c++ - 我正在尝试将 char* 转换为 float*
问题描述
我正在尝试使用 ALSA 录制音频并将其传递给进行处理。音频样本从 this 返回,它是 char* 到 float*
我尝试了很多解决方案,我想我明白它不是真正的字符缓冲区而是字节缓冲区,而是我如何让它成为浮点数。
这将返回缓冲区:
const unsigned char* arBuffer(void)
{
return buffer;
}
我需要将麦克风的输出作为浮点数消耗
int32_t O_DecodeAudioBuffer(float *audioBuffer, int size, void *oxyingObject)
{
Core *oxying = (COxyCore*)oxyingObject;
//Decode audioBuffer to check if begin token is found, we should keep previous buffer to check if token was started in previous
//var mDecoding > 0 when token has been found, once decoding is finished, mDecoding = 0
return oxying->mDecoder->DecodeAudioBuffer(audioBuffer, size);
}
我正在编写一个程序以将上述内容作为 api 使用:
void* mOxyCore; is declared
然后,我尝试传递无法按预期工作的 arBuffer()。
while(arIsRunning())
{
int ret = DecodeAudioBuffer(arBuffer(), arBufferSize(), mCore);
}
阿尔萨:
/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include <pthread.h>
#include "settings.h"
#include "audiorecorder.h"
pthread_t thr;
pthread_mutex_t mutex;
snd_pcm_t *handle;
snd_pcm_uframes_t frames;
unsigned char* buffer;
BOOL running;
size_t buffersize;
BOOL arIsRunning(void)
{
return running;
}
void arAcquireBuffer(void)
{
//printf("Acquired buffer\n");
pthread_mutex_lock(&mutex);
}
void arReleaseBuffer(void)
{
//printf("Released buffer\n");
pthread_mutex_unlock(&mutex);
}
const unsigned char* arBuffer(void)
{
return buffer;
}
const size_t arBufferSize(void)
{
return buffersize;
}
void* entry_point(void *arg)
{
int rc;
fprintf(stderr, "Listening...\n");
while (running)
{
arAcquireBuffer();
rc = snd_pcm_readi(handle, buffer, frames);
//stream to stdout - useful for testing/debugging
//write(1, buffer, buffersize);
arReleaseBuffer();
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));
running = FALSE;
}
else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
}
return NULL;
}
int arInitialise(void)
{
snd_pcm_hw_params_t *params;
unsigned int val;
int rc, dir;
running = FALSE;
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&handle, RECORDER_DEVICE, SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0) {
fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc));
return rc;
}
else
{
fprintf(stderr, "Successfully opened default capture device.\n");
}
/* 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);
fprintf(stderr, "Format set to PCM Signed 16bit Little Endian.\n");
/* Channels */
snd_pcm_hw_params_set_channels(handle, params, NUM_CHANNELS);
fprintf(stderr, "Channels set to %d.\n", NUM_CHANNELS);
/* sampling rate */
val = SAMPLE_RATE;
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
fprintf(stderr, "Samplerate set to %d.\n", val);
/* Set period to FRAMES_PER_BUFFER frames. */
frames = FRAMES_PER_BUFFER;
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));
return rc;
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
buffersize = frames * 2 * NUM_CHANNELS; /* 2 bytes/sample * channels */
buffer = (unsigned char*) malloc(buffersize);
/* We want to loop forever */
//snd_pcm_hw_params_get_period_time(params, &val, &dir);
return 0;
}
int arStartRecording(void)
{
if(running) return 1;
if(pthread_mutex_init(&mutex, NULL))
{
printf("Unable to initialize mutex\n");
return -1;
}
if(pthread_create(&thr, NULL, &entry_point, NULL))
{
fprintf(stderr, "Could not create recorder thread!\n");
running = FALSE;
return -1;
}
running = TRUE;
return 0;
}
void arStopRecording(void)
{
running = FALSE;
}
void arFree(void)
{
running = FALSE;
sleep(500);
snd_pcm_drain(handle);
snd_pcm_close(handle);
pthread_mutex_destroy(&mutex);
free(buffer);
}
解决方案
这里的问题不是演员表,而是代表问题。
音频通常表示为一系列样本。有很多方法可以表示每个样本:从-1.0f
到+1.0f
,或-32767
到+32767
,或许多其他的尺度。
Alsa其实支持很多格式,你选的就是SND_PCM_FORMAT_S16_LE
这样。假设您的 C++ 环境是 Little-Endian(几乎可以肯定),您可以将其转换为。您不能将其转换为 ,因为您需要要求-32767
+32767
std::int16_t*
float*
SND_PCM_FORMAT_FLOAT_LE
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