Alsa 库 32 位

Question

我已经尝试使用 ALSA 库一段时间了，但我不明白应该如何使用它。

我拿了一个示例程序，我试图修改它以使用 float（32 位）而不是 unsigned char（8 位）。但是现在当我运行它时，我在第二个循环中出现了分段错误。

这是我的代码：

#include <alsa/asoundlib.h>




snd_pcm_t *create_pcm(const char* name, snd_pcm_stream_t mode, snd_pcm_format_t format, snd_pcm_access_t access, unsigned int nbChannel, unsigned int rate, int softSample, unsigned int latency)
{
    int err;
    snd_pcm_t *handle;

    if ((err = snd_pcm_open(&handle, name, mode, 0)) < 0) {
        printf("Playback open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }

    if ((err = snd_pcm_set_params(handle,
                                  format,
                                  access,
                                  nbChannel,
                                  rate,
                                  softSample,
                                  latency)) < 0) {   /* 0.5sec */
        printf("Playback open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }

    return handle;
}



int main(void)
{
    unsigned int i;
    snd_pcm_t *handle;
    snd_pcm_sframes_t frames;
    float buffer[16*1024];              /* some random data */


    handle = create_pcm("default", // name of the device used by the sound card
                        SND_PCM_STREAM_PLAYBACK, // to use the device in output
                        SND_PCM_FORMAT_FLOAT, // use the device with 32bit depth (float)
                        SND_PCM_ACCESS_RW_INTERLEAVED,
                        1, // use 1 channel
                        48000, // use 48000 Hz (dvd quality)
                        1, // soft resample ON
                        500000); // 0.5s of latency


    // building random data
    for(i = 0; i < sizeof(buffer); i++)
        buffer[i] = i % 255; // random();





    for (i = 0; i < 16; i++) {
        frames = snd_pcm_writei(handle, buffer, sizeof(buffer)); // segmentation fault
        if(frames < 0)
            frames = snd_pcm_recover(handle, frames, 0);
        if (frames < 0) {
            printf("snd_pcm_writei failed: %s\n", snd_strerror(frames));
            break;
        }
        if (frames > 0 && frames < (long)sizeof(buffer))
            printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), frames);
    }

    snd_pcm_close(handle);
    return 0;
}

如何在 32 位上使用这个库？

我试过这种格式和其他格式，如小端或大端。唯一没有崩溃的是 SND_PCM_FORMAT_FLOAT，但它出错了：

ALSA lib pcm.c:8507:(snd_pcm_set_params) Sample format not available for PLAYBACK: Invalid argument
Playback open error: Invalid argument

提前致谢。

P.S.: Linux, Ubuntu 19.10 64 位

Answer 1

当您写入 buffer:

时，分段错误可能已经发生

for(i = 0; i < sizeof(buffer); i++)
    buffer[i] = i % 255; // random();

sizeof(buffer) 将为您提供以字节为单位的大小，而不是元素的数量。它们仅对 char（和 unsigned char）相等，因为 sizeof(char) 是 1。您很可能想要迭代元素：

for(i = 0; i < sizeof buffer/sizeof *buffer; i++)
    buffer[i] = i % 255; // random();

Answer 2

这确实是我的循环和我的 snd_pcm_writei()

中的条件问题

感谢@Osiris，这是没有错误的代码：

#include <alsa/asoundlib.h>




snd_pcm_t *create_pcm(const char* name, snd_pcm_stream_t mode, snd_pcm_format_t format, snd_pcm_access_t access, unsigned int nbChannel, unsigned int rate, int softSample, unsigned int latency)
{
    int err;
    snd_pcm_t *handle;

    if ((err = snd_pcm_open(&handle, name, mode, 0)) < 0) {
        printf("Playback open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }

    if ((err = snd_pcm_set_params(handle,
                                  format,
                                  access,
                                  nbChannel,
                                  rate,
                                  softSample,
                                  latency)) < 0) {   /* 0.5sec */
        printf("Playback open error: %s\n", snd_strerror(err));
        exit(EXIT_FAILURE);
    }

    return handle;
}



int main(void)
{
    unsigned int i;
    snd_pcm_t *handle;
    snd_pcm_sframes_t frames;
    float buffer[16*1024];              /* some random data */


    handle = create_pcm("default", // name of the device used by the sound card
                        SND_PCM_STREAM_PLAYBACK, // to use the device in output
                        SND_PCM_FORMAT_FLOAT, // use the device with 32bit depth (float)
                        SND_PCM_ACCESS_RW_INTERLEAVED,
                        1, // use 1 channel
                        48000, // use 48000 Hz (dvd quality)
                        1, // soft resample ON
                        500000); // 0.5s of latency


    // building random data
    for(i = 0; i < sizeof(buffer) / sizeof(*buffer); i++)
        buffer[i] = i % 0xffffffff; // random();





    for (i = 0; i < 16; i++) {
        frames = snd_pcm_writei(handle, buffer, sizeof(buffer) / sizeof(*buffer)); // segmentation fault
        if(frames < 0)
            frames = snd_pcm_recover(handle, frames, 0);
        if (frames < 0) {
            printf("snd_pcm_writei failed: %s\n", snd_strerror(frames));
            break;
        }
        if (frames > 0 && frames < (long)(sizeof(buffer) / sizeof(*buffer)))
            printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), frames);
    }

    snd_pcm_close(handle);
    return 0;
}

Alsa 库 32 位

Alsa lib 32 bits

c

linux

audio

alsa