我正在尝试将char *转换为float * [英] Im trying to convert a char* to float*
问题描述
我正在尝试使用ALSA录制音频并将其传递以进行处理.音频样本从其中返回，即char *到float *
Im trying to record audio using ALSA and pass it to be processed. The audio sample is returned from this which is char* to a float*
我已经尝试了很多解决方案，我想我知道它实际上不是char缓冲区，而是字节缓冲区，而是如何使它成为浮点数.
Ive tried so many solutions I think I understand that it's not really a char buffer but a byte buffer but how I get it a float.
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值使用
I need to consume the output of the microphone as a float
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);
}
我正在编写程序以将上述内容用作api:
Im writing a program to consume the the above as api:
void* mOxyCore; is declared
然后我尝试传递arBuffer()，该参数不能按预期工作.
I then try and pass the arBuffer() which wouldn't work as expected.
while(arIsRunning())
{
int ret = DecodeAudioBuffer(arBuffer(), arBufferSize(), mCore);
}
Alsa:
/* 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);
}
推荐答案
这里的问题不是演员，而是代表制问题.
The problem here isn't a cast, but a representation issue.
通常将音频表示为一系列样本.表示每个样本的方法有很多:从-1.0f到+1.0f或从-32767到+32767的标度，或者许多其他方法.
Audio is generally represented as a series of samples. There are quite a few ways to represent each sample: on a scale from -1.0f to +1.0f, or -32767 to +32767, or many others.
Alsa实际上支持许多格式，然后选择SND_PCM_FORMAT_S16_LE，因此将-32767至+32767.您可以将其强制转换为std::int16_t*，前提是您的C ++环境是Little-Endian(几乎可以肯定).您无法将其投放到float*，因为您需要输入SND_PCM_FORMAT_FLOAT_LE
Alsa supports in fact many formats, and you chose SND_PCM_FORMAT_S16_LE so that's -32767 to +32767. You could cast that to std::int16_t*, assuming your C++ environment is Little-Endian (almost certain). You can't cast it to float*, for that you'd need to ask for SND_PCM_FORMAT_FLOAT_LE
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