檔案大小 ＝ 取樣頻率 × 樣本大小 × 聲道數 × 秒數 CD音質(44100Hz, 16bits, Stereo)來錄音一分鐘 Size = 44100*2bytes*2*60s = 10584000(KB)
線上播放多了一個位元速率 (BitRate) 的規格， 它是每秒所佔用的頻寬，也就是壓縮過後每秒真正所佔用的空間。 例如 20Kbps，代表每秒鐘的資料量為 20Kb， 只要你的網路速度在 20Kbps 以上，就可以直接線上收聽， 不用等到整個檔案下載才播放。 20Kbps 的 RM 檔每分鐘佔用多少的空間： 每秒所佔空間： 20(Kb) ÷ 8 ＝ 2.5 (KB) 每分鐘所佔空間： 2.5 × 60 ＝ 150 (KB)
ref : Introduction to Sound Programming with ALSA
Simple Sound Recording
/* This example reads from the default PCM device and writes to standard output for 5 seconds of data. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); //can use "dsnoop_test", "pulghw", or others to replace "default" if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(¶ms); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val; while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; }16 bits, frames=32, 2 channel: size = frames * 4
16 bits, frames=32, 1 channel: size = frames * 2
- CD quality audio is stereo 16 bits 44.1 kHz which means that there are two channels each sampled at 44.1 kHz, and the raw data rate is therefore 44100 * 2 * 16 = 1411200 bits/sec = 176400 bytes/sec.
- 'stereo' = number of channels: 2
1 analog sample is represented with 16 bits = 2 bytes
1 frame represents 1 analog sample from all channels; here we have 2 channels, and so:
1 frame = (num_channels) * (1 sample in bytes) = (2 channels) * (2 bytes (16 bits) per sample) = 4 bytes (32 bits) - 假設某音频信号是採樣率為8kHz、雙通道、16bit，20ms一帧，则一帧音频数据的大小為: size = 8000 x 2 x 16bit x 0.02s = 5120 bit = 640 bytes
*/
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <alsa/asoundlib.h>
int main() {
long loops;
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
char *buffer;
/* Open PCM device for playback. */
rc = snd_pcm_open(&handle, "default",
SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 2);
/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);
/* Set period size to 32 frames. */
frames = 32;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params, &frames,
&dir);
size = frames * 4; /* 2 bytes/sample, 2 channels */
buffer = (char *) malloc(size);
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
/* 5 seconds in microseconds divided by
* period time */
loops = 5000000 / val;
while (loops > 0) {
loops--;
rc = read(0, buffer, size);
if (rc == 0) {
fprintf(stderr, "end of file on input\n");
break;
} else if (rc != size) {
fprintf(stderr,
"short read: read %d bytes\n", rc);
}
rc = snd_pcm_writei(handle, buffer, frames);
if (rc == -EPIPE) {
/* EPIPE means underrun */
fprintf(stderr, "underrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,
"error from writei: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr,
"short write, write %d frames\n", rc);
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
return 0;
}
compile:run:
./recode > sound.raw
./play < sound.raw
or
mplayer -rawaudio samplesize=2:channels=2 -demuxer rawaudio sound.raw
ref :
Introduction to Sound Programming with ALSA
FramesPeriods
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