snd_pcm_hw_params_set_buffer_time_near

snd_pcm_hw_params_set_buffer_time_near函数是一个用于设置PCM硬件参数中缓冲区时间的函数。它接受一个PCM硬件参数对象和一个所需的缓冲区时间，并尝试将实际的缓冲区时间设置为最接近所需时间的值。该函数通常与其他设置PCM参数的函数一起使用，以确保设置合适的缓冲区时间来满足应用程序的需求。

相关问题

if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) { printf("Error setting parameters: %s\n", snd_strerror(err)); return 1; } snd_pcm_hw_params_free(hw_params); unsigned int frames_per_period = 2048; snd_pcm_uframes_t period_size = frames_per_period * CHANNELS * 2; snd_pcm_uframes_t buffer_size = period_size * 2; if ((err = snd_pcm_set_params(capture_handle, FORMAT, SND_PCM_ACCESS_RW_INTERLEAVED, CHANNELS, rate, 1, frames_per_period)) < 0) { printf("Error setting parameters: %s\n", snd_strerror(err)); return 1; } FILE *file = fopen(argv[1], "wb"); if (!file) { printf("Error opening file for writing!\n"); return 1; } uint32_t chunk_size = 0; uint32_t subchunk_size = 16; uint16_t audio_format = 1; uint16_t num_channels = CHANNELS; uint32_t sample_rate = rate; uint32_t byte_rate = (rate * CHANNELS * 2); uint16_t block_align = (CHANNELS * 2); uint16_t bits_per_sample = 16; uint32_t data_size = 0; fwrite("RIFF", 1, 4, file); fwrite(&chunk_size, 4, 1, file); fwrite("WAVE", 1, 4, file); fwrite("fmt ", 1, 4, file); fwrite(&subchunk_size, 4, 1, file); fwrite(&audio_format, 2, 1, file); fwrite(&num_channels, 2, 1, file); fwrite(&sample_rate, 4, 1, file); fwrite(&byte_rate, 4, 1, file); fwrite(&block_align, 2, 1, file); fwrite(&bits_per_sample, 2, 1, file); fwrite("data", 1, 4, file); fwrite(&data_size, 4, 1, file); uint16_t buffer[frames_per_period * CHANNELS]; while (1) { int n = snd_pcm_readi(capture_handle, buffer, frames_per_period); if (n < 0) { printf("Error reading from PCM device: %s\n", snd_strerror(n)); break; } fwrite(buffer, 2, n * CHANNELS, file); data_size += n * CHANNELS * 2; } uint32_t file_size = data_size + 36; fseek(file, 4, SEEK_SET); fwrite(&file_size, 4, 1, file); fseek(file, 40, SEEK_SET); fwrite(&data_size, 4, 1, file); fclose(file); snd_pcm_close(capture_handle); 加上注释

/*
这段代码是一个录音程序，使用 ALSA 库进行音频捕获，将捕获到的音频写入到一个 WAV 文件中。
在程序中，首先进行 ALSA 库的初始化，然后设置音频捕获参数，包括采样率、声道数、采样大小等。
然后打开一个 WAV 文件，写入 WAV 文件头部信息，开始捕获音频并将捕获到的音频数据写入到 WAV 文件中。
最后，关闭 WAV 文件和 ALSA 库。
*/

// 引入 ALSA 库的头文件
#include <alsa/asoundlib.h>

int main(int argc, char **argv) {

// 定义 ALSA 相关的变量
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
int err;

// 打开默认音频捕获设备并进行初始化
if ((err = snd_pcm_open(&capture_handle, "default", SND_PCM_STREAM_CAPTURE, 0)) < 0) {
printf("Error opening PCM device: %s\n", snd_strerror(err));
return 1;
}

// 申请 ALSA 配置参数结构体，进行初始化
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
printf("Error allocating hardware parameter structure: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
printf("Error initializing hardware parameter structure: %s\n", snd_strerror(err));
return 1;
}

// 设置音频捕获参数
unsigned int rate = 44100;
int CHANNELS = 2;
if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
printf("Error setting access type: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
printf("Error setting sample format: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, CHANNELS)) < 0) {
printf("Error setting channel count: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &rate, 0)) < 0) {
printf("Error setting sample rate: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params, &frames_per_period, 0)) < 0) {
printf("Error setting period size: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_buffer_size_near(capture_handle, hw_params, &buffer_size)) < 0) {
printf("Error setting buffer size: %s\n", snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
printf("Error setting parameters: %s\n", snd_strerror(err));
return 1;
}

