一段时间以来我一直在尝试使用ALSA库,但我不知道该如何使用它。
我以示例程序为例,我尝试将其修改为使用float
(32位)而不是unsigned char
(8位)。但是现在,当我运行它时,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;
}
如何使用32位库?
我已经尝试过这种格式以及其他类似小端或大端的格式。唯一不会崩溃的是SND_PCM_FORMAT_FLOAT
,但它正在产生错误:
ALSA lib pcm.c:8507:(snd_pcm_set_params) Sample format not available for PLAYBACK: Invalid argument
Playback open error: Invalid argument
提前感谢。
P.S.:Linux,Ubuntu 19.10 64位
当您写入buffer
时可能已经出现分段错误:
for(i = 0; i < sizeof(buffer); i++)
buffer[i] = i % 255; // random();
sizeof(buffer)
将为您提供字节大小,而不是元素数。它们仅对于char
(和unsigned char
)相等,因为sizeof(char)
为1
。您最有可能想要遍历元素:
for(i = 0; i < sizeof buffer/sizeof *buffer; i++)
buffer[i] = i % 255; // random();
这确实是我的循环和我的snd_pcm_writei()
中的条件问题>
这里是没有错误的代码,感谢@Osiris:
#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;
}