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我正在按照此处的示例代码在 Linux 中使用 C++ 中的 ALSA API 生成纯音正弦波：http: //equalarea.com/paul/alsa-audio.html

我正在使用的代码位于“A Minimal Interrupt-Driven Program”下。原样的代码无法编译，我对实际编译的代码的修改版本发出了一些声音，如下所示：

#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 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;
}

程序发出连续音，但不是 1 kHz。我不确定这是因为输出饱和（它非常响亮，我还没有弄清楚如何降低音量），或者我是否错误地生成/发送正弦波样本到硬件。

我的问题如下：

  1. snd_pcm_writei() 的音频缓冲区 (buf[]) 参数是否应该包含交错的 2 通道数据？它是否应该包含以下格式的音频样本：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, ...，还是应该只包含通道 A 的数据，而硬件将相同的数据复制到通道 B？

  2. snd_pcm_writei() 是阻塞函数吗？这个程序return err;只有在 buf[] 中的所有内容都发送到播放硬件后才能到达吗？

  3. 在 ALSA api 中减少音量的正确方法是什么？我尝试通过计算降低正弦波的幅度，tempSin = sin(2.0*M_PI*1000*i/48000)*0.1;但输出仍然是失真的音调，只是小了很多。什么是用于减小音量的 ALSA 功能？

  4. 是否有可用的清晰示例代码显示如何从/向记录/播放硬件逐帧读取/写入音频数据？我还需要它具有某种中断机制，当一帧数据准备好接收/发送时，将调用回调函数。

这专门针对运行树莓派操作系统的树莓派零 W（如果相关）。

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