使用PCM格式的ALSA进行音频录制 [英] Audio recording using ALSA in PCM format
本文介绍了使用PCM格式的ALSA进行音频录制的处理方法，对大家解决问题具有一定的参考价值，需要的朋友们下面随着小编来一起学习吧！
问题描述
我正在Linux平台上使用ALSA进行音频捕获.
I am working on audio capturing using ALSA in linux platform.
我可以使用下面的代码捕获音频，在该代码中，我将默认"设备传递给参数，它将音频数据转储到 in.pcm 文件中.
I am able to capture audio using below code where I am passing "default" device into argument, and it will dump audio data into in.pcm file.
但是，当我尝试播放 in.pcm 文件时，我只听到杂音.我正在尝试使用以下命令播放音频:
However, when I tried to play in.pcm file, I hear only noise. I am trying to play audio using below command:
ffplay -autoexit -f f32le -ac 1 -ar 44100 in.pcm
代码:
#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
main (int argc, char *argv[])
{
int i;
int err;
char *buffer;
int buffer_frames = 128;
unsigned int rate = 44100;
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
snd_pcm_format_t format = SND_PCM_FORMAT_FLOAT;
// snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
argv[1],
snd_strerror (err));
exit (1);
}
fprintf(stdout, "audio interface opened\n");
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params allocated\n");
if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params initialized\n");
if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params access setted\n");
if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params format setted\n");
if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params rate setted\n");
if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params channels setted\n");
if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params setted\n");
snd_pcm_hw_params_free (hw_params);
fprintf(stdout, "hw_params freed\n");
if ((err = snd_pcm_prepare (capture_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "audio interface prepared\n");
buffer = malloc(128 * snd_pcm_format_width(format) / 8 * 2);
fprintf(stdout, "buffer allocated\n");
FILE *fp = fopen("in.pcm", "a+");
//for (i = 0; i < 10; ++i) {
i = 0;
while(++i) {
snd_pcm_wait(capture_handle, 1000);
if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) {
fprintf (stderr, "read from audio interface failed (%s)\n",
snd_strerror (err));
exit (1);
}
fwrite(buffer, 1, buffer_frames, fp);
fprintf(stdout, "read %d done\n", i);
}
fclose(fp);
free(buffer);
fprintf(stdout, "buffer freed\n");
snd_pcm_close (capture_handle);
fprintf(stdout, "audio interface closed\n");
exit (0);
}
有人可以告诉我问题是什么吗?
Can some one tell me what is issue ?
提前谢谢.
推荐答案
我通过更改write函数的缓冲区大小并附加了我的工作代码来解决了此问题:
I fixed this issue by changing buffer size of write function, attaching my working code:
#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
main (int argc, char *argv[])
{
int i;
int err;
char *buffer;
int buffer_frames = 128;
unsigned int rate = 44100;
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
snd_pcm_format_t format = SND_PCM_FORMAT_FLOAT;
// snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
argv[1],
snd_strerror (err));
exit (1);
}
fprintf(stdout, "audio interface opened\n");
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params allocated\n");
if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params initialized\n");
if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params access setted\n");
if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params format setted\n");
if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params rate setted\n");
if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params channels setted\n");
if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "hw_params setted\n");
snd_pcm_hw_params_free (hw_params);
fprintf(stdout, "hw_params freed\n");
if ((err = snd_pcm_prepare (capture_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
fprintf(stdout, "audio interface prepared\n");
buffer = malloc(128 * snd_pcm_format_width(format) / 8 * 2);
fprintf(stdout, "buffer allocated %d\n", snd_pcm_format_width(format) / 8 * 2);
int fd = open("in.pcm", O_CREAT | O_RDWR, 0666);
//for (i = 0; i < 10; ++i) {
i = 0;
while(++i) {
//snd_pcm_wait(capture_handle, 1000);
if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) {
fprintf (stderr, "read from audio interface failed (%s)\n",
snd_strerror (err));
exit (1);
}
write(fd, buffer, 128 * snd_pcm_format_width(format) / 8 * 2);
fprintf(stdout, "read %d done\n", i);
}
close(fd);
free(buffer);
fprintf(stdout, "buffer freed\n");
snd_pcm_close (capture_handle);
fprintf(stdout, "audio interface closed\n");
exit (0);
}
这篇关于使用PCM格式的ALSA进行音频录制的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！
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