当前位置:天才代写 > tutorial > C语言/C++ 教程 > 基于libmad的MP3解码播放器

基于libmad的MP3解码播放器

2017-11-03 08:00 星期五 所属: C语言/C++ 教程 浏览:1637

副标题#e#

libmad:是一个开源的高精度mpeg音频解码库,支持 MPEG-1(Layer I, Layer II 和 LayerIII(也就是 MP3)。LIBMAD 提供 24-bit 的 PCM 输出,完全是定点计较,很是适合没有浮点支持的平台上利用。利用 libmad 提供的一系列 API,就可以很是简朴地实现 MP3 数据解码事情。在 libmad 的源代码文件目次下的 mad.h 文件中,可以看到绝大部门该库的数据布局和 API 等。

PCM编码:即为脉冲代码调制编码。

PCM通过抽样,量化,编码三个步调将持续的模仿信号转换成数字编码。

libmad中的主要数据布局:

主要数据布局 浸染
struct mad_stream 存放解码前的Bitstream数据
struct mad_synth 存放解码合成滤波后的PCM数据
struct mad_pcm 界说了音频的采样率,声道个数和PCM采样数据,用来初始化音频
struct mad_frame 记录MPEG帧解码后PCM数据的数据布局,个中的mad_header用来记录MPEG帧的根基信息,好比MPEG层数、声道模式、流比特率、采样比特率。声道模式包罗单声道、双声道、连系立体混音道以及一般立体声。

MAD通过回调函数机制来实现解码,每个回调函数会返回一个列举范例mad_flow,通过mad_flow可以节制解码的进程。在未经处理惩罚的环境下,MAD一般输出32bit,以little endian名目存放在mad_fixed_t中的数据。可是大大都的声卡并能支持输出高达32bit精度的数据,因而还必需对mad_fixed_t举办量化,圆滑处理惩罚以及发抖,使到采样信号降到16bit精度。MAD认真的只是解码的进程,它事情进程是:从外部获取输入,逐帧解码,在解码的进程中返复书息,然后获得解码功效。开拓人员要手动配置输入输出。

在libmad中提供了一个解码源措施minimad.c,实现了将MP3文件解码成pcm数据,并将其数据显示在终端上。

此刻就以该源码措施为例,来写出我们本身的基于libmad的MP3播放器。

在我们打开我们的音频措施之时同时也打开我们的音频设备"/dev/dsp"。

static int sfd;     
if((sfd = open("/dev/dsp", O_WRONLY)) < 0)      
{     
    printf("can not open device!!!/n");     
    return 1;     
}


#p#副标题#e#

一般来说,我们的MP3文件都是立体音,有2个声道,由于要把pcm采样后并处理惩罚的数据放入一个char型的数组,而并行的阁下声道的每个采样要在字符数组中处理惩罚成2个,所以字符数组中的数据的个数应该是pcm音频采样数的4倍。又因为把阁下声道的数据合在一个字符数组里串行处理惩罚,所以播放的速度应该是pcm音频采样率的两倍。

static
enum mad_flow output(void *data,     
             struct mad_header const *header, struct mad_pcm *pcm)     
{     
    unsigned int nchannels, nsamples, n;     
    mad_fixed_t const *left_ch, *right_ch;     
    unsigned char Output[6912], *OutputPtr;     
    int fmt, wrote, speed;     
        
    nchannels = pcm->channels;     
    n = nsamples = pcm->length;     
    left_ch = pcm->samples[0];     
    right_ch = pcm->samples[1];     
        
    fmt = AFMT_S16_LE;     
    speed = pcm->samplerate * 2;    /*播放速度是采样率的两倍 */
    ioctl(sfd, SNDCTL_DSP_SPEED, &(speed));     
    ioctl(sfd, SNDCTL_DSP_SETFMT, &fmt);     
    ioctl(sfd, SNDCTL_DSP_CHANNELS, &(pcm->channels));     
    OutputPtr = Output;     
    while (nsamples--) {     
    signed int sample;     
    sample = scale(*left_ch++);     
    *(OutputPtr++) = sample >> 0;     
    *(OutputPtr++) = sample >> 8;     
    if (nchannels == 2) {     
        sample = scale(*right_ch++);     
        *(OutputPtr++) = sample >> 0;     
        *(OutputPtr++) = sample >> 8;     
    }     
    }     
    n *= 4;         /*数据长度为pcm音频采样的4倍 */
    OutputPtr = Output;     
    while (n) {     
    wrote = write(sfd, OutputPtr, n);     
    OutputPtr += wrote;     
    n -= wrote;     
    }     
    OutputPtr = Output;     
    return MAD_FLOW_CONTINUE;     
}

这样就可以实现我们的播放器了…..