// 释放 ALSA 配置参数结构体
snd_pcm_hw_params_free(hw_params);

// 计算每个周期的帧数和缓冲区大小
unsigned int frames_per_period = 2048;
snd_pcm_uframes_t period_size = frames_per_period * CHANNELS * 2;
snd_pcm_uframes_t buffer_size = period_size * 2;

// 设置音频捕获参数
if ((err = snd_pcm_set_params(capture_handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, CHANNELS, rate, 1, frames_per_period)) < 0) {
printf("Error setting parameters: %s\n", snd_strerror(err));
return 1;
}

// 打开要写入的 WAV 文件
FILE *file = fopen(argv[1], "wb");
if (!file) {
printf("Error opening file for writing!\n");
return 1;
}

// 写入 WAV 文件头部信息
uint32_t chunk_size = 0;
uint32_t subchunk_size = 16;
uint16_t audio_format = 1;
uint16_t num_channels = CHANNELS;
uint32_t sample_rate = rate;
uint32_t byte_rate = (rate * CHANNELS * 2);
uint16_t block_align = (CHANNELS * 2);
uint16_t bits_per_sample = 16;
uint32_t data_size = 0;
fwrite("RIFF", 1, 4, file);
fwrite(&chunk_size, 4, 1, file);
fwrite("WAVE", 1, 4, file);
fwrite("fmt ", 1, 4, file);
fwrite(&subchunk_size, 4, 1, file);
fwrite(&audio_format, 2, 1, file);
fwrite(&num_channels, 2, 1, file);
fwrite(&sample_rate, 4, 1, file);
fwrite(&byte_rate, 4, 1, file);
fwrite(&block_align, 2, 1, file);
fwrite(&bits_per_sample, 2, 1, file);
fwrite("data", 1, 4, file);
fwrite(&data_size, 4, 1, file);

// 读取音频数据并写入 WAV 文件
uint16_t buffer[frames_per_period * CHANNELS];
while (1) {
int n = snd_pcm_readi(capture_handle, buffer, frames_per_period);
if (n < 0) {
printf("Error reading from PCM device: %s\n", snd_strerror(n));
break;
}
fwrite(buffer, 2, n * CHANNELS, file);
data_size += n * CHANNELS * 2;
}

// 计算文件大小并更新 WAV 文件头部信息
uint32_t file_size = data_size + 36;
fseek(file, 4, SEEK_SET);
fwrite(&file_size, 4, 1, file);
fseek(file, 40, SEEK_SET);
fwrite(&data_size, 4, 1, file);

// 关闭 WAV 文件和 ALSA 库
fclose(file);
snd_pcm_close(capture_handle);

return 0;
}

tinyalsa pcm读取

TinyALSA是一个轻量级的音频库，可以在嵌入式平台上使用。它包含了PCM读取函数，可以用来读取音频数据。

以下是使用TinyALSA读取PCM数据的步骤：

1. 打开PCM设备：

int pcm = open("/dev/snd/pcmC0D0p", O_RDONLY);

2. 配置PCM参数，例如采样率、声道数、音频格式等：

unsigned int rate = 44100;
unsigned int channels = 2;
unsigned int format = SND_PCM_FORMAT_S16_LE;

snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(&params);
snd_pcm_hw_params_any(pcm, params);
snd_pcm_hw_params_set_access(pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(pcm, params, format);
snd_pcm_hw_params_set_channels(pcm, params, channels);
snd_pcm_hw_params_set_rate_near(pcm, params, &rate, 0);
snd_pcm_hw_params(pcm, params);

3. 分配音频缓冲区：

#define BUFFER_SIZE 1024
char buffer[BUFFER_SIZE];

4. 读取PCM数据：

while (true) {
    int count = read(pcm, buffer, BUFFER_SIZE);
    if (count <= 0) {
        break;
    }
    // 处理PCM数据
}

5. 关闭PCM设备：

close(pcm);

需要注意的是，以上代码只是一个简单的示例，实际使用中需要根据具体情况进行修改。同时，需要处理可能出现的错误情况，例如打开设备失败、配置参数失败等。