/*
* libmad - MPEG audio decoder library
* Copyright (C) 2000-2004 Underbit Technologies, Inc.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* $Id: minimad.c,v 1.4 2004/01/23 09:41:32 rob Exp $
*/
# include <stdio.h>
# include <unistd.h>
# include <sys/stat.h>
# include <sys/mman.h>
#include <string.h>
#include<fcntl.h>
#include <stdlib.h>
#include <sys/ioctl.h>
#include <alsa/asoundlib.h>
# include "mad.h"
/*
* 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(unsigned char const *, unsigned long);
int set_pcm();
snd_pcm_t* handle=NULL; //PCI设备句柄
snd_pcm_hw_params_t* params=NULL;//硬件信息和PCM流配置
int main(int argc, char *argv[])
{
struct stat stat;
void *fdm;
if (argc != 2)
{
printf("Usage: minimad + mp3 file name");
return 1;
}
int fd;
fd=open(argv[1],O_RDWR);
if(fd<0)
{
perror("open file failed:");
return 1;
}
if (fstat(fd, &stat) == -1 ||stat.st_size == 0)
{
printf("fstat failed:\n");
return 2;
}
//printf("stat.st_size=%d\n",stat.st_size);
fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);
if (fdm == MAP_FAILED)
return 3;
if(set_pcm()!=0) //设置pcm 参数
{
printf("set_pcm fialed:\n");
return 1;
}
decode(fdm, stat.st_size);
if (munmap(fdm, stat.st_size) == -1)
return 4;
snd_pcm_drain(handle);
snd_pcm_close(handle);
return 0;
}
int set_pcm()
{
int rc;
int dir=0;
int rate = 44100;; /* 采样频率 44.1KHz*/
int format = SND_PCM_FORMAT_S16_LE; /* 量化位数 16 */
int channels = 2; /* 声道数 2 */
rc=snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if(rc<0)
{
perror("\nopen PCM device failed:");
exit(1);
}
snd_pcm_hw_params_alloca(¶ms); //分配params结构体
rc=snd_pcm_hw_params_any(handle, params);//初始化params
if(rc<0)
{
perror("\nsnd_pcm_hw_params_any:");
exit(1);
}
rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); //初始化访问权限
if(rc<0)
{
perror("\nsed_pcm_hw_set_access:");
exit(1);
}
rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); //设置16位采样精度
if(rc<0)
{
perror("snd_pcm_hw_params_set_format failed:");
exit(1);
}
rc=snd_pcm_hw_params_set_channels(handle, params, channels); //设置声道,1表示单声>道，2表示立体声
if(rc<0)
{
perror("\nsnd_pcm_hw_params_set_channels:");
exit(1);
}
rc=snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir); //设置>频率
if(rc<0)
{
perror("\nsnd_pcm_hw_params_set_rate_near:");
exit(1);
}
rc = snd_pcm_hw_params(handle, params);
if(rc<0)
{
perror("\nsnd_pcm_hw_params: ");
exit(1);
}
return 0;
}
/*
* 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 {
unsigned char const *start;
unsigned long length;
};
/*
* 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)
{
struct buffer *buffer = data;
printf("this is input\n");
if (!buffer->length)
return MAD_FLOW_STOP;
mad_stream_buffer(stream, buffer->start, buffer->length);
buffer->length = 0;
return MAD_FLOW_CONTINUE;
}
/*
* 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_FRACBITS - 16));
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return 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,n;
mad_fixed_t const *left_ch, *right_ch;
/* pcm->samplerate contains the sampling frequency */
nchannels = pcm->channels;
n=nsamples = pcm->length;
left_ch = pcm->samples[0];
right_ch = pcm->samples[1];
unsigned char Output[6912], *OutputPtr;
int fmt, wrote, speed, exact_rate, err, dir;
// printf("This is output\n");
OutputPtr = Output;
while (nsamples--)
{
signed int sample;
/* output sample(s) in 16-bit signed little-endian PCM */
sample = scale(*left_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
if (nchannels == 2)
{
sample = scale (*right_ch++);
*(OutputPtr++) = sample >> 0;
*(OutputPtr++) = sample >> 8;
}
}
OutputPtr = Output;
snd_pcm_writei (handle, OutputPtr, n);
OutputPtr = Output;
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)
{
struct buffer *buffer = data;
printf("this is mad_flow error\n");
fprintf(stderr, "decoding error 0x%04x (%s) at byte offset %u\n",
stream->error, mad_stream_errorstr(stream),
stream->this_frame - buffer->start);
/* 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(unsigned char const *start, unsigned long length)
{
struct buffer buffer;
struct mad_decoder decoder;
int result;
/* initialize our private message structure */
buffer.start = start;
buffer.length = length;
/* configure input, output, and error functions */
mad_decoder_init(&decoder, &buffer,
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;
}
二、WAV播放器：
本设计思路：先打开一个普通wav音频文件，从定义的文件头前面的44个字节中，取出文件头的定义消息，置于一个文件头的结构体中。然后打开alsa音频驱动，从文件头结构体取出采样精度，声道数，采样频率三个重要参数，利用alsa音频驱动的API设置好参数，最后打开wav文件，定位到数据区，把音频数据依次写到音频驱动中去，开始播放，当写入完成后，退出写入的循环。
注意：本设计需要alsa的libasound-dev的库，编译链接时需要连接 —lasound.