下面就以一个简朴的实例来说明问题:

# include <stdio.h>     
# include <stdlib.h>     
# include <unistd.h>     
# include <sys/stat.h>     
# include <sys/mman.h>     
# include <sys/soundcard.h>     
# include <sys/ioctl.h>     
# include <sys/fcntl.h>     
# include <sys/types.h>     
# include <mad.h>     
struct buffer {     
    unsigned char const *start;     
    unsigned long length;     
};     
static int sfd;         /*声音设备的描写符 */
static int decode(unsigned char const *, unsigned long);     
int main(int argc, char *argv[])     
{     
    struct stat stat;     
    void *fdm;     
    char const *file;     
    int fd;     
    file = argv[1];     
    fd = open(file, O_RDONLY);     
    if ((sfd = open("/dev/dsp", O_WRONLY)) < 0) {     
    printf("can not open device!!!/n");     
    return 5;     
    }     
    ioctl(sfd, SNDCTL_DSP_SYNC, 0); /*此句可以不要 */
    if (fstat(fd, &stat) == -1 || stat.st_size == 0)     
    return 2;     
    fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);     
    if (fdm == MAP_FAILED)     
    return 3;     
    decode(fdm, stat.st_size);     
    if (munmap(fdm, stat.st_size) == -1)     
    return 4;     
    ioctl(sfd, SNDCTL_DSP_RESET, 0);     
    close(sfd);     
    return 0;     
}     
static
enum mad_flow input(void *data, struct mad_stream *stream)     
{     
    struct buffer *buffer = data;     
    if (!buffer->length)     
    return MAD_FLOW_STOP;     
    mad_stream_buffer(stream, buffer->start, buffer->length);     
    buffer->length = 0;     
    return MAD_FLOW_CONTINUE;     
}     
/*这一段是处理惩罚采样后的pcm音频 */
static inline signed int scale(mad_fixed_t sample)     
{     
    sample += (1L << (MAD_F_FRACBITS - 16));     
    if (sample >= MAD_F_ONE)     
    sample = MAD_F_ONE - 1;     
    else if (sample < -MAD_F_ONE)     
    sample = -MAD_F_ONE;     
    return sample >> (MAD_F_FRACBITS + 1 - 16);     
}     
static
enum mad_flow output(void *data,     
             struct mad_header const *header, struct mad_pcm *pcm)     
{     
    unsigned int nchannels, nsamples, n;     
    mad_fixed_t const *left_ch, *right_ch;     
    unsigned char Output[6912], *OutputPtr;     
    int fmt, wrote, speed;     
        
    nchannels = pcm->channels;     
    n = nsamples = pcm->length;     
    left_ch = pcm->samples[0];     
    right_ch = pcm->samples[1];     
        
    fmt = AFMT_S16_LE;     
    speed = pcm->samplerate * 2;    /*播放速度是采样率的两倍 */
    ioctl(sfd, SNDCTL_DSP_SPEED, &(speed));     
    ioctl(sfd, SNDCTL_DSP_SETFMT, &fmt);     
    ioctl(sfd, SNDCTL_DSP_CHANNELS, &(pcm->channels));     
    OutputPtr = Output;     
    while (nsamples--) {     
    signed int sample;     
    sample = scale(*left_ch++);     
    *(OutputPtr++) = sample >> 0;     
    *(OutputPtr++) = sample >> 8;     
    if (nchannels == 2) {     
        sample = scale(*right_ch++);     
        *(OutputPtr++) = sample >> 0;     
        *(OutputPtr++) = sample >> 8;     
    }     
    }     
    n *= 4;         /*数据长度为pcm音频采样的4倍 */
    OutputPtr = Output;     
    while (n) {     
    wrote = write(sfd, OutputPtr, n);     
    OutputPtr += wrote;     
    n -= wrote;     
    }     
    OutputPtr = Output;     
    return MAD_FLOW_CONTINUE;     
}     
        
static
enum mad_flow error(void *data,     
            struct mad_stream *stream, struct mad_frame *frame)     
{     
    return MAD_FLOW_CONTINUE;     
}     
        
static
int decode(unsigned char const *start, unsigned long length)     
{     
    struct buffer buffer;     
    struct mad_decoder decoder;     
    int result;     
    buffer.start = start;     
    buffer.length = length;     
    mad_decoder_init(&decoder, &buffer, input, 0, 0, output, error, 0);     
    mad_decoder_options(&decoder, 0);     
    result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);     
    mad_decoder_finish(&decoder);     
    return result;     
}

本文出自 “驿落薄暮” 博客,请务必保存此出处http://yiluohuanghun.blog.51cto.com/3407300/867922

 

    关键字:

天才代写-代写联系方式