3", "wb");

const int PCM_SIZE = 18424;
const int MP3_SIZE = 13424;

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

lame_t lame = lame_init();

if ( lame == NULL ) {
    cout<<"Unable to initialize MP3"<<endl;
    return -1;
}

lame_set_num_channels(lame, 1);
lame_set_in_samplerate(lame, 8000);
lame_set_out_samplerate(lame, 8000);
lame_set_brate(lame, 128);
lame_set_mode(lame, MONO);
lame_set_quality(lame, 2);
lame_set_bWriteVbrTag(lame, 0);
if (( lame_init_params(lame)) < 0) {
    cout << "Unable to initialize MP3 parameters"<<endl;
    return -1;
}

do {
    read = fread(pcm_buffer,sizeof(short), 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);

return 0;
}

I get an MP3 file of very poor quality (and a shorter duration than the original).

Any ideas to solve the problem?

Here's my info.

Wav:
- Frequency= 8000
- BlocSize= 16
- Canal Nomber= 1 (mono)
- BytePerSec= 16000
- BytePerBloc= 2
- BitsPerSample= 16
- Bitrate= 128
- .wav size: 13466 (ie size of PCM: 13422)

Thank you.

A sample rate of 8000 Hz will never sound very good. I wish I could find documentation for the brate parameter, that might be key too. — Mark Ransom
@MarkRansom according to this code, it looks like 8kHz is also the input file's sample rate, therefore I'd tend to dismiss this parameter as the cause of the quality loss. — SirDarius

SBM SBM · Accepted Answer · 2013-12-19T10:57:59

 @try
 {
int read, write;
FILE *pcm = fopen([cafFilePath cStringUsingEncoding:1], "rb");  //source
fseek(pcm, 4*1024, SEEK_CUR);                                   //skip file header

mp3 = fopen([mp3FilePath cStringUsingEncoding:1], "wb");  //output
const int PCM_SIZE = 8192*3;
const int MP3_SIZE = 8192*3;
short int pcm_buffer[PCM_SIZE*2];
unsigned char mp3_buffer[MP3_SIZE];

lame_t lame = lame_init();
lame_set_in_samplerate(lame, 11025*2);
lame_set_VBR(lame, vbr_default);
lame_init_params(lame);

int nTotalRead=0;

do {
    read = fread(pcm_buffer, 2*sizeof(short int), PCM_SIZE, pcm);

    nTotalRead+=read*4;

    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);
    // write = lame_encode_buffer(lame, pcm_buffer,pcm_buffer, read, mp3_buffer, MP3_SIZE);

    fwrite(mp3_buffer, write, 1, mp3);
} while (read != 0);

lame_close(lame);
fclose(mp3);
fclose(pcm);
}
@catch (NSException *exception)
{
    NSLog(@"%@",[exception description]);
}

Audio conversion (LAME) C++ WAV -> MP3

1 Answers