How to know how much audio has been played?

[yangyagami]

I tried to get an answer from AI, but the result has discrepancies.

#include <stdio.h>

#include <iostream>
#include <cmath>
#include <thread>

#include <alsa/asoundlib.h>

#define SAMPLE_RATE 44100
#define FREQUENCY 440.0 // A4 note
#define DURATION 5.0    // seconds

using namespace std::chrono_literals;

int main() {
    snd_pcm_t *handle;
    snd_pcm_hw_params_t *params;
    unsigned int rate = SAMPLE_RATE;
    int dir;
    int pcm;

    pcm = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
    if (pcm < 0) {
        std::cerr << "Unable to open PCM device: " << snd_strerror(pcm) << std::endl;
        return 1;
    }

    snd_pcm_hw_params_alloca(&params);
    snd_pcm_hw_params_any(handle, params);

    snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
    snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
    snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir);
    snd_pcm_hw_params_set_channels(handle, params, 1);

    pcm = snd_pcm_hw_params(handle, params);
    if (pcm < 0) {
        std::cerr << "Unable to set HW parameters: " << snd_strerror(pcm) << std::endl;
        return 1;
    }

    int samples = static_cast<int>(DURATION * SAMPLE_RATE);
    int16_t *buffer = new int16_t[samples];

    for (int i = 0; i < samples; ++i) {
        buffer[i] = static_cast<int16_t>(32767 * sin(2 * M_PI * FREQUENCY * i / SAMPLE_RATE));
    }

    pcm = snd_pcm_writei(handle, buffer, samples);
    if (pcm < 0) {
        std::cerr << "Playback error: " << snd_strerror(pcm) << std::endl;
    }

    while (true) {
      snd_pcm_sframes_t delay;
      int rc = snd_pcm_delay(handle, &delay);
      if (rc == 0) {
        double played_sec = (samples - delay) / 44100.0f;
        printf("Played: %.2f sec\n", played_sec);
      } else {
        std::cerr << snd_strerror(rc) << std::endl;
        break;
      }
    }

    snd_pcm_drain(handle);

    delete[] buffer;
    snd_pcm_close(handle);

    return 0;
}

[perexg]

There is no simple answer. The audio buffer size has some limits usually and you should do non-blocking writes, if you want responsivity. The assumptions in this example are really wrong.