让我们做一个Mp3吧！
by panasonic.lin@163.com
前记：续wav音乐播放器的bug
1.//注册设备
static int __init init_sound(void)
{
...
/* And the output buffer. */
kfifo_init(&sound_fifo,&SoundWrite_buf, BUF_SIZE);
/* And the output info */
sound_output_active = 0;
spin_lock_init (&sound_out_lock);
/* Fill the sound_task structure, used for the bottom half handler. */
sound_task.routine = sound_do_task;
sound_task.data = NULL; /* unused */
/* And the output info */
sound_output_active = 0;
spin_lock_init (&sound_out_lock);
/* Fill the sound_task structure, used for the bottom half handler. */
sound_task.routine = sound_do_task;
sound_task.data = NULL; /* unused */
上次发布的wav音乐播放器的驱动有个bug，只能打开设备一次，第二次打开就没反应了，原因在于初始化的数据结构的函数放在
了init_sound只在注册模块的时候调用一次，所以在没有卸载模块之前，第二次打开设备使得程序在不确定状态。解决方法是把上面
这些初始化数据结构的函数放在dev_open函数就可以拉。
static int dev_open(struct inode *inode,struct file *filp)
修改好uda1314.c的源码后，简单的make就可以生成模块uda1314.o备用。
一点牢骚：
在这之前，我已经把libmad和madlld编译修改好，初步调试，可以正常解码mp3，文档和翻译我都记录下来了，准备共享给大家。开会回来，电脑黑屏，重新启动，opensuse提示找不到/bin/init,用opensuse的dvd修复系统时候才发现，硬盘分区表包括MBR都
坏了！修复也没用，ext4格式的分区让我无从下手，眼睁睁的看着数据丢失，一切重头来过。所谓同行业质保最长的三心硬盘又一次的
伤害了我幼小的心灵！但是讽刺的是这块arm的板子就是三心的cpu！
准备之前，参考uboot移植的文章，往板子上面移植个正常的uboot上去，然后再编译一个正常的uclinux核心，这些在44b0平台可以
参考我之前的博客文章。这里说些重要的，音频设备最重要的就是采样率，采样率跟时钟频率相关。我这里只实验了16khz的采样率，采样位数16bit，立体声。
//IIS master初始化,MCLK=60mHz,60,000,000/256/16=14,649Hz=fs?
//14,649*2/16=1831Hz=serial bit clock
//SCLK的频率=通(声)道数×采样频率×采样位数
/*LRCK的频率等于采样频率*/
rIISCON = 0x22; //使能Tx DMA，disable receive DMA,IISLRCK(in Tx and Rx generate),使能预分频器,iis停止
rIISMOD = 0x89; //主模式，发送方式，iis格式，16bit/通道，codeclk = 256fs,Serial bit clock = 32fs
rIISPSR = 0x77; //使能预分频器a/b enable division factor=16,so,MCLK/16=256fs->fs=65.536M/256*16=16khz
rIISFCON = 0xa00; //Tx DMA access mode，RX Normal access,enable TX FIFO
//使能IIS
rIISCON |= 0x1;
由iis的采样率反推cpu的主频大概是65.536M，约等于66Mhz。uboot和uclinux都应该设定一样的cpu主频!由于之前我板子上面的uboot是66M的主频，我在编译uclinux的时候选择的主频是60M，导致启动uclinux的时候start kernel后控制台全是乱码。
uboot的主频设定比较简单，在/cpu/s3c44b0/start.S可以设定pllcon寄存器，另外在/driver/serial/找到
44b0串口的代码，修改相应主频下的波特率分频因子，最后在/include/configs配置文件配置主频＝＝66M。
uclinux在make menuconfig的时候并没有主频可以填写，而早uclinux的其他文件提到配置cpu主频的很多，到底那个才是有效的？不得而知。
1是在/vendors/samsung/44b0/config.linux-2.4.x
# CONFIG_CPU_WITH_MCR_INSTRUCTION is not set
CONFIG_ARM_CLK=66000000
CONFIG_SERIAL_S3C44B0X=y
2是在/linux2.4.x/arch/armnommu/config.in，这个应该是配置默认值
if [ "$CONFIG_BOARD_MBA44" = "y" ]; then
define_string CONFIG_SPU_NAME "S3C44B0X"
define_bool CONFIG_CPU_S3C44B0X y
define_bool CONFIG_CPU_ARM710 y
define_bool CONFIG_CPU_32v4 y
define_bool CONFIG_CPU_32 y
define_bool CONFIG_CPU_26 n
define_bool CONFIG_NO_PGT_CACHE y
define_bool CONFIG_CPU_WITH_CACHE y
define_bool CONFIG_CPU_WITH_MCR_INSTRUCTION n
define_int CONFIG_ARM_CLK 66000000
define_bool CONFIG_SERIAL_S3C44B0X y
不过怎么样，最后的配置config后都生成/linux2.4.x/include/linux/autoconf.h头文件，它被编译到系统（很多文件都包含这个头文件），所以以这个为准。
#define CONFIG_CPU_WITH_CACHE 1
#undef CONFIG_CPU_WITH_MCR_INSTRUCTION
#define CONFIG_ARM_CLK (66000000)
#define CONFIG_SERIAL_S3C44B0X 1
好，环境都准备好了，最后最好弄个nfs的开发环境，修改文件比较方便。另外根文件系统的busybox应该有常用的程序如cat，insmod，lsmod，rmmod，echo，mount，和shell。最后用mknod在根文件系统的/dev目录下建立一个设备文件iis，主
编号iis_major＝14，次编号0，名字是iis，不管怎么样，目标一个：跟下面的函数保持一致。
result = register_chrdev(iis_major,DEV_IIS,&sound_dev_ops);
下一步是把编译好的模块uda1314.o拷贝到根文件系统，然后插入到核心中，
#insmod ./uda1314.o
可以看到如下提示
#lsmod 可以看到当前活动的模块
mp3播放的基础就是raw或者说wav格式的数据能直接播放的话，那基于这层的mp3解码，只是把mp3解码成pcm的bit流，然后往这个
device写就是了，从这个意义上说，mp3播放软件是应用软件，靠的就是底层核心态的pcm驱动！
我们可以往设备写pcm的wave文件检验驱动的正确性。写入之前确保dream这个wave文件是16k采样，16bit，立体声的格式，可以通过cooledit或者windows的录音机来转换格式。
#cat ./dream.wav > /dev/iis
驱动正常工作，开始mp3解码成pcm的工作。用libmad的有两种方法，一种是简单的利用高层的api写，另外一种是用底层的api，下面两种方法都会介绍到。
下载libmad和madlld。
第一步是编译libmad库：libmad.a和mad.h，后续的mp3应用程序需要链接到这个库。
./configure --help获得可用的配置选项。
panasonic@linux-p5tw:~/soundcard/libmad-0.15.1b> ./configure --help
`configure' configures MPEG Audio Decoder 0.15.1b to adapt to many kinds of systems.
Usage: ./configure [OPTION]... [VAR=VALUE]...
To assign environment variables (e.g., CC, CFLAGS...), specify them as
VAR=VALUE. See below for descriptions of some of the useful variables.
Defaults for the options are specified in brackets.
Configuration:
-h, --help display this help and exit
--help=short display options specific to this package
--help=recursive display the short help of all the included packages
-V, --version display version information and exit
-q, --quiet, --silent do not print `checking...' messages
--cache-file=FILE cache test results in FILE [disabled]
-C, --config-cache alias for `--cache-file=config.cache'
-n, --no-create do not create output files
--srcdir=DIR find the sources in DIR [configure dir or `..']