#include<stdio.h>
#include<stdlib.h>
#include <string.h>
#include <alsa/asoundlib.h>
struct WAV_HEADER
{
char rld[4]; //riff 标志符号
int rLen;
char wld[4]; //格式类型（wave）
char fld[4]; //"fmt"
int fLen; //sizeof(wave format matex)
short wFormatTag; //编码格式
short wChannels; //声道数
int nSamplesPersec ; //采样频率
int nAvgBitsPerSample;//WAVE文件采样大小
short wBlockAlign; //块对齐
short wBitsPerSample; //WAVE文件采样大小
char dld[4]; //”data“
int wSampleLength; //音频数据的大小
} wav_header;
int set_pcm_play(FILE *fp);
int main(int argc,char *argv[])
{
if(argc!=2)
{
printf("Usage:wav-player+wav file name\n");
exit(1);
}
int nread;
FILE *fp;
fp=fopen(argv[1],"rb");
if(fp==NULL)
{
perror("open file failed:\n");
exit(1);
}
nread=fread(&wav_header,1,sizeof(wav_header),fp);
printf("nread=%d\n",nread);
//printf("RIFF 标志%s\n",wav_header.rld);
printf("文件大小rLen：%d\n",wav_header.rLen);
//printf("wld=%s\n",wav_header.wld);
//printf("fld=%s\n",wav_header.fld);
// printf("fLen=%d\n",wav_header.fLen);
//printf("wFormatTag=%d\n",wav_header.wFormatTag);
printf("声道数：%d\n",wav_header.wChannels);
printf("采样频率：%d\n",wav_header.nSamplesPersec);
//printf("nAvgBitsPerSample=%d\n",wav_header.nAvgBitsPerSample);
//printf("wBlockAlign=%d\n",wav_header.wBlockAlign);
printf("采样的位数：%d\n",wav_header.wBitsPerSample);
// printf("data=%s\n",wav_header.dld);
printf("wSampleLength=%d\n",wav_header.wSampleLength);
set_pcm_play(fp);
return 0;
}
int set_pcm_play(FILE *fp)
{
int rc;
int ret;
int size;
snd_pcm_t* handle; //PCI设备句柄
snd_pcm_hw_params_t* params;//硬件信息和PCM流配置
unsigned int val;
int dir=0;
snd_pcm_uframes_t frames;
char *buffer;
int channels=wav_header.wChannels;
int frequency=wav_header.nSamplesPersec;
int bit=wav_header.wBitsPerSample;
int datablock=wav_header.wBlockAlign;
unsigned char ch[100]; //用来存储wav文件的头信息
rc=snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
if(rc<0)
{
perror("\nopen PCM device failed:");
exit(1);
}
snd_pcm_hw_params_alloca(¶ms); //分配params结构体
if(rc<0)
{
perror("\nsnd_pcm_hw_params_alloca:");
exit(1);
}
rc=snd_pcm_hw_params_any(handle, params);//初始化params
if(rc<0)
{
perror("\nsnd_pcm_hw_params_any:");
exit(1);
}
rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); //初始化访问权限
if(rc<0)
{
perror("\nsed_pcm_hw_set_access:");
exit(1);
}
//采样位数
switch(bit/8)
{
case 1:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_U8);
break ;
case 2:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
break ;
case 3:snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S24_LE);
break ;
}
rc=snd_pcm_hw_params_set_channels(handle, params, channels); //设置声道,1表示单声>道，2表示立体声
if(rc<0)
{
perror("\nsnd_pcm_hw_params_set_channels:");
exit(1);
}
val = frequency;
rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); //设置>频率
if(rc<0)
{
perror("\nsnd_pcm_hw_params_set_rate_near:");
exit(1);
}
rc = snd_pcm_hw_params(handle, params);
if(rc<0)
{
perror("\nsnd_pcm_hw_params: ");
exit(1);
}
rc=snd_pcm_hw_params_get_period_size(params, &frames, &dir); /*获取周期
长度*/
if(rc<0)
{
perror("\nsnd_pcm_hw_params_get_period_size:");
exit(1);
}
size = frames * datablock; /*4 代表数据快长度*/
buffer =(char*)malloc(size);
fseek(fp,58,SEEK_SET); //定位歌曲到数据区
while (1)
{
memset(buffer,0,sizeof(buffer));
ret = fread(buffer, 1, size, fp);
if(ret == 0)
{
printf("歌曲写入结束\n");
break;
}
else if (ret != size)
{
}
// 写音频数据到PCM设备
while(ret = snd_pcm_writei(handle, buffer, frames)<0)
{
usleep(2000);
if (ret == -EPIPE)
{
/* EPIPE means underrun */
fprintf(stderr, "underrun occurred\n");
//完成硬件参数设置，使设备准备好
snd_pcm_prepare(handle);
}
else if (ret < 0)
{
fprintf(stderr,
"error from writei: %s\n",
snd_strerror(ret));
}
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
return 0;
}
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