Generating puretone sinewave in raspberry pi zero using C++ and ALSA api

I am trying to figure out how to generate 1 kHz tone in raspberry pi zero W board using ALSA api and C++, but I don't seem to be getting anywhere. Strangely there seems to be no beginner friendly step-by-step tutorial on generating sound using ALSA, when I thought it was the most common way of generating sound in linux.

I am using a pi zero W board with raspberry pi OS. I have a USB sound card connected to a USB hub, which is connected to the USB port on the pi zero board. I can play wav files from both VLC and command line using aplay, so there is no problem with the hardware. I can also run the small test program given here: https://forums.raspberrypi.com/viewtopic.php?t=15075, so I don't think there is any problem with my ALSA installation either.

Attempt 1:
I have tried running the test/pcm.c sample code given in ALSA's website: https://www.alsa-project.org/alsa-do...c-example.html
I compile this as
Code:gcc main.cpp -o Main -lasound -Wall -fpermissiveI get a couple of warnings and then an error saying no match for operator++ for snd_pcm_format_t (which is an enum type). the error is specifically at the line Code:for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { (under case 'o' in the part that parses the command line arguments). I am stuck after this and not sure what to do.

Attempt 2:
I am trying out the program given here: http://equalarea.com/paul/alsa-audio.html, under "A Minimal Interrupt-Driven Program". This program compiles after some modifications, and I am manually calculating 1kHz sinewave samples at 48000 Hz sampling frequency. Problem is the generated tone doesn't sound anything like 1 kHz pure tone. I have a hunch this distortion is because the output is saturated, since it is quite loud, but I need some confirmation that this is the case. I have tried to simply reduce the amplitude of the sine function by multiplying it with 0.1, but that still generates the same distorted tone sound, only a lot smaller. I can't find the proper way to reduce the volume in ALSA api either. This is the code I am testing:

Code:#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)- 1);)
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)-1) );
}
#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 frames\n", 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 occured\n");
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 failed\n");
break;
}
}

snd_pcm_close (playback_handle);
exit (0);

return 0;
}Can someone please point to me what I am doing wrong in attempts 1 and 2 above?