Installation directories:
注意在正式congfigure之前，由于arm－elf－gcc的特殊性，需要export一些变量，否则，真的很难解决这个问题:问题是./configure会报告说
arm-elf-gcc没有能力生成elf的执行文件。对了，arm-elf-gcc需要-elf2flt这个CFLAGS！
#export CC=arm-elf-gcc CFLAGS=-Wl,-elf2flt
panasonic@linux-p5tw:~/soundcard/libmad-0.15.1b> ./configure --enable-fpm=arm --host=arm-elf --disable-debugging --prefix=/home/panasonic/soundcard/libmad-0.15.1b/install --disable-shared --enable-static --enable-speed
configure: WARNING: If you wanted to set the --build type, don't use --host.
If a cross compiler is detected then cross compile mode will be used.
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for gawk... gawk
checking whether make sets $(MAKE)... yes
checking for arm-elf-strip... arm-elf-strip
make TARGET_ARCH=arm CROSS=arm-elf-
make install
最后生成mad.h和libmad.a静态库。
有了libmad.a库，下面写个简单的应用软件，这是用libmad高层的api写，比较简单，其实就是来自libmad自带的minimad.c
，test_music.c的基础就是前面介绍的wav播放器，所以
这个程序也可以播放wav音乐文件，判断命令行传递的argv参数就知道是mp3文件还是wav文件。
#include <sys/ioctl.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <string.h>
//add for mp3 play
# include <sys/stat.h>
# include <sys/mman.h>
# include <sys/types.h>
# include "./mad.h"
#define BUF_SIZE 16*1024
#define DEVICE_NAME "/dev/iis"
static int sfd; /*声音设备的描述符 */
static int decode(unsigned char const *, unsigned long);
void delay(long x)
{
unsigned long i;
for(i=0; i<x; i++);
}
struct buffer {
unsigned char const *start;
unsigned long length;
};
int main(int argc, char *argv[])
{
int i=0;
struct stat stat;
void *fdm;
FILE *file_fd; //文件
int fd;
long file_len; //文件长度
long loops;
int fread_ret;
unsigned char audio_buffer[BUF_SIZE];
unsigned char *file_name;
printf("This is a wav&mp3 play program.\n");
printf("Please add a file name and the audio rate!\nsuch as \"./test_music test.mp3\" \n\n\n");
delay(100000);
file_name = argv[1];
file_len = strlen(file_name);
if(file_len < 3)
{
printf("file is too short!\n");
exit(0);
}
printf("File name : %s \n",file_name);
delay(100000);
//if file format is mp3//
if(file_name[file_len-3]=='m' && file_name[file_len-2]=='p' && file_name[file_len-1]=='3')
{
printf("Play mp3!\n");
fd = open(file_name, O_RDONLY);
if ((sfd = open("/dev/iis", O_WRONLY)) < 0)
{
printf("can not open device!!!\n");
return 5;
}
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;
close(sfd);
}
else
{
/*打开音频设备，准备play*/
if ((sfd = open(DEVICE_NAME, O_WRONLY)) == -1)
{
printf("open error\n");
return -1;
}
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread_ret=fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
/*关闭设备和文件*/
fclose(file_fd);
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;
}
编译生成flt格式的程序，放置到根文件系统目录的bin下面。
panasonic@linux-p5tw:~/soundcard/uda1314> arm-elf-gcc -static -Wl,-elf2flt -o test_music test_music.c ./libmad.a
用cooledit转换一下mp3文件为16k的采样率，生成wav文件和相应的mp3文件，也一起放置在bin目录下。
测试结果表明，wav文件播放没有问题，mp3文件的大小如果大于1M的话，uclinux会报告内存不够，但是查看系统内存显示内存还有6M大呢!mmap内存映射是要申请内存的。
但是uclinux默认的可申请的内存在slab.c里面指定了，这可以在编译uclinux的时候改，但是也不是办法吧，用于学习可以，实际用途中，如果一个mp3文件很大很大，那
不是要申请好多内存。
不过这种方法的好处就是高速，一切都是内存读写，连续解码，实际试听也是，播放mp3很流畅。
播放wav文件时候，播放结束后，又遇到另外一个bug，就是歌曲播放结束了，但是程序还是拼命的往iis驱动write数据，结果导致耳机里面像火车一样的声音呜呜的响，我在驱动程序里面找了半天，后来在应用程序test_music的fread(audio_buffer, BUF_SIZE, 1, file_fd)函数打印的loops数字找到了答案，loops太大了！
ftell(file_fd)函数返回long类型的数值，loops应该是file_len除BUF_SIZE溢出了吧，最后定睛一看BUF_SIZE的宏，少了括号！导致除于16后乘1024了！
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread_ret=fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
fread_ret=1,loops=988845056,i=942
#define BUF_SIZE 16*1024
<1>.fread_ret=1,loops=943,i=941
<1>.fread_ret=1,loops=943,i=942
<1>.<4>enter release function..
.<1>.<1>@<4>exit IISClose..
解码mp3基本上就是这样了^_^为了解决上面的问题,我想更进一步，弄个更实用的。
想了解libmad多一点，最好使用libmad的底层api，这个竟然没有文档！用linux世界的话说就是：一切都在代码里面。
madlld就是libmad的文档工程，使用libmad底层的api写的解码mp3的程序，由stdin输入，解码后写入到stdout。
madlld只有三个c文件，核心是madlld的MpegAudioDecoder(FILE *InputFp)函数。
本来这三个c文件我是翻译好的了，但是。。。。。。哎呀，硬盘没了，大家将就点吧，啃点E文对大家都有好处。
长篇大论我就说了，我只附上我对madlld.c的修改patch补丁。
https://static.assets-stash.eet-china.com/album/old-resources/2010/4/24/c7b73f03-e863-4502-9cf6-9071fe329c82.zip
为了我的附件不丢失，我还是把代码贴上来吧，免得到时候附件没了，还可以从博文里面复制麻。
/* HTAB = 4 */
/****************************************************************************
* madlld.c -- A simple program decoding an mpeg audio stream to 16-bit *
* PCM from stdin to stdout. This program is just a simple sample *
* demonstrating how the low-level libmad API can be used. *
*--------------------------------------------------------------------------*
* (c) 2001--2004 Bertrand Petit *
* *
* Redistribution and use in source and binary forms, with or without *
* modification, are permitted provided that the following conditions *
* are met: *
* *
* 1. Redistributions of source code must retain the above copyright *
* notice, this list of conditions and the following disclaimer. *
* *
* 2. Redistributions in binary form must reproduce the above *
* copyright notice, this list of conditions and the following *
* disclaimer in the documentation and/or other materials provided *
* with the distribution. *
* *
* 3. Neither the name of the author nor the names of its contributors *
* may be used to endorse or promote products derived from this *
* software without specific prior written permission. *
* *
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' *
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED *
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A *
* PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR *
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, *
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT *
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF *
* USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND *
* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, *
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT *
* OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF *
* SUCH DAMAGE. *
* *
****************************************************************************/
/*
* $Name: v1_1 $
* $Date: 2004/02/22 03:27:05 $
* $Revision: 1.18 $
*/
/****************************************************************************
* Includes *
****************************************************************************/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <limits.h>
#include <errno.h>
#include <math.h> /* for pow() and log10() */
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
//add for mp3 play
# include <sys/stat.h>
# include <sys/mman.h>
# include <sys/types.h>
# include "./mad.h"
#include "bstdfile.h"
#define BUF_SIZE 16*1024
#define DEVICE_NAME "/dev/iis"
static int sfd; /*声音设备的描述符 */
/****************************************************************************
* Global variables. *
****************************************************************************/
/* Keeps a pointer to the program invocation name for the error
* messages.
*/
const char *ProgName;
/****************************************************************************
* Return an error string associated with a mad error code. *
****************************************************************************/
/* Mad version 0.14.2b introduced the mad_stream_errorstr() function.
* For previous library versions a replacement is provided below.
