How do I generate a 1kHz pure tone using asoundlib ALSA API in C++?

  alsa, c++, libasound, linux, raspberry-pi

I am following the sample code here to generate a pure tone sinewave using ALSA API in C++ in Linux: http://equalarea.com/paul/alsa-audio.html

The code I am using is under "A Minimal Interrupt-Driven Program". The code as is does not compile and my modified version of the code that actually compiles makes some sound is given below:

#include <stdio.h>
#include <stdlib.h>
#include <errno.h>
#include <poll.h>
#include <math.h>
#include <alsa/asoundlib.h>
      
snd_pcm_t *playback_handle;
// short buf[4096];
short buf[48*2];        // try 1000 Hz @48000, 48 samples @48000 = 1 ms

void sine_gen(){
    // call once to generate the sine values
    // if this works do this inside callback function later on
    double tempSin = 0.0;
    #if 1
    // check if buf[] contains 2 channels interleaved
    for(int i = 0; i<48*2; i+=2){
        tempSin = sin(2.0*M_PI*1000*i/48000);
        buf[i] = (short)(tempSin * pow(2.0, 15) );
        buf[i+1] = buf[i];
    }
    #else
    // check if buf[] contains single channel data
    for(int i = 0; i<48*2; i++){
        tempSin = sin(2*M_PI*1000*i/48000);
        buf[i] = (short)(tempSin * pow(2.0, 15) );
    }   
    #endif
    
}

void SetAlsaMasterVolume(long volume)
{
    long min, max;
    snd_mixer_t *handle;
    snd_mixer_selem_id_t *sid;
    const char *card = "default";
    const char *selem_name = "Master";

    snd_mixer_open(&handle, 0);
    snd_mixer_attach(handle, card);
    snd_mixer_selem_register(handle, NULL, NULL);
    snd_mixer_load(handle);

    snd_mixer_selem_id_alloca(&sid);
    snd_mixer_selem_id_set_index(sid, 0);
    snd_mixer_selem_id_set_name(sid, selem_name);
    snd_mixer_elem_t* elem = snd_mixer_find_selem(handle, sid);

    snd_mixer_selem_get_playback_volume_range(elem, &min, &max);
    snd_mixer_selem_set_playback_volume_all(elem, volume * max / 100);

    snd_mixer_close(handle);
}

int
playback_callback (snd_pcm_sframes_t nframes)
{
    int err;

    // printf ("playback callback called with %u framesn", nframes);

    /* ... fill buf with data ... */
    // if this part is empty, buf[] should have been filled correctly inside sine_gen()
    
    if ((err = snd_pcm_writei (playback_handle, buf, nframes)) < 0) {
        fprintf (stderr, "write failed (%s)n", snd_strerror (err));
    }
    
    return err;
}
      
int main (int argc, char *argv[])
{

    snd_pcm_hw_params_t *hw_params;
    snd_pcm_sw_params_t *sw_params;
    snd_pcm_sframes_t frames_to_deliver;
    int nfds;
    int err;
    struct pollfd *pfds;
    
    sine_gen();     // call this once to generate sinewave

    if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
        fprintf (stderr, "cannot open audio device %s (%s)n", 
             argv[1],
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
        fprintf (stderr, "cannot allocate hardware parameter structure (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
        fprintf (stderr, "cannot initialize hardware parameter structure (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
        fprintf (stderr, "cannot set access type (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
        fprintf (stderr, "cannot set sample format (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    unsigned int f_s = 48000;
    // if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0)) < 0) {        // causes segmentation fault; see // https://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___h_w___params.html#ga6014e0e1ec7934f8c745290e83e59199
    if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, &f_s, 0)) < 0) {
        fprintf (stderr, "cannot set sample rate (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
        fprintf (stderr, "cannot set channel count (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
        fprintf (stderr, "cannot set parameters (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    snd_pcm_hw_params_free (hw_params);
    
