ALSA programming

hello, i try to write a program to record a sound stream from my audiocard(integrated realtek), briefly, without error processing it looks like:

Code:

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
int main(int argc, char **argv) //argv[1] = "hw:0,0"
int i;  
int err;
long buf[128];
int exact_rate = 44100;
int time_s = 3;
static unsigned int channels = 2;        
snd_pcm_t *playback_handle;
snd_pcm_hw_params_t *hw_params;    
char *pcm_name = strdup("plughw:0,0");        
snd_pcm_t *capture_handle;    
    
snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0);
snd_pcm_hw_params_malloc (&hw_params);
snd_pcm_hw_params_any (capture_handle, hw_params);
snd_pcm_hw_params_set_access(capture_handle,hw_params,SND_PCM_ACCESS_RW_INTERLEAVED));
nd_pcm_hw_params_set_format (capture_handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &exact_rate, 0);
snd_pcm_hw_params_set_channels (capture_handle, hw_params, channels);
snd_pcm_hw_params (capture_handle, hw_params);
    
snd_pcm_hw_params_free (hw_params);
snd_pcm_prepare (capture_handle);
    
    
int fd = open("file_rec", O_WRONLY), b;
if(!fd)
 printf("file didn't open\n");
        
        
for (i = 0; i < time_s*44100/128; i++)
    {            
        b = snd_pcm_readi(capture_handle, buf, 128);              
                write(fd,buf,128);
    }        
close(fd);    
snd_pcm_close (capture_handle);
snd_pcm_open(&playback_handle,pcm_name,SND_PCM_STREAM_PLAYBACK, 0);
snd_pcm_hw_params_malloc (&hw_params);
snd_pcm_hw_params_any (playback_handle, hw_params);
snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE);        
snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &exact_rate,0);
snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2);
snd_pcm_hw_params (playback_handle, hw_params);
    
snd_pcm_hw_params_free (hw_params);
snd_pcm_prepare (playback_handle);

fd = open("file_rec", O_RDONLY);
while((b=read(fd, buf, 128)) > 0)        
  snd_pcm_writei (playback_handle, buf, b);            
        
snd_pcm_close (playback_handle);
close(fd);
exit (0);
}

i'm recording a stream in standard audiorecording programm(from mic) and ones in my programm, and get in my programm sound much dirty with a lot of noises. can you help, what can i do??

Well the first thing I would do would be to fix your buffer sizes.

128 bytes (which is all you're saying, not what you actually have) is pitifully small for 16-bit stereo - it's only 32 samples.

Coupled with a high sample rate of 44100, you're looking at around 0.75mS to fill the buffer (and for you to deal with it).

Compare a 10 second recording using standard s/w with a 10 second recording using your s/w. Is there a big difference in file size?
If yours is a much smaller file, then you're missing data!

so, i increasaed buffer up to 65536 bytes(with bigger ones for some reason it doesn't work proper), now i get enough distinct sound, but the output is cycled, something like it every time writes in file the same first buf
what may be wrong?

This could happen if you are using a non-blocking mode audio stream, so that your attempts to write() to the device are just replacing the data in the ring buffer, before it can be fully played back. The ring buffer will then repeatedly play back its last contents, making a looping noise.

Make sure the stream (both the capture and playback) are in blocking mode instead of async mode.