*/
#if (MAD_VERSION_MAJOR>=1) || \
((MAD_VERSION_MAJOR==0) && \
(((MAD_VERSION_MINOR==14) && \
(MAD_VERSION_PATCH>=2)) || \
(MAD_VERSION_MINOR>14)))
#define MadErrorString(x) mad_stream_errorstr(x)
#else
static const char *MadErrorString(const struct mad_stream *Stream)
{
switch(Stream->error)
{
/* Generic unrecoverable errors. */
case MAD_ERROR_BUFLEN:
return("input buffer too small (or EOF)");
case MAD_ERROR_BUFPTR:
return("invalid (null) buffer pointer");
case MAD_ERROR_NOMEM:
return("not enough memory");
/* Frame header related unrecoverable errors. */
case MAD_ERROR_LOSTSYNC:
return("lost synchronization");
case MAD_ERROR_BADLAYER:
return("reserved header layer value");
case MAD_ERROR_BADBITRATE:
return("forbidden bitrate value");
case MAD_ERROR_BADSAMPLERATE:
return("reserved sample frequency value");
case MAD_ERROR_BADEMPHASIS:
return("reserved emphasis value");
/* Recoverable errors */
case MAD_ERROR_BADCRC:
return("CRC check failed");
case MAD_ERROR_BADBITALLOC:
return("forbidden bit allocation value");
case MAD_ERROR_BADSCALEFACTOR:
return("bad scalefactor index");
case MAD_ERROR_BADFRAMELEN:
return("bad frame length");
case MAD_ERROR_BADBIGVALUES:
return("bad big_values count");
case MAD_ERROR_BADBLOCKTYPE:
return("reserved block_type");
case MAD_ERROR_BADSCFSI:
return("bad scalefactor selection info");
case MAD_ERROR_BADDATAPTR:
return("bad main_data_begin pointer");
case MAD_ERROR_BADPART3LEN:
return("bad audio data length");
case MAD_ERROR_BADHUFFTABLE:
return("bad Huffman table select");
case MAD_ERROR_BADHUFFDATA:
return("Huffman data overrun");
case MAD_ERROR_BADSTEREO:
return("incompatible block_type for JS");
/* Unknown error. This swich may be out of sync with libmad's
* defined error codes.
*/
default:
return("Unknown error code");
}
}
#endif
/****************************************************************************
* Converts a sample from mad's fixed point number format to a signed *
* short (16 bits). *
****************************************************************************/
static signed short MadFixedToSshort(mad_fixed_t Fixed)
{
/* A fixed point number is formed of the following bit pattern:
*
* SWWWFFFFFFFFFFFFFFFFFFFFFFFFFFFF
* MSB LSB
* S ==> Sign (0 is positive, 1 is negative)
* W ==> Whole part bits
* F ==> Fractional part bits
*
* This pattern contains MAD_F_FRACBITS fractional bits, one
* should alway use this macro when working on the bits of a fixed
* point number. It is not guaranteed to be constant over the
* different platforms supported by libmad.
*
* The signed short value is formed, after clipping, by the least
* significant whole part bit, followed by the 15 most significant
* fractional part bits. Warning: this is a quick and dirty way to
* compute the 16-bit number, madplay includes much better
* algorithms.
*/
/* Clipping */
if(Fixed>=MAD_F_ONE)
return(SHRT_MAX);
if(Fixed<=-MAD_F_ONE)
return(-SHRT_MAX);
/* Conversion. */
Fixed=Fixed>>(MAD_F_FRACBITS-15);
return((signed short)Fixed);
}
/****************************************************************************
* Print human readable informations about an audio MPEG frame. *
****************************************************************************/
static int PrintFrameInfo(FILE *fp, struct mad_header *Header)
{
const char *Layer,
*Mode,
*Emphasis;
/* Convert the layer number to it's printed representation. */
switch(Header->layer)
{
case MAD_LAYER_I:
Layer="I";
break;
case MAD_LAYER_II:
Layer="II";
break;
case MAD_LAYER_III:
Layer="III";
break;
default:
Layer="(unexpected layer value)";
break;
}
/* Convert the audio mode to it's printed representation. */
switch(Header->mode)
{
case MAD_MODE_SINGLE_CHANNEL:
Mode="single channel";
break;
case MAD_MODE_DUAL_CHANNEL:
Mode="dual channel";
break;
case MAD_MODE_JOINT_STEREO:
Mode="joint (MS/intensity) stereo";
break;
case MAD_MODE_STEREO:
Mode="normal LR stereo";
break;
default:
Mode="(unexpected mode value)";
break;
}
/* Convert the emphasis to it's printed representation. Note that
* the MAD_EMPHASIS_RESERVED enumeration value appread in libmad
* version 0.15.0b.
*/
switch(Header->emphasis)
{
case MAD_EMPHASIS_NONE:
Emphasis="no";
break;
case MAD_EMPHASIS_50_15_US:
Emphasis="50/15 us";
break;
case MAD_EMPHASIS_CCITT_J_17:
Emphasis="CCITT J.17";
break;
#if (MAD_VERSION_MAJOR>=1) || \
((MAD_VERSION_MAJOR==0) && (MAD_VERSION_MINOR>=15))
case MAD_EMPHASIS_RESERVED:
Emphasis="reserved(!)";
break;
#endif
default:
Emphasis="(unexpected emphasis value)";
break;
}
fprintf(fp,"%s: %lu kb/s audio mpeg layer %s stream %s crc, "
"%s with %s emphasis at %d Hz sample rate\n",
ProgName,Header->bitrate,Layer,
Header->flags&MAD_FLAG_PROTECTION?"with":"without",
Mode,Emphasis,Header->samplerate);
return(ferror(fp));
}
/****************************************************************************
* Main decoding loop. This is where mad is used. *
****************************************************************************/
#define INPUT_BUFFER_SIZE (5*8192)
#define OUTPUT_BUFFER_SIZE 8192 /* Must be an integer multiple of 4. */
static int MpegAudioDecoder(FILE *InputFp)
{
struct mad_stream Stream;
struct mad_frame Frame;
struct mad_synth Synth;
mad_timer_t Timer;
unsigned char InputBuffer[INPUT_BUFFER_SIZE+MAD_BUFFER_GUARD],
OutputBuffer[OUTPUT_BUFFER_SIZE],
*OutputPtr=OutputBuffer;
const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
int Status=0,
i;
unsigned long FrameCount=0;
bstdfile_t *BstdFile;
/* First the structures used by libmad must be initialized. */
mad_stream_init(&Stream);
mad_frame_init(&Frame);
mad_synth_init(&Synth);
mad_timer_reset(&Timer);
/* Decoding options can here be set in the options field of the
* Stream structure.
*/
/* {1} When decoding from a file we need to know when the end of
* the file is reached at the same time as the last bytes are read
* (see also the comment marked {3} bellow). Neither the standard
* C fread() function nor the posix read() system call provides
* this feature. We thus need to perform our reads through an
* interface having this feature, this is implemented here by the
* bstdfile.c module.
*/
BstdFile=NewBstdFile(InputFp);
if(BstdFile==NULL)
{
fprintf(stderr,"%s: can't create a new bstdfile_t (%s).\n",
ProgName,strerror(errno));
return(1);
}
/* This is the decoding loop. */
do
{
/* The input bucket must be filled if it becomes empty or if
* it's the first execution of the loop.
*/
if(Stream.buffer==NULL || Stream.error==MAD_ERROR_BUFLEN)
{
size_t ReadSize,
Remaining;
unsigned char *ReadStart;
/* {2} libmad may not consume all bytes of the input
* buffer. If the last frame in the buffer is not wholly
* contained by it, then that frame's start is pointed by
* the next_frame member of the Stream structure. This
* common situation occurs when mad_frame_decode() fails,
* sets the stream error code to MAD_ERROR_BUFLEN, and
* sets the next_frame pointer to a non NULL value. (See
* also the comment marked {4} bellow.)
