转载：https://www.cnblogs.com/cpuimage/p/8282397.html
近期偶然间看到一个开源项目minimp3
Minimalistic MP3 decoder single header library
项目地址:
https://github.com/lieff/minimp3
单文件头的最小mp3解码器。
一直很想抽时间好好看上一看。
最好的学习方式就是写个实用性的工程项目。
例如实现mp3转wav格式。
嗯，这篇博文就是这么来的。
阅读了下minimp3的源码，有一两处小bug，
这个解码算法可以进一步提速优化的地方还有不少。
后面有时间，再好好庖丁解牛。
基于这个库，实现mp3转wav的代码行数不到300行。
小巧而简洁，算是简单的抛砖引玉了。
个人习惯，很少写注释，
所以尽可能把代码写得清晰易懂，当然也有犯懒的时候。
完整代码:
#define _CRT_SECURE_NO_WARNINGS
#define _CRT_SECURE_NO_DEPRECATE 1
#define _CRT_NONSTDC_NO_DEPRECATE 1
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <time.h>
#include <iostream>
// ref:https://github.com/lieff/minimp3/blob/master/minimp3.h
#define MINIMP3_IMPLEMENTATION
#include "minimp3.h"
#include <sys/stat.h>
auto const epoch = clock();
static double now()
{
return (clock() - epoch);
};
template <typename FN>
static double bench(const FN &fn)
{
auto took = -now();
return (fn(), took + now()) / 1000;
}
//写wav文件
void wavWrite_int16(char* filename, int16_t* buffer, int sampleRate, uint32_t totalSampleCount, int channels = 1) {
FILE* fp = fopen(filename, "wb");
if (fp == NULL) {
printf("文件打开失败.
");
return;
}
//修正写入的buffer长度
totalSampleCount *= sizeof(int16_t)*channels;
int nbit = 16;
int FORMAT_PCM = 1;
int nbyte = nbit / 8;
char text[4] = { 'R', 'I', 'F', 'F' };
uint32_t long_number = 36 + totalSampleCount;
fwrite(text, 1, 4, fp);
fwrite(&long_number, 4, 1, fp);
text[0] = 'W';
text[1] = 'A';
text[2] = 'V';
text[3] = 'E';
fwrite(text, 1, 4, fp);
text[0] = 'f';
text[1] = 'm';
text[2] = 't';
text[3] = ' ';
fwrite(text, 1, 4, fp);
long_number = 16;
fwrite(&long_number, 4, 1, fp);
int16_t short_number = FORMAT_PCM;//默认音频格式
fwrite(&short_number, 2, 1, fp);
short_number = channels; // 音频通道数
fwrite(&short_number, 2, 1, fp);
long_number = sampleRate; // 采样率
fwrite(&long_number, 4, 1, fp);
long_number = sampleRate * nbyte; // 比特率
fwrite(&long_number, 4, 1, fp);
short_number = nbyte; // 块对齐
fwrite(&short_number, 2, 1, fp);
short_number = nbit; // 采样精度
fwrite(&short_number, 2, 1, fp);
char data[4] = { 'd', 'a', 't', 'a' };
fwrite(data, 1, 4, fp);
long_number = totalSampleCount;
fwrite(&long_number, 4, 1, fp);
fwrite(buffer, totalSampleCount, 1, fp);
fclose(fp);
}
//读取文件buffer
char *getFileBuffer(const char *fname, int *size)
{
FILE * fd = fopen(fname, "rb");
if (fd == 0)
return 0;
struct stat st;
char *file_buf = 0;
if (fstat(fileno(fd), &st) < 0)
goto doexit;
file_buf = (char *)malloc(st.st_size + 1);
if (file_buf != NULL)
{
if (fread(file_buf, st.st_size, 1, fd) < 1)
{
fclose(fd);
return 0;
}
file_buf[st.st_size] = 0;
}
if (size)
*size = st.st_size;
doexit:
fclose(fd);
return file_buf;
}
//mp3解码
int16_t* DecodeMp3ToBuffer(char* filename, uint32_t *sampleRate, uint32_t *totalSampleCount, unsigned int *channels)
{
int music_size = 0;
int alloc_samples = 1024 * 1024, num_samples = 0;
int16_t *music_buf = (int16_t *)malloc(alloc_samples * 2 * 2);
unsigned char *file_buf = (unsigned char *)getFileBuffer(filename, &music_size);