    /* tell ALSA to wake us up whenever 4096 or more frames
       of playback data can be delivered. Also, tell
       ALSA that we'll start the device ourselves.
    */

    if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0) {
        fprintf (stderr, "cannot allocate software parameters structure (%s)n",
             snd_strerror (err));
        exit (1);
    }
    if ((err = snd_pcm_sw_params_current (playback_handle, sw_params)) < 0) {
        fprintf (stderr, "cannot initialize software parameters structure (%s)n",
             snd_strerror (err));
        exit (1);
    }
    // if ((err = snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, 4096)) < 0) {        // change this as per the size of the buffer used
    if ((err = snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, 48*2)) < 0) {       
        fprintf (stderr, "cannot set minimum available count (%s)n",
             snd_strerror (err));
        exit (1);
    }
    if ((err = snd_pcm_sw_params_set_start_threshold (playback_handle, sw_params, 0U)) < 0) {
        fprintf (stderr, "cannot set start mode (%s)n",
             snd_strerror (err));
        exit (1);
    }
    if ((err = snd_pcm_sw_params (playback_handle, sw_params)) < 0) {
        fprintf (stderr, "cannot set software parameters (%s)n",
             snd_strerror (err));
        exit (1);
    }

    /* the interface will interrupt the kernel every 4096 frames, and ALSA
       will wake up this program very soon after that.
    */

    if ((err = snd_pcm_prepare (playback_handle)) < 0) {
        fprintf (stderr, "cannot prepare audio interface for use (%s)n",
             snd_strerror (err));
        exit (1);
    }
    
    // SetAlsaMasterVolume(0);      // attempt changing volume      // didn't make any difference

    while (1) {

        /* wait till the interface is ready for data, or 1 second
           has elapsed.
        */

        if ((err = snd_pcm_wait (playback_handle, 1000)) < 0) {
                fprintf (stderr, "poll failed (%s)n", strerror (errno));
                break;
        }              

        /* find out how much space is available for playback data */

        if ((frames_to_deliver = snd_pcm_avail_update (playback_handle)) < 0) {
            if (frames_to_deliver == -EPIPE) {
                fprintf (stderr, "an xrun occuredn");
                break;
            } else {
                fprintf (stderr, "unknown ALSA avail update return value (%d)n", 
                     frames_to_deliver);
                break;
            }
        }

        // frames_to_deliver = frames_to_deliver > 4096 ? 4096 : frames_to_deliver;
        frames_to_deliver = frames_to_deliver > (48*2) ? (48*2) : frames_to_deliver;

        /* deliver the data */

        if (playback_callback (frames_to_deliver) != frames_to_deliver) {
                fprintf (stderr, "playback callback failedn");
            break;
        }
    }

    snd_pcm_close (playback_handle);
    exit (0);
    
    return 0;
}  

The program is making a continuous tone but it is not 1 kHz. I am not sure if it is because the output is saturated (it is very loud and I haven’t figured out how to reduce the volume yet), or if I am generating/sending the sinewave samples to hardware incorrectly.

My questions are the following:

  1. Is the audio buffer (buf[]) argument to snd_pcm_writei() supposed to contains interleaved 2 channel data? Is it supposed to be contain audio samples in the following format: buf[0] = Ch_A_0, buf[1] = Ch_B_0, buf[2] = Ch_A_1, buf[3] = Ch_B_1, buf[4] = Ch_A_2, buf[5] = Ch_B_2, …, or is it only supposed to contain data for channel A and the hardware copied the same data to channel B?

  2. Is snd_pcm_writei() a blocking function? Will this program reach return err; only after everything in buf[] has been sent out to the playback hardware?

  3. What is the correct way to reduce volume in the ALSA api? I tried reducing the amplitude of the sinewave by calculating it as tempSin = sin(2.0*M_PI*1000*i/48000)*0.1; but the output is still the distorted tone sound, just a lot smaller. What is the ALSA function for reducing volume?

  4. Is there clear sample code available that shows how to read/write audio data frame by frame from/to the record/playback hardware? I also need it to have some kind of interrupt mechanism, where a callback function would be called when a frame of data is ready to be received/sent.

This is specifically for the raspberry pi zero W running raspberry pi OS, if that is relevant.

Source: Windows Questions C++

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