*
* When this occurs, the remaining unused bytes must be
* put back at the beginning of the buffer and taken in
* account before refilling the buffer. This means that
* the input buffer must be large enough to hold a whole
* frame at the highest observable bit-rate (currently 448
* kb/s). XXX=XXX Is 2016 bytes the size of the largest
* frame? (448000*(1152/32000))/8
*/
if(Stream.next_frame!=NULL)
{
Remaining=Stream.bufend-Stream.next_frame;
memmove(InputBuffer,Stream.next_frame,Remaining);
ReadStart=InputBuffer+Remaining;
ReadSize=INPUT_BUFFER_SIZE-Remaining;
}
else
ReadSize=INPUT_BUFFER_SIZE,
ReadStart=InputBuffer,
Remaining=0;
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
ReadSize=BstdRead(ReadStart,1,ReadSize,BstdFile);
if(ReadSize<=0)
{
if(ferror(InputFp))
{
fprintf(stderr,"%s: read error on bitstream (%s)\n",
ProgName,strerror(errno));
Status=1;
}
if(feof(InputFp))
fprintf(stderr,"%s: end of input stream\n",ProgName);
break;
}
/* {3} When decoding the last frame of a file, it must be
* followed by MAD_BUFFER_GUARD zero bytes if one wants to
* decode that last frame. When the end of file is
* detected we append that quantity of bytes at the end of
* the available data. Note that the buffer can't overflow
* as the guard size was allocated but not used the the
* buffer managment code. (See also the comment marked
* {1}.)
*
* In a message to the mad-dev mailing list on May 29th,
* 2001, Rob leslie explains the guard zone as follows:
*
* "The reason for MAD_BUFFER_GUARD has to do with the
* way decoding is performed. In Layer III, Huffman
* decoding may inadvertently read a few bytes beyond
* the end of the buffer in the case of certain invalid
* input. This is not detected until after the fact. To
* prevent this from causing problems, and also to
* ensure the next frame's main_data_begin pointer is
* always accessible, MAD requires MAD_BUFFER_GUARD
* (currently 8) bytes to be present in the buffer past
* the end of the current frame in order to decode the
* frame."
*/
if(BstdFileEofP(BstdFile))
{
memset(ReadStart+ReadSize,0,MAD_BUFFER_GUARD);
ReadSize+=MAD_BUFFER_GUARD;
}
/* Pipe the new buffer content to libmad's stream decoder
* facility.
*/
mad_stream_buffer(&Stream,InputBuffer,ReadSize+Remaining);
Stream.error=0;
}
/* Decode the next mpeg frame. The streams is read from the
* buffer, its constituents are break down and stored the the
* Frame structure, ready for examination/alteration or PCM
* synthesis. Decoding options are carried in the Frame
* structure from the Stream structure.
*
* Error handling: mad_frame_decode() returns a non zero value
* when an error occurs. The error condition can be checked in
* the error member of the Stream structure. A mad error is
* recoverable or fatal, the error status is checked with the
* MAD_RECOVERABLE macro.
*
* {4} When a fatal error is encountered all decoding
* activities shall be stopped, except when a MAD_ERROR_BUFLEN
* is signaled. This condition means that the
* mad_frame_decode() function needs more input to achieve
* it's work. One shall refill the buffer and repeat the
* mad_frame_decode() call. Some bytes may be left unused at
* the end of the buffer if those bytes forms an incomplete
* frame. Before refilling, the remainign bytes must be moved
* to the begining of the buffer and used for input for the
* next mad_frame_decode() invocation. (See the comments marked
* {2} earlier for more details.)
*
* Recoverable errors are caused by malformed bit-streams, in
* this case one can call again mad_frame_decode() in order to
* skip the faulty part and re-sync to the next frame.
*/
if(mad_frame_decode(&Frame,&Stream))
{
if(MAD_RECOVERABLE(Stream.error))
{
fprintf(stderr,"%s: recoverable frame level error (%s)\n",
ProgName,MadErrorString(&Stream));
fflush(stderr);
continue;
}
else
if(Stream.error==MAD_ERROR_BUFLEN)
continue;
else
{
fprintf(stderr,"%s: unrecoverable frame level error (%s).\n",
ProgName,MadErrorString(&Stream));
Status=1;
break;
}
}
/* The characteristics of the stream's first frame is printed
* on stderr. The first frame is representative of the entire
* stream.
*/
if(FrameCount==0)
if(PrintFrameInfo(stderr,&Frame.header))
{
Status=1;
break;
}
/* Accounting. The computed frame duration is in the frame
* header structure. It is expressed as a fixed point number
* whole data type is mad_timer_t. It is different from the
* samples fixed point format and unlike it, it can't directly
* be added or substracted. The timer module provides several
* functions to operate on such numbers. Be careful there, as
* some functions of mad's timer module receive some of their
* mad_timer_t arguments by value!
*/
FrameCount++;
mad_timer_add(&Timer,Frame.header.duration);
/* Between the frame decoding and samples synthesis we can
* perform some operations on the audio data. We do this only
* if some processing was required. Detailed explanations are
* given in the ApplyFilter() function.
*/
//if(DoFilter)
// ApplyFilter(&Frame);
/* Once decoded the frame is synthesized to PCM samples. No errors
* are reported by mad_synth_frame();
*/
mad_synth_frame(&Synth,&Frame);
/* Synthesized samples must be converted from mad's fixed
* point number to the consumer format. Here we use unsigned
* 16 bit big endian integers on two channels. Integer samples
* are temporarily stored in a buffer that is flushed when
* full.
*/
for(i=0;i<Synth.pcm.length;i++)
{
signed short Sample;
/* Left channel */
Sample=MadFixedToSshort(Synth.pcm.samples[0]);
*(OutputPtr++)=Sample>>0;
*(OutputPtr++)=Sample>>8;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if(MAD_NCHANNELS(&Frame.header)==2)
Sample=MadFixedToSshort(Synth.pcm.samples[1]);
*(OutputPtr++)=Sample>>0;
*(OutputPtr++)=Sample>>8;
/* Flush the output buffer if it is full. */
if(OutputPtr==OutputBufferEnd)
{
//if(fwrite(OutputBuffer,1,OUTPUT_BUFFER_SIZE,OutputFp)!=OUTPUT_BUFFER_SIZE)
if(write(sfd, OutputBuffer, OUTPUT_BUFFER_SIZE)!=OUTPUT_BUFFER_SIZE)
{
fprintf(stderr,"%s: PCM write error (%s).\n",
ProgName,strerror(errno));
Status=2;
break;
}
OutputPtr=OutputBuffer;
}
}
}while(1);
/* The input file was completely read; the memory allocated by our
* reading module must be reclaimed.
*/
BstdFileDestroy(BstdFile);
/* Mad is no longer used, the structures that were initialized must
* now be cleared.
*/
mad_synth_finish(&Synth);
mad_frame_finish(&Frame);
mad_stream_finish(&Stream);
/* If the output buffer is not empty and no error occured during
* the last write, then flush it.
*/
if(OutputPtr!=OutputBuffer && Status!=2)
{
size_t BufferSize=OutputPtr-OutputBuffer;
//if(fwrite(OutputBuffer,1,BufferSize,OutputFp)!=BufferSize)
if(write(sfd,OutputBuffer,BufferSize)!=BufferSize)
{
fprintf(stderr,"%s: PCM write error (%s).\n",
ProgName,strerror(errno));
Status=2;
}
}
/* Accounting report if no error occured. */
if(!Status)
{
char Buffer[80];
/* The duration timer is converted to a human readable string
* with the versatile, but still constrained mad_timer_string()
* function, in a fashion not unlike strftime(). The main
* difference is that the timer is broken into several
* values according some of it's arguments. The units and
* fracunits arguments specify the intended conversion to be
* executed.
