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mp3格式转wav格式 附完整C++算法实现代码

时间:2018-01-14 12:06:25      阅读:734      评论:0      收藏:0      [点我收藏+]

标签:channel   rate   orm   就是   方式   channels   body   其他   sprintf   

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.\n");
        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 = \0;
        }
    }
    else if (drv)
        *drv = \0;
    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 = \0;
    }
    if (dir) {
        for (s = path; s < p;)
            *dir++ = *s++;
        *dir = \0;
    }
}


int main(int argc, char* argv[])
{
    std::cout << "Audio Processing " << std::endl;
    std::cout << "博客:http://tntmonks.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 \n" << 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。

  

若有其他相关问题或者需求也可以邮件联系俺探讨。

邮箱地址是: 
gaozhihan@vip.qq.com

 

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mp3格式转wav格式 附完整C++算法实现代码

标签:channel   rate   orm   就是   方式   channels   body   其他   sprintf   

原文地址:https://www.cnblogs.com/tntmonks/p/8282397.html

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