if (file_buf != NULL)
{
unsigned char *buf = file_buf;
mp3dec_frame_info_t info;
mp3dec_t dec;
mp3dec_init(&dec);
for (;;)
{
int16_t frame_buf[2 * 1152];
int samples = mp3dec_decode_frame(&dec, buf, music_size, frame_buf, &info);
if (alloc_samples < (num_samples + samples))
{
alloc_samples *= 2;
int16_t* tmp = (int16_t *)realloc(music_buf, alloc_samples * 2 * info.channels);
if (tmp)
music_buf = tmp;
}
if (music_buf)
memcpy(music_buf + num_samples*info.channels, frame_buf, samples*info.channels * 2);
num_samples += samples;
if (info.frame_bytes <= 0 || music_size <= (info.frame_bytes + 4))
break;
buf += info.frame_bytes;
music_size -= info.frame_bytes;
}
if (alloc_samples > num_samples)
{
int16_t* tmp = (int16_t *)realloc(music_buf, num_samples * 2 * info.channels);
if (tmp)
music_buf = tmp;
}
if (sampleRate)
*sampleRate = info.hz;
if (channels)
*channels = info.channels;
if (num_samples)
*totalSampleCount = num_samples;
free(file_buf);
return music_buf;
}
if (music_buf)
free(music_buf);
return 0;
}
//分割路径函数
void splitpath(const char* path, char* drv, char* dir, char* name, char* ext)
{
const char* end;
const char* p;
const char* s;
if (path[0] && path[1] == ':') {
if (drv) {
*drv++ = *path++;
*drv++ = *path++;
*drv = ' ';
}
}
else if (drv)
*drv = ' ';
for (end = path; *end && *end != ':';)
end++;
for (p = end; p > path && *--p != '\' && *p != '/';)
if (*p == '.') {
end = p;
break;
}
if (ext)
for (s = end; (*ext = *s++);)
ext++;
for (p = end; p > path;)
if (*--p == '\' || *p == '/') {
p++;
break;
}
if (name) {
for (s = p; s < end;)
*name++ = *s++;
*name = ' ';
}
if (dir) {
for (s = path; s < p;)
*dir++ = *s++;
*dir = ' ';
}
}
int main(int argc, char* argv[])
{
std::cout << "Audio Processing " << std::endl;
std::cout << "博客:http://cpuimage.cnblogs.com/" << std::endl;
std::cout << "mp3 转 wav." << std::endl;
if (argc < 2) return -1;
char* in_file = argv[1];
//总音频采样数
uint32_t totalSampleCount = 0;
//音频采样率
uint32_t sampleRate = 0;
//通道数
unsigned int channels = 0;
int16_t* wavBuffer = NULL;
double nLoadTime = bench([&]
{
wavBuffer = DecodeMp3ToBuffer(in_file, &sampleRate, &totalSampleCount, &channels);
});
std::cout << " 加载耗时: " << int(nLoadTime * 1000) << " 毫秒" << std::endl;
//保存结果
double nSaveTime = bench([&]
{
char drive[3];
char dir[256];
char fname[256];
char ext[256];
char out_file[1024];
splitpath(in_file, drive, dir, fname, ext);
sprintf(out_file, "%s%s%s.wav", drive, dir, fname);
wavWrite_int16(out_file, wavBuffer, sampleRate, totalSampleCount, channels);
});
std::cout << " 保存耗时: " << int(nSaveTime * 1000) << " 毫秒" << std::endl;
if (wavBuffer)
{
free(wavBuffer);
}
getchar();
std::cout << "按任意键退出程序
" << std::endl;
return 0;
}
示例具体流程为：
加载mp3(拖放mp3文件到可执行文件上)->解码mp3->保存wav
并对 加载，保存 这2个环节都进行了耗时计算并输出。
代码使用c++11标准编译，如果是之前老标准，则会编译报错。
编译： g++ -o mp3twav mp32wav.cpp --std=c++11
执行：./mp3twav output.mp3
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