*
* The conversion unit (MAD_UNIT_MINUTES in our example) also
* specify the order and kind of conversion specifications
* that can be used in the format string.
*
* It is best to examine mad's timer.c source-code for details
* of the available units, fraction of units, their meanings,
* the format arguments, etc.
*/
mad_timer_string(Timer,Buffer,"%lu:%02lu.%03u",
MAD_UNITS_MINUTES,MAD_UNITS_MILLISECONDS,0);
fprintf(stderr,"%s: %lu frames decoded (%s).\n",
ProgName,FrameCount,Buffer);
}
/* That's the end of the world (in the H. G. Wells way). */
return(Status);
}
/****************************************************************************
* Program entry point. *
****************************************************************************/
int main(int argc, char *argv[])
{
int Status;
int i=0;
FILE *file_fd; //文件
long file_len; //文件长度
long loops;
unsigned char audio_buffer[BUF_SIZE];
unsigned char *file_name;
printf("This is a wav&mp3 play program.\n");
printf("Please add a file name and the audio rate!\nsuch as \"./test_music test.mp3\" \n\n\n");
//delay(100000);
ProgName=argv[0];
file_name = argv[1];
file_len = strlen(file_name);
if(file_len < 3)
{
printf("file is too short!\n");
exit(0);
}
printf("File name : %s \n",file_name);
//delay(100000);
//if file format is mp3//
if(file_name[file_len-3]=='m' && file_name[file_len-2]=='p' && file_name[file_len-1]=='3')
{
printf("Play mp3!\n");
// fd = open(file_name, O_RDONLY);
file_fd = fopen(file_name, "r");
if ((sfd = open("/dev/iis", O_WRONLY)) < 0)
{
printf("can not open device!!!\n");
return 5;
}
/* Decode stdin to stdout. */
Status=MpegAudioDecoder(file_fd);
if(Status)
fprintf(stderr,"%s: an error occured during decoding.\n",ProgName);
/* All done. */
fclose(file_fd);
close(sfd);
return(Status);
}
else
{
/*打开音频设备，准备play*/
if ((sfd = open(DEVICE_NAME, O_WRONLY)) == -1)
{
printf("open error\n");
return -1;
}
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
/*关闭设备和文件*/
fclose(file_fd);
close(sfd);
}
return 0;
}
/*
* Local Variables:
* tab-width: 4
* End:
*/
/****************************************************************************
* End of file madllc.c *
****************************************************************************/
好，准备好这些文件后，开始动手，编译连接libmad，生成应用程序madlld。
arm-elf-gcc -static -Wl,-elf2flt -o madlld madlld.c bstdfile.c ./libmad.a
同样放到根文件系统bin下面，开始测试：
哈哈，总算成功拉！
这是源码code大礼包！
https://static.assets-stash.eet-china.com/album/old-resources/2010/4/24/f2c44bca-d69a-4e25-9a8b-2dc226c37f06.zip
来张照片合影下
这是手稿
by panasonic.lin@163.com
前记：续wav音乐播放器的bug
1.//注册设备
static int __init init_sound(void)
{
...
/* And the output buffer. */
kfifo_init(&sound_fifo,&SoundWrite_buf, BUF_SIZE);
/* And the output info */
sound_output_active = 0;
spin_lock_init (&sound_out_lock);
/* Fill the sound_task structure, used for the bottom half handler. */
sound_task.routine = sound_do_task;
sound_task.data = NULL; /* unused */
/* And the output info */
sound_output_active = 0;
spin_lock_init (&sound_out_lock);
/* Fill the sound_task structure, used for the bottom half handler. */
sound_task.routine = sound_do_task;
sound_task.data = NULL; /* unused */
上次发布的wav音乐播放器的驱动有个bug，只能打开设备一次，第二次打开就没反应了，原因在于初始化的数据结构的函数放在
了init_sound只在注册模块的时候调用一次，所以在没有卸载模块之前，第二次打开设备使得程序在不确定状态。解决方法是把上面
这些初始化数据结构的函数放在dev_open函数就可以拉。
static int dev_open(struct inode *inode,struct file *filp)
修改好uda1314.c的源码后，简单的make就可以生成模块uda1314.o备用。
一点牢骚：
在这之前，我已经把libmad和madlld编译修改好，初步调试，可以正常解码mp3，文档和翻译我都记录下来了，准备共享给大家。开会回来，电脑黑屏，重新启动，opensuse提示找不到/bin/init,用opensuse的dvd修复系统时候才发现，硬盘分区表包括MBR都
坏了！修复也没用，ext4格式的分区让我无从下手，眼睁睁的看着数据丢失，一切重头来过。所谓同行业质保最长的三心硬盘又一次的
伤害了我幼小的心灵！但是讽刺的是这块arm的板子就是三心的cpu！
准备之前，参考uboot移植的文章，往板子上面移植个正常的uboot上去，然后再编译一个正常的uclinux核心，这些在44b0平台可以
参考我之前的博客文章。这里说些重要的，音频设备最重要的就是采样率，采样率跟时钟频率相关。我这里只实验了16khz的采样率，采样位数16bit，立体声。
//IIS master初始化,MCLK=60mHz,60,000,000/256/16=14,649Hz=fs?
//14,649*2/16=1831Hz=serial bit clock
//SCLK的频率=通(声)道数×采样频率×采样位数
/*LRCK的频率等于采样频率*/
rIISCON = 0x22; //使能Tx DMA，disable receive DMA,IISLRCK(in Tx and Rx generate),使能预分频器,iis停止
rIISMOD = 0x89; //主模式，发送方式，iis格式，16bit/通道，codeclk = 256fs,Serial bit clock = 32fs
rIISPSR = 0x77; //使能预分频器a/b enable division factor=16,so,MCLK/16=256fs->fs=65.536M/256*16=16khz
rIISFCON = 0xa00; //Tx DMA access mode，RX Normal access,enable TX FIFO
//使能IIS
rIISCON |= 0x1;
由iis的采样率反推cpu的主频大概是65.536M，约等于66Mhz。uboot和uclinux都应该设定一样的cpu主频!由于之前我板子上面的uboot是66M的主频，我在编译uclinux的时候选择的主频是60M，导致启动uclinux的时候start kernel后控制台全是乱码。
uboot的主频设定比较简单，在/cpu/s3c44b0/start.S可以设定pllcon寄存器，另外在/driver/serial/找到
44b0串口的代码，修改相应主频下的波特率分频因子，最后在/include/configs配置文件配置主频＝＝66M。
uclinux在make menuconfig的时候并没有主频可以填写，而早uclinux的其他文件提到配置cpu主频的很多，到底那个才是有效的？不得而知。
1是在/vendors/samsung/44b0/config.linux-2.4.x
# CONFIG_CPU_WITH_MCR_INSTRUCTION is not set
CONFIG_ARM_CLK=66000000
CONFIG_SERIAL_S3C44B0X=y
2是在/linux2.4.x/arch/armnommu/config.in，这个应该是配置默认值
if [ "$CONFIG_BOARD_MBA44" = "y" ]; then
define_string CONFIG_SPU_NAME "S3C44B0X"
define_bool CONFIG_CPU_S3C44B0X y
define_bool CONFIG_CPU_ARM710 y
define_bool CONFIG_CPU_32v4 y
define_bool CONFIG_CPU_32 y
define_bool CONFIG_CPU_26 n
define_bool CONFIG_NO_PGT_CACHE y
define_bool CONFIG_CPU_WITH_CACHE y
define_bool CONFIG_CPU_WITH_MCR_INSTRUCTION n
define_int CONFIG_ARM_CLK 66000000
define_bool CONFIG_SERIAL_S3C44B0X y
不过怎么样，最后的配置config后都生成/linux2.4.x/include/linux/autoconf.h头文件，它被编译到系统（很多文件都包含这个头文件），所以以这个为准。
#define CONFIG_CPU_WITH_CACHE 1
#undef CONFIG_CPU_WITH_MCR_INSTRUCTION
#define CONFIG_ARM_CLK (66000000)
#define CONFIG_SERIAL_S3C44B0X 1
好，环境都准备好了，最后最好弄个nfs的开发环境，修改文件比较方便。另外根文件系统的busybox应该有常用的程序如cat，insmod，lsmod，rmmod，echo，mount，和shell。最后用mknod在根文件系统的/dev目录下建立一个设备文件iis，主
编号iis_major＝14，次编号0，名字是iis，不管怎么样，目标一个：跟下面的函数保持一致。
result = register_chrdev(iis_major,DEV_IIS,&sound_dev_ops);
下一步是把编译好的模块uda1314.o拷贝到根文件系统，然后插入到核心中，
#insmod ./uda1314.o
可以看到如下提示
#lsmod 可以看到当前活动的模块
mp3播放的基础就是raw或者说wav格式的数据能直接播放的话，那基于这层的mp3解码，只是把mp3解码成pcm的bit流，然后往这个
device写就是了，从这个意义上说，mp3播放软件是应用软件，靠的就是底层核心态的pcm驱动！
我们可以往设备写pcm的wave文件检验驱动的正确性。写入之前确保dream这个wave文件是16k采样，16bit，立体声的格式，可以通过cooledit或者windows的录音机来转换格式。
#cat ./dream.wav > /dev/iis
驱动正常工作，开始mp3解码成pcm的工作。用libmad的有两种方法，一种是简单的利用高层的api写，另外一种是用底层的api，下面两种方法都会介绍到。
下载libmad和madlld。
第一步是编译libmad库：libmad.a和mad.h，后续的mp3应用程序需要链接到这个库。
./configure --help获得可用的配置选项。
panasonic@linux-p5tw:~/soundcard/libmad-0.15.1b> ./configure --help
`configure' configures MPEG Audio Decoder 0.15.1b to adapt to many kinds of systems.
