libmad使用一步步进阶
卜瀚漠
原文地址：http://my.oschina.net/mjRao/blog/57874
libmad是一个开源mp3解码库，其对mp3解码算法做了很多优化，性能较好，很多播放器如mplayer、xmms等都是使用这个开源库进行解码的；如果要设计mp3播放器而又不想研究mp3解码算法的话，libmad是个不错的选择，可是问题来了：
- libmad配套的相关文档太少，可以说几乎没有，只有一个示例程序minimad.c，但没有一定经验的人根本不知道怎么编译这个minimad.c，就算是编译了也不知道怎么运行、怎么播放mp3；
- 网上讲libmad和minimad.c的文章很多，但能解释清楚的少之又少，大家都是抄来抄去，要么是不懂装懂，要么是懂了一点就自以为精通了，这样一来的结果是：在网上搜两天也弄不明白libmad究竟怎么使用。
所幸手里有Altera公司的一个工程，借助对该工程的分析、minimad.c中少的可怜的注释和网上搜索的Linux音频方面的相关知识，反复思考编码，总算把libmad库用起来了，现记录一下其使用方法，在帮助别人的同时也方便自己回头查询。 在开始之前，最好先把mp3文件格式和Linux音频编程方面的知识先学习一下，不然后面有的东西可能听不懂，还有就是一定要熟悉Linux系统，后面的代码都是在linux系统中用gcc编译的，在Windows下不能用的。 首先看下面几个问题，这也是我一开始最迷惑的，弄明白这几个问题了，也就对libmad库的使用相当熟悉了：
- minimad.c怎么编译？编译后怎么运行？运行时的输入输出分别是什么，或者说运行时什么效果？
- 怎样播放minimad输出的数据？或者说怎么播放解码后的数据？
- minimad运行时，mp3数据来源是标准输入，能不能改为从文件中读入数据？该怎么改？
- minimad运行时首先要将整个mp3文件读入内存，能不能改成边解码边读入的形式，比如每次读入16K，解码完再读入16K，而又不影响播放的连贯性，这样可以节省内存开销，方便在嵌入式系统中使用；
- 怎样用libmad做一个简单的mp3播放器?
一个一个来讲吧。
minimad.c怎么编译？编译后怎么运行？运行时的输入输出分别是什么，或者说运行时什么效果？ 在Linux下(我前面说了，本文所有的工作都是在Linux进行)先安装libmad，说白了就是把libmad库导入C标准库，安装方法见libmad-0.15.1b中的README和INSTALL文件。 安装libmad后，新建一个文件夹，将libmad-0.15.1b中的minimad.c和mad.h复制过来，用gcc编译minimad.c，编译命令为(假设要生成的可执行程序为minimad)：gcc -o minimad minimad.c -lmad minimad程序从标准输入读入mp3文件，然后将解码后的音频数据送到标准输出，我们可以用重定向的方式从文件中读入数据并将结果写至文件，命令如下：
./minimad tmp.pcm
怎样播放minimad输出的数据？或者说怎么播放解码后的数据？ 假设你有Linux音频编程方面的基础的话，这个应该不成问题，如果没有也没关系，在Linux的设计理念中，一切皆是文件，音频设备也是文件，只需要打 开/dev/dsp（音频设备)这个文件，然后将解码后的数据写入这个文件即可实现播放，新建pcmplay.c文件，拷入如下代码：
#include
#include
#include
#include
#include
#include
int main(int argc, char *argv[])
{
int id, fd, i;
char buf[1024];
int rate; /*simple rate 44.1KHz*/
int format; /*quatize args*/
int channels; /*sound channel*/
if(argc != 2)
{
fprintf(stderr, "usage : %s \n", argv[0]);
exit(-1);
}
if((fd = open(argv[1], O_RDONLY)) < 0)
{
fprintf(stderr, "Can't open sound file!\n");
exit(-2);
}
if((id = open("/dev/dsp", O_WRONLY)) 0)
{
write(id, buf, i);
//printf("i=%d\n", i);
}
close(fd);
close(id);
exit(0);
}gcc -o pcmplay pcmplay.c
./minimad tmp.pcm
./pcmplay tmp.pcmstruct buffer {
FILE *fp; /*file pointer*/
unsigned int flen; /*file length*/
unsigned int fpos; /*current position*/
unsigned char fbuf[BUFSIZE]; /*buffer*/
unsigned int fbsize; /*indeed size of buffer*/
};
typedef struct buffer mp3_file;static
enum mad_flow input(void *data,
struct mad_stream *stream)
{
mp3_file *mp3fp;
int ret_code;
int unproc_data_size; /*the unprocessed data's size*/
int copy_size;
mp3fp = (mp3_file *)data;
if(mp3fp->fpos flen)
{
unproc_data_size = stream->bufend - stream->next_frame;
memcpy(mp3fp->fbuf, mp3fp->fbuf+mp3fp->fbsize-unproc_data_size, unproc_data_size);
