第一篇文章，我正在尝试使用 libsamplerate 和 libsndfile 制作一个简单的音高移位器。我通过制作一个简单的采样率转换器然后破解它以最基本的形式实现了这一点，我通过改变比率浮点值来改变音高。pitchshifter 移位 - 在正弦音上听起来非常好 - 如果您将它用于音频，您可以听到声音块之间的间隙 - 特别是如果您将文件调高。我想知道是否有一种方法可以使代码更有效地应对这种情况或某种插值函数或库，我可以毫不费力地实现它们。我对 C 很陌生——以前只通过 PD 处理声音，这是我的第一个项目——据我所知 libsamplerate 不是
谢谢
这是我的代码
#include <stdio.h>
#include </usr/local/include/sndfile.h>
#include </usr/local/include/samplerate.h>
#define BUFFER_LEN 44100 //defines buffer length
#define MAX_CHANNELS 2 //defines max channels
int main ()
{
static float datain [BUFFER_LEN]; //static defines as a global variable
static float dataout [BUFFER_LEN]; //static defines as a global variable
SNDFILE *infile, *outfile; //determines file open function + pointers
/*descriptor*/SF_INFO /*sf_open*/ sfinfo, sfinfo2;
int readcount;//used to store data in while ((readcount = sf...
const char *infilename/*pointer*/ = "/tmp/input.wav"; //const means that it is a value that cannot change
//after initialisation
const char *outfilename/*pointer*/ = "/tmp/soundchanged.wav"; //const means that it is a value that cannot change
//after initialisation
SRC_DATA src_data; //struct from libsamplerate library
//http://www.mega-nerd.com/SRC/api_misc.html#SRC_DATA
infile = sf_open (infilename/*pointer*/, SFM_READ, &sfinfo); //infile declares a file variable, SFM_READ-reads file
//sfinfo -function of sfopen
outfile = sf_open (outfilename/*pointer*/, SFM_WRITE, &sfinfo); //outfile declares a file variable, SFM_WRITE-writes file
//sfinfo -function of sfopen
src_data.data_in = datain; //used to pass audio data into the converter
src_data.input_frames = BUFFER_LEN; //supply the converter with the lengths of the arrays
//(in frames) pointed to by the data_in
src_data.data_out = dataout; //supplies the converter with an array to hold the converter's output
src_data.output_frames = BUFFER_LEN; //supply the converter with the lengths of the arrays
//(in frames) pointed to by the data_out
/*------->*/src_data.src_ratio = 0.2 /*changing this number changes the pitch of the output file*/;
//specifies the conversion ratio defined as the input sample rate
//divided by the output sample rate
src_simple (&src_data/*reference address of src_data*/, SRC_SINC_BEST_QUALITY, 1);//
while ((readcount = sf_read_float (infile, datain, BUFFER_LEN)))//while readcount is equal to
//sf_read_float - function call: infile,datain and BUFFER_LEN
//this data is then fed into the converter argument below
{
src_simple (&src_data, SRC_SINC_BEST_QUALITY, 1); //selects converter
//http://www.mega-nerd.com/SRC/api_misc.html#SRC_DATA
sf_write_float (outfile, dataout, readcount);
//write the data in the array pointed to by ptr to the file
};
sf_close (infile);
sf_close (outfile); // closes infile,outfile
//The close function closes the file, deallocates
//its internal buffers and returns 0 on success or an error value otherwise.
sf_open ("/tmp/soundchanged.wav", SFM_READ, &sfinfo2/*reference address of sfinfo2*/);
printf("%d", sfinfo2.samplerate);//outputs samplerate
return 0;
}