Usage: ./configure [OPTION]... [VAR=VALUE]...
To assign environment variables (e.g., CC, CFLAGS...), specify them as
VAR=VALUE. See below for descriptions of some of the useful variables.
Defaults for the options are specified in brackets.
Configuration:
-h, --help display this help and exit
--help=short display options specific to this package
--help=recursive display the short help of all the included packages
-V, --version display version information and exit
-q, --quiet, --silent do not print `checking...' messages
--cache-file=FILE cache test results in FILE [disabled]
-C, --config-cache alias for `--cache-file=config.cache'
-n, --no-create do not create output files
--srcdir=DIR find the sources in DIR [configure dir or `..']
Installation directories:
注意在正式congfigure之前，由于arm－elf－gcc的特殊性，需要export一些变量，否则，真的很难解决这个问题:问题是./configure会报告说
arm-elf-gcc没有能力生成elf的执行文件。对了，arm-elf-gcc需要-elf2flt这个CFLAGS！
#export CC=arm-elf-gcc CFLAGS=-Wl,-elf2flt
panasonic@linux-p5tw:~/soundcard/libmad-0.15.1b> ./configure --enable-fpm=arm --host=arm-elf --disable-debugging --prefix=/home/panasonic/soundcard/libmad-0.15.1b/install --disable-shared --enable-static --enable-speed
configure: WARNING: If you wanted to set the --build type, don't use --host.
If a cross compiler is detected then cross compile mode will be used.
checking for a BSD-compatible install... /usr/bin/install -c
checking whether build environment is sane... yes
checking for gawk... gawk
checking whether make sets $(MAKE)... yes
checking for arm-elf-strip... arm-elf-strip
make TARGET_ARCH=arm CROSS=arm-elf-
make install
最后生成mad.h和libmad.a静态库。
有了libmad.a库，下面写个简单的应用软件，这是用libmad高层的api写，比较简单，其实就是来自libmad自带的minimad.c
，test_music.c的基础就是前面介绍的wav播放器，所以
这个程序也可以播放wav音乐文件，判断命令行传递的argv参数就知道是mp3文件还是wav文件。
#include <sys/ioctl.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#include <string.h>
//add for mp3 play
# include <sys/stat.h>
# include <sys/mman.h>
# include <sys/types.h>
# include "./mad.h"
#define BUF_SIZE 16*1024
#define DEVICE_NAME "/dev/iis"
static int sfd; /*声音设备的描述符 */
static int decode(unsigned char const *, unsigned long);
void delay(long x)
{
unsigned long i;
for(i=0; i<x; i++);
}
struct buffer {
unsigned char const *start;
unsigned long length;
};
int main(int argc, char *argv[])
{
int i=0;
struct stat stat;
void *fdm;
FILE *file_fd; //文件
int fd;
long file_len; //文件长度
long loops;
int fread_ret;
unsigned char audio_buffer[BUF_SIZE];
unsigned char *file_name;
printf("This is a wav&mp3 play program.\n");
printf("Please add a file name and the audio rate!\nsuch as \"./test_music test.mp3\" \n\n\n");
delay(100000);
file_name = argv[1];
file_len = strlen(file_name);
if(file_len < 3)
{
printf("file is too short!\n");
exit(0);
}
printf("File name : %s \n",file_name);
delay(100000);
//if file format is mp3//
if(file_name[file_len-3]=='m' && file_name[file_len-2]=='p' && file_name[file_len-1]=='3')
{
printf("Play mp3!\n");
fd = open(file_name, O_RDONLY);
if ((sfd = open("/dev/iis", O_WRONLY)) < 0)
{
printf("can not open device!!!\n");
return 5;
}
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;
close(sfd);
}
else
{
/*打开音频设备，准备play*/
if ((sfd = open(DEVICE_NAME, O_WRONLY)) == -1)
{
printf("open error\n");
return -1;
}
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread_ret=fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
/*关闭设备和文件*/
fclose(file_fd);
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;
}
编译生成flt格式的程序，放置到根文件系统目录的bin下面。
panasonic@linux-p5tw:~/soundcard/uda1314> arm-elf-gcc -static -Wl,-elf2flt -o test_music test_music.c ./libmad.a
用cooledit转换一下mp3文件为16k的采样率，生成wav文件和相应的mp3文件，也一起放置在bin目录下。
测试结果表明，wav文件播放没有问题，mp3文件的大小如果大于1M的话，uclinux会报告内存不够，但是查看系统内存显示内存还有6M大呢!mmap内存映射是要申请内存的。
但是uclinux默认的可申请的内存在slab.c里面指定了，这可以在编译uclinux的时候改，但是也不是办法吧，用于学习可以，实际用途中，如果一个mp3文件很大很大，那
不是要申请好多内存。
不过这种方法的好处就是高速，一切都是内存读写，连续解码，实际试听也是，播放mp3很流畅。
播放wav文件时候，播放结束后，又遇到另外一个bug，就是歌曲播放结束了，但是程序还是拼命的往iis驱动write数据，结果导致耳机里面像火车一样的声音呜呜的响，我在驱动程序里面找了半天，后来在应用程序test_music的fread(audio_buffer, BUF_SIZE, 1, file_fd)函数打印的loops数字找到了答案，loops太大了！
ftell(file_fd)函数返回long类型的数值，loops应该是file_len除BUF_SIZE溢出了吧，最后定睛一看BUF_SIZE的宏，少了括号！导致除于16后乘1024了！
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread_ret=fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
fread_ret=1,loops=988845056,i=942
#define BUF_SIZE 16*1024
<1>.fread_ret=1,loops=943,i=941
<1>.fread_ret=1,loops=943,i=942
<1>.<4>enter release function..
.<1>.<1>@<4>exit IISClose..