copy_size = BUFSIZE - unproc_data_size;
if(mp3fp->fpos + copy_size > mp3fp->flen)
{
copy_size = mp3fp->flen - mp3fp->fpos;
}
fread(mp3fp->fbuf+unproc_data_size, 1, copy_size, mp3fp->fp);
mp3fp->fbsize = unproc_data_size + copy_size;
mp3fp->fpos += copy_size;
/*Hand off the buffer to the mp3 input stream*/
mad_stream_buffer(stream, mp3fp->fbuf, mp3fp->fbsize);
ret_code = MAD_FLOW_CONTINUE;
}
else
{
ret_code = MAD_FLOW_STOP;
}
return ret_code;
}#include
#include
#include
#include
#include
#include
#include
#include
#include
#include "mad.h"
#define BUFSIZE 8192
/*
* This is a private message structure. A generic pointer to this structure
* is passed to each of the callback functions. Put here any data you need
* to access from within the callbacks.
*/
struct buffer {
FILE *fp; /*file pointer*/
unsigned int flen; /*file length*/
unsigned int fpos; /*current position*/
unsigned char fbuf[BUFSIZE]; /*buffer*/
unsigned int fbsize; /*indeed size of buffer*/
};
typedef struct buffer mp3_file;
int soundfd; /*soundcard file*/
unsigned int prerate = 0; /*the pre simple rate*/
int writedsp(int c)
{
return write(soundfd, (char *)&c, 1);
}
void set_dsp()
{
int format = AFMT_S16_LE;
int channels = 2;
soundfd = open("/dev/dsp", O_WRONLY);
ioctl(soundfd, SNDCTL_DSP_SETFMT, &format);
ioctl(soundfd, SNDCTL_DSP_CHANNELS, &channels);
}
/*
* This is perhaps the simplest example use of the MAD high-level API.
* Standard input is mapped into memory via mmap(), then the high-level API
* is invoked with three callbacks: input, output, and error. The output
* callback converts MAD's high-resolution PCM samples to 16 bits, then
* writes them to standard output in little-endian, stereo-interleaved
* format.
*/
static int decode(mp3_file *mp3fp);
int main(int argc, char *argv[])
{
long flen, fsta, fend;
int dlen;
mp3_file *mp3fp;
if (argc != 2)
return 1;
mp3fp = (mp3_file *)malloc(sizeof(mp3_file));
if((mp3fp->fp = fopen(argv[1], "r")) == NULL)
{
printf("can't open source file.\n");
return 2;
}
fsta = ftell(mp3fp->fp);
fseek(mp3fp->fp, 0, SEEK_END);
fend = ftell(mp3fp->fp);
flen = fend - fsta;
if(flen fp, 0, SEEK_SET);
fread(mp3fp->fbuf, 1, BUFSIZE, mp3fp->fp);
mp3fp->fbsize = BUFSIZE;
mp3fp->fpos = BUFSIZE;
mp3fp->flen = flen;
set_dsp();
decode(mp3fp);
close(soundfd);
fclose(mp3fp->fp);
return 0;
}
/*
* This is the input callback. The purpose of this callback is to (re)fill
* the stream buffer which is to be decoded. In this example, an entire file
* has been mapped into memory, so we just call mad_stream_buffer() with the
* address and length of the mapping. When this callback is called a second
* time, we are finished decoding.
*/
static
enum mad_flow input(void *data,
struct mad_stream *stream)
{
mp3_file *mp3fp;
int ret_code;
int unproc_data_size; /*the unprocessed data's size*/
int copy_size;
mp3fp = (mp3_file *)data;
if(mp3fp->fpos flen)
{
unproc_data_size = stream->bufend - stream->next_frame;
memcpy(mp3fp->fbuf, mp3fp->fbuf+mp3fp->fbsize-unproc_data_size, unproc_data_size);
copy_size = BUFSIZE - unproc_data_size;
if(mp3fp->fpos + copy_size > mp3fp->flen)
{
copy_size = mp3fp->flen - mp3fp->fpos;
}
fread(mp3fp->fbuf+unproc_data_size, 1, copy_size, mp3fp->fp);
mp3fp->fbsize = unproc_data_size + copy_size;
mp3fp->fpos += copy_size;
/*Hand off the buffer to the mp3 input stream*/
mad_stream_buffer(stream, mp3fp->fbuf, mp3fp->fbsize);
ret_code = MAD_FLOW_CONTINUE;
}
else
{
ret_code = MAD_FLOW_STOP;
}
return ret_code;
}
/*
* The following utility routine performs simple rounding, clipping, and
* scaling of MAD's high-resolution samples down to 16 bits. It does not
* perform any dithering or noise shaping, which would be recommended to
* obtain any exceptional audio quality. It is therefore not recommended to
* use this routine if high-quality output is desired.
*/
static inline
signed int scale(mad_fixed_t sample)
{
/* round */
sample += (1L <= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample > (MAD_F_FRACBITS + 1 - 16);
}
/*
* This is the output callback function. It is called after each frame of
* MPEG audio data has been completely decoded. The purpose of this callback
* is to output (or play) the decoded PCM audio.
*/
static
enum mad_flow output(void *data,
struct mad_header const *header,
struct mad_pcm *pcm)
{
unsigned int nchannels, nsamples;
unsigned int rate;
mad_fixed_t const *left_ch, *right_ch;
/* pcm->samplerate contains the sampling frequency */
rate= pcm->samplerate;
nchannels = pcm->channels;
nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
/* update the sample rate of dsp*/
if(rate != prerate)
{
ioctl(soundfd, SNDCTL_DSP_SPEED, &rate);
prerate = rate;
}
while (nsamples--) {
signed int sample;
/* output sample(s) in 16-bit signed little-endian PCM */
sample = scale(*left_ch++);
writedsp((sample >> 0) & 0xff);
writedsp((sample >> 8) & 0xff);
if (nchannels == 2) {
sample = scale(*right_ch++);
writedsp((sample >> 0) & 0xff);
writedsp((sample >> 8) & 0xff);
}
}
return MAD_FLOW_CONTINUE;
}
/*
* This is the error callback function. It is called whenever a decoding
* error occurs. The error is indicated by stream->error; the list of
* possible MAD_ERROR_* errors can be found in the mad.h (or stream.h)
* header file.
*/
static enum mad_flow error(void *data,
struct mad_stream *stream,
struct mad_frame *frame)
{
mp3_file *mp3fp = data;
fprintf(stderr, "decoding error 0x%04x (%s) at byte offset %u\n",
stream->error, mad_stream_errorstr(stream),
stream->this_frame - mp3fp->fbuf);
/* return MAD_FLOW_BREAK here to stop decoding (and propagate an error) */
return MAD_FLOW_CONTINUE;
}
/*
* This is the function called by main() above to perform all the decoding.
* It instantiates a decoder object and configures it with the input,
* output, and error callback functions above. A single call to
* mad_decoder_run() continues until a callback function returns
* MAD_FLOW_STOP (to stop decoding) or MAD_FLOW_BREAK (to stop decoding and
* signal an error).
*/
static int decode(mp3_file *mp3fp)
{
struct mad_decoder decoder;
int result;
/* configure input, output, and error functions */
mad_decoder_init(&decoder, mp3fp,
input, 0 /* header */, 0 /* filter */, output,
error, 0 /* message */);
/* start decoding */
result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);
/* release the decoder */
mad_decoder_finish(&decoder);
return result;
}
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