将 void* 转换为 memcpy 以获得浮点值
Cast void* to memcpy to get a float value
这是我试图解决的问题示例 我从麦克风获取缓冲区并尝试处理它的内容。根据这个问题
的友好指导
逻辑我声明一个向量来保存我想要的浮点数,然后将它的大小调整为 ArBuffer() 的大小,然后复制到向量。
ArBuffer() 是否必须将其转换为 memcpy?
#include "Lib_api.h"
#include <alsa/asoundlib.h>
#include <stdio.h>
#include "audiorecorder.h"
#include "Globals.h"
#include <iostream>
#include <inttypes.h>
#include <string.h>
#include <stdlib.h>
#include <vector>
#include <cstring>
using namespace std;
//Declare Creation
void* mCore;
int main(void)
{
// recorder
int rc;
int mode = 3;
const float sampleRate = 44100; //max 22Hz
int bufferSize = 1024; //Check this should be good 1024
//initialise
mCore = OXY_Create();
//initialise audio recorder
rc = arInitialise();
OXY_Configure(mode, sampleRate, bufferSize, mCore);
//initialise check hardware
if(rc)
{
std::cerr << "Fatal error: Audio could not be initialised" << rc << std::endl <<std::endl;
arFree();
exit(1);
}
//start recording
rc = arStartRecording();
//application loop
while(arIsRunning())
{
//declare vector
std::vector<float> values;
//resize values to size of arbuffersize
values.resize(arBufferSize(), sizeof(float));
//arBufferSize()/sizeof(float);
//need to cast this arBuffer() to memcpy?
std::memcpy(arBuffer(), &values[0], sizeof(values[0]));
// values[0] this will hold the latest data from the microphone?
int ret = OXY_DecodeAudioBuffer(&values[0], values.size(), mCore);
if (ret == -2)
{
std::cerr << "FOUND_TOKEN ---> -2 " << std::endl << std::endl;
}
else if(ret>=0)
{
std::cerr << "Decode started ---> -2 " << ret << std::endl << std::endl;
}
else if (ret == -3)
{
//int sizeStringDecoded = OXY_GetDecodedData(mStringDecoded, mCore);
std::cerr << "STRING DECODED ---> -2 " << std::endl << std::endl;
// ...
}
else
{
std::cerr << "No data found in this buffer" << std::endl << std::endl;
//no data found in this buffer
}
}
//Clean up
arFree();
return 0;
}
我将格式从 SND_PCM_FORMAT_S16_LE 更改为 SND_PCM_FORMAT_FLOAT_LE,这是另一个 SO 问题的善意建议。
* 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_FLOAT_LE)
/* 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);
}
values.resize(arBufferSize(), sizeof(float))
好吧,那不是我在其他评论中写的。您需要将缓冲区大小(以字节为单位)除以每个浮点数的字节数以获得浮点数:arBufferSize() / sizeof(float)
std::memcpy(arBuffer(), &values[0], sizeof(values[0]));
memcpy
由于历史原因,它的目的地和来源颠倒了。 const*
错误是因为您要求 memcpy
写入 arBuffer
.
此外,sizeof(values[0])
是一个浮点数的大小,以字节为单位。您已经有了 arBufferSize()
,这正是 memcpy
需要的大小。
这是我试图解决的问题示例 我从麦克风获取缓冲区并尝试处理它的内容。根据这个问题
逻辑我声明一个向量来保存我想要的浮点数,然后将它的大小调整为 ArBuffer() 的大小,然后复制到向量。
ArBuffer() 是否必须将其转换为 memcpy?
#include "Lib_api.h"
#include <alsa/asoundlib.h>
#include <stdio.h>
#include "audiorecorder.h"
#include "Globals.h"
#include <iostream>
#include <inttypes.h>
#include <string.h>
#include <stdlib.h>
#include <vector>
#include <cstring>
using namespace std;
//Declare Creation
void* mCore;
int main(void)
{
// recorder
int rc;
int mode = 3;
const float sampleRate = 44100; //max 22Hz
int bufferSize = 1024; //Check this should be good 1024
//initialise
mCore = OXY_Create();
//initialise audio recorder
rc = arInitialise();
OXY_Configure(mode, sampleRate, bufferSize, mCore);
//initialise check hardware
if(rc)
{
std::cerr << "Fatal error: Audio could not be initialised" << rc << std::endl <<std::endl;
arFree();
exit(1);
}
//start recording
rc = arStartRecording();
//application loop
while(arIsRunning())
{
//declare vector
std::vector<float> values;
//resize values to size of arbuffersize
values.resize(arBufferSize(), sizeof(float));
//arBufferSize()/sizeof(float);
//need to cast this arBuffer() to memcpy?
std::memcpy(arBuffer(), &values[0], sizeof(values[0]));
// values[0] this will hold the latest data from the microphone?
int ret = OXY_DecodeAudioBuffer(&values[0], values.size(), mCore);
if (ret == -2)
{
std::cerr << "FOUND_TOKEN ---> -2 " << std::endl << std::endl;
}
else if(ret>=0)
{
std::cerr << "Decode started ---> -2 " << ret << std::endl << std::endl;
}
else if (ret == -3)
{
//int sizeStringDecoded = OXY_GetDecodedData(mStringDecoded, mCore);
std::cerr << "STRING DECODED ---> -2 " << std::endl << std::endl;
// ...
}
else
{
std::cerr << "No data found in this buffer" << std::endl << std::endl;
//no data found in this buffer
}
}
//Clean up
arFree();
return 0;
}
我将格式从 SND_PCM_FORMAT_S16_LE 更改为 SND_PCM_FORMAT_FLOAT_LE,这是另一个 SO 问题的善意建议。
* 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_FLOAT_LE)
/* 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);
}
values.resize(arBufferSize(), sizeof(float))
好吧,那不是我在其他评论中写的。您需要将缓冲区大小(以字节为单位)除以每个浮点数的字节数以获得浮点数:arBufferSize() / sizeof(float)
std::memcpy(arBuffer(), &values[0], sizeof(values[0]));
memcpy
由于历史原因,它的目的地和来源颠倒了。 const*
错误是因为您要求 memcpy
写入 arBuffer
.
此外,sizeof(values[0])
是一个浮点数的大小,以字节为单位。您已经有了 arBufferSize()
,这正是 memcpy
需要的大小。