解码mp3基本上就是这样了^_^为了解决上面的问题,我想更进一步，弄个更实用的。
想了解libmad多一点，最好使用libmad的底层api，这个竟然没有文档！用linux世界的话说就是：一切都在代码里面。
madlld就是libmad的文档工程，使用libmad底层的api写的解码mp3的程序，由stdin输入，解码后写入到stdout。
madlld只有三个c文件，核心是madlld的MpegAudioDecoder(FILE *InputFp)函数。
本来这三个c文件我是翻译好的了，但是。。。。。。哎呀，硬盘没了，大家将就点吧，啃点E文对大家都有好处。
长篇大论我就说了，我只附上我对madlld.c的修改patch补丁。
https://static.assets-stash.eet-china.com/album/old-resources/2010/4/24/c7b73f03-e863-4502-9cf6-9071fe329c82.zip
为了我的附件不丢失，我还是把代码贴上来吧，免得到时候附件没了，还可以从博文里面复制麻。
/* HTAB = 4 */
/****************************************************************************
* madlld.c -- A simple program decoding an mpeg audio stream to 16-bit *
* PCM from stdin to stdout. This program is just a simple sample *
* demonstrating how the low-level libmad API can be used. *
*--------------------------------------------------------------------------*
* (c) 2001--2004 Bertrand Petit *
* *
* Redistribution and use in source and binary forms, with or without *
* modification, are permitted provided that the following conditions *
* are met: *
* *
* 1. Redistributions of source code must retain the above copyright *
* notice, this list of conditions and the following disclaimer. *
* *
* 2. Redistributions in binary form must reproduce the above *
* copyright notice, this list of conditions and the following *
* disclaimer in the documentation and/or other materials provided *
* with the distribution. *
* *
* 3. Neither the name of the author nor the names of its contributors *
* may be used to endorse or promote products derived from this *
* software without specific prior written permission. *
* *
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' *
* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED *
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A *
* PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR *
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, *
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT *
* LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF *
* USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND *
* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, *
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT *
* OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF *
* SUCH DAMAGE. *
* *
****************************************************************************/
/*
* $Name: v1_1 $
* $Date: 2004/02/22 03:27:05 $
* $Revision: 1.18 $
*/
/****************************************************************************
* Includes *
****************************************************************************/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <limits.h>
#include <errno.h>
#include <math.h> /* for pow() and log10() */
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
//add for mp3 play
# include <sys/stat.h>
# include <sys/mman.h>
# include <sys/types.h>
# include "./mad.h"
#include "bstdfile.h"
#define BUF_SIZE 16*1024
#define DEVICE_NAME "/dev/iis"
static int sfd; /*声音设备的描述符 */
/****************************************************************************
* Global variables. *
****************************************************************************/
/* Keeps a pointer to the program invocation name for the error
* messages.
*/
const char *ProgName;
/****************************************************************************
* Return an error string associated with a mad error code. *
****************************************************************************/
/* Mad version 0.14.2b introduced the mad_stream_errorstr() function.
* For previous library versions a replacement is provided below.
*/
#if (MAD_VERSION_MAJOR>=1) || \
((MAD_VERSION_MAJOR==0) && \
(((MAD_VERSION_MINOR==14) && \
(MAD_VERSION_PATCH>=2)) || \
(MAD_VERSION_MINOR>14)))
#define MadErrorString(x) mad_stream_errorstr(x)
#else
static const char *MadErrorString(const struct mad_stream *Stream)
{
switch(Stream->error)
{
/* Generic unrecoverable errors. */
case MAD_ERROR_BUFLEN:
return("input buffer too small (or EOF)");
case MAD_ERROR_BUFPTR:
return("invalid (null) buffer pointer");
case MAD_ERROR_NOMEM:
return("not enough memory");
/* Frame header related unrecoverable errors. */
case MAD_ERROR_LOSTSYNC:
return("lost synchronization");
case MAD_ERROR_BADLAYER:
return("reserved header layer value");
case MAD_ERROR_BADBITRATE:
return("forbidden bitrate value");
case MAD_ERROR_BADSAMPLERATE:
return("reserved sample frequency value");
case MAD_ERROR_BADEMPHASIS:
return("reserved emphasis value");
/* Recoverable errors */
case MAD_ERROR_BADCRC:
return("CRC check failed");
case MAD_ERROR_BADBITALLOC:
return("forbidden bit allocation value");
case MAD_ERROR_BADSCALEFACTOR:
return("bad scalefactor index");
case MAD_ERROR_BADFRAMELEN:
return("bad frame length");
case MAD_ERROR_BADBIGVALUES:
return("bad big_values count");
case MAD_ERROR_BADBLOCKTYPE:
return("reserved block_type");
case MAD_ERROR_BADSCFSI:
return("bad scalefactor selection info");
case MAD_ERROR_BADDATAPTR:
return("bad main_data_begin pointer");
case MAD_ERROR_BADPART3LEN:
return("bad audio data length");
case MAD_ERROR_BADHUFFTABLE:
return("bad Huffman table select");
case MAD_ERROR_BADHUFFDATA:
return("Huffman data overrun");
case MAD_ERROR_BADSTEREO:
return("incompatible block_type for JS");
/* Unknown error. This swich may be out of sync with libmad's
* defined error codes.
*/
default:
return("Unknown error code");
}
}
#endif
/****************************************************************************
* Converts a sample from mad's fixed point number format to a signed *
* short (16 bits). *
****************************************************************************/
static signed short MadFixedToSshort(mad_fixed_t Fixed)
{
/* A fixed point number is formed of the following bit pattern:
*
* SWWWFFFFFFFFFFFFFFFFFFFFFFFFFFFF
* MSB LSB
* S ==> Sign (0 is positive, 1 is negative)
* W ==> Whole part bits
* F ==> Fractional part bits
*
* This pattern contains MAD_F_FRACBITS fractional bits, one
* should alway use this macro when working on the bits of a fixed
* point number. It is not guaranteed to be constant over the
* different platforms supported by libmad.
*
* The signed short value is formed, after clipping, by the least
* significant whole part bit, followed by the 15 most significant
* fractional part bits. Warning: this is a quick and dirty way to
* compute the 16-bit number, madplay includes much better
* algorithms.
*/
/* Clipping */
if(Fixed>=MAD_F_ONE)
return(SHRT_MAX);
if(Fixed<=-MAD_F_ONE)
return(-SHRT_MAX);
/* Conversion. */
Fixed=Fixed>>(MAD_F_FRACBITS-15);
return((signed short)Fixed);
}
/****************************************************************************
* Print human readable informations about an audio MPEG frame. *
****************************************************************************/
static int PrintFrameInfo(FILE *fp, struct mad_header *Header)
{
const char *Layer,
*Mode,
*Emphasis;
/* Convert the layer number to it's printed representation. */
switch(Header->layer)
{
case MAD_LAYER_I:
Layer="I";
break;
case MAD_LAYER_II:
Layer="II";
break;
case MAD_LAYER_III:
Layer="III";
break;
default:
Layer="(unexpected layer value)";
break;
}
/* Convert the audio mode to it's printed representation. */
switch(Header->mode)
{
case MAD_MODE_SINGLE_CHANNEL:
Mode="single channel";
break;
case MAD_MODE_DUAL_CHANNEL:
Mode="dual channel";
break;
case MAD_MODE_JOINT_STEREO:
Mode="joint (MS/intensity) stereo";
break;
case MAD_MODE_STEREO:
Mode="normal LR stereo";
break;
default:
Mode="(unexpected mode value)";
break;
}
/* Convert the emphasis to it's printed representation. Note that
* the MAD_EMPHASIS_RESERVED enumeration value appread in libmad
* version 0.15.0b.
*/
switch(Header->emphasis)
{
case MAD_EMPHASIS_NONE:
Emphasis="no";
break;
case MAD_EMPHASIS_50_15_US:
Emphasis="50/15 us";
break;
case MAD_EMPHASIS_CCITT_J_17:
Emphasis="CCITT J.17";
break;
#if (MAD_VERSION_MAJOR>=1) || \
((MAD_VERSION_MAJOR==0) && (MAD_VERSION_MINOR>=15))
case MAD_EMPHASIS_RESERVED:
Emphasis="reserved(!)";
break;
#endif
default:
Emphasis="(unexpected emphasis value)";
break;
}
fprintf(fp,"%s: %lu kb/s audio mpeg layer %s stream %s crc, "
"%s with %s emphasis at %d Hz sample rate\n",
ProgName,Header->bitrate,Layer,
Header->flags&MAD_FLAG_PROTECTION?"with":"without",
Mode,Emphasis,Header->samplerate);
return(ferror(fp));
}
/****************************************************************************
* Main decoding loop. This is where mad is used. *
****************************************************************************/
#define INPUT_BUFFER_SIZE (5*8192)
#define OUTPUT_BUFFER_SIZE 8192 /* Must be an integer multiple of 4. */
static int MpegAudioDecoder(FILE *InputFp)
{
struct mad_stream Stream;
struct mad_frame Frame;
struct mad_synth Synth;
mad_timer_t Timer;
unsigned char InputBuffer[INPUT_BUFFER_SIZE+MAD_BUFFER_GUARD],
OutputBuffer[OUTPUT_BUFFER_SIZE],
*OutputPtr=OutputBuffer;
const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
int Status=0,
i;
unsigned long FrameCount=0;
bstdfile_t *BstdFile;
/* First the structures used by libmad must be initialized. */
mad_stream_init(&Stream);
mad_frame_init(&Frame);
mad_synth_init(&Synth);
mad_timer_reset(&Timer);
/* Decoding options can here be set in the options field of the
* Stream structure.
*/
/* {1} When decoding from a file we need to know when the end of
* the file is reached at the same time as the last bytes are read
* (see also the comment marked {3} bellow). Neither the standard
* C fread() function nor the posix read() system call provides
* this feature. We thus need to perform our reads through an
* interface having this feature, this is implemented here by the
* bstdfile.c module.
*/
BstdFile=NewBstdFile(InputFp);
if(BstdFile==NULL)
{
fprintf(stderr,"%s: can't create a new bstdfile_t (%s).\n",
ProgName,strerror(errno));
return(1);
}
/* This is the decoding loop. */
do
{
/* The input bucket must be filled if it becomes empty or if
* it's the first execution of the loop.
*/
if(Stream.buffer==NULL || Stream.error==MAD_ERROR_BUFLEN)
{
size_t ReadSize,
Remaining;
unsigned char *ReadStart;
/* {2} libmad may not consume all bytes of the input
* buffer. If the last frame in the buffer is not wholly
* contained by it, then that frame's start is pointed by
* the next_frame member of the Stream structure. This
* common situation occurs when mad_frame_decode() fails,
* sets the stream error code to MAD_ERROR_BUFLEN, and
* sets the next_frame pointer to a non NULL value. (See
* also the comment marked {4} bellow.)
*
* When this occurs, the remaining unused bytes must be
* put back at the beginning of the buffer and taken in
* account before refilling the buffer. This means that
* the input buffer must be large enough to hold a whole
* frame at the highest observable bit-rate (currently 448
* kb/s). XXX=XXX Is 2016 bytes the size of the largest
* frame? (448000*(1152/32000))/8
*/
if(Stream.next_frame!=NULL)
{
Remaining=Stream.bufend-Stream.next_frame;
memmove(InputBuffer,Stream.next_frame,Remaining);
ReadStart=InputBuffer+Remaining;
ReadSize=INPUT_BUFFER_SIZE-Remaining;
}
else
ReadSize=INPUT_BUFFER_SIZE,
ReadStart=InputBuffer,
Remaining=0;
/* Fill-in the buffer. If an error occurs print a message
* and leave the decoding loop. If the end of stream is
* reached we also leave the loop but the return status is
* left untouched.
*/
ReadSize=BstdRead(ReadStart,1,ReadSize,BstdFile);
if(ReadSize<=0)
{
if(ferror(InputFp))
{
fprintf(stderr,"%s: read error on bitstream (%s)\n",
ProgName,strerror(errno));
Status=1;
}
if(feof(InputFp))
fprintf(stderr,"%s: end of input stream\n",ProgName);
break;
}
/* {3} When decoding the last frame of a file, it must be
* followed by MAD_BUFFER_GUARD zero bytes if one wants to
* decode that last frame. When the end of file is
* detected we append that quantity of bytes at the end of
* the available data. Note that the buffer can't overflow
* as the guard size was allocated but not used the the
* buffer managment code. (See also the comment marked
* {1}.)
*
* In a message to the mad-dev mailing list on May 29th,
* 2001, Rob leslie explains the guard zone as follows:
*
* "The reason for MAD_BUFFER_GUARD has to do with the
* way decoding is performed. In Layer III, Huffman
* decoding may inadvertently read a few bytes beyond
* the end of the buffer in the case of certain invalid
* input. This is not detected until after the fact. To
* prevent this from causing problems, and also to
* ensure the next frame's main_data_begin pointer is
* always accessible, MAD requires MAD_BUFFER_GUARD
* (currently 8) bytes to be present in the buffer past
* the end of the current frame in order to decode the
* frame."
*/
if(BstdFileEofP(BstdFile))
{
memset(ReadStart+ReadSize,0,MAD_BUFFER_GUARD);
ReadSize+=MAD_BUFFER_GUARD;
}
/* Pipe the new buffer content to libmad's stream decoder
* facility.
*/
mad_stream_buffer(&Stream,InputBuffer,ReadSize+Remaining);
Stream.error=0;
}
/* Decode the next mpeg frame. The streams is read from the
* buffer, its constituents are break down and stored the the
* Frame structure, ready for examination/alteration or PCM
* synthesis. Decoding options are carried in the Frame
* structure from the Stream structure.
*
* Error handling: mad_frame_decode() returns a non zero value
* when an error occurs. The error condition can be checked in
* the error member of the Stream structure. A mad error is
* recoverable or fatal, the error status is checked with the
* MAD_RECOVERABLE macro.
*
* {4} When a fatal error is encountered all decoding
* activities shall be stopped, except when a MAD_ERROR_BUFLEN
* is signaled. This condition means that the
* mad_frame_decode() function needs more input to achieve
* it's work. One shall refill the buffer and repeat the
* mad_frame_decode() call. Some bytes may be left unused at
* the end of the buffer if those bytes forms an incomplete
* frame. Before refilling, the remainign bytes must be moved
* to the begining of the buffer and used for input for the
* next mad_frame_decode() invocation. (See the comments marked
* {2} earlier for more details.)
*
* Recoverable errors are caused by malformed bit-streams, in
* this case one can call again mad_frame_decode() in order to
* skip the faulty part and re-sync to the next frame.
*/
if(mad_frame_decode(&Frame,&Stream))
{
if(MAD_RECOVERABLE(Stream.error))
{
fprintf(stderr,"%s: recoverable frame level error (%s)\n",
ProgName,MadErrorString(&Stream));
fflush(stderr);
continue;
}
else
if(Stream.error==MAD_ERROR_BUFLEN)
continue;
else
{
fprintf(stderr,"%s: unrecoverable frame level error (%s).\n",
ProgName,MadErrorString(&Stream));
Status=1;
break;
}
}
/* The characteristics of the stream's first frame is printed
* on stderr. The first frame is representative of the entire
* stream.
*/
if(FrameCount==0)
if(PrintFrameInfo(stderr,&Frame.header))
{
Status=1;
break;
}
/* Accounting. The computed frame duration is in the frame
* header structure. It is expressed as a fixed point number
* whole data type is mad_timer_t. It is different from the
* samples fixed point format and unlike it, it can't directly
* be added or substracted. The timer module provides several
* functions to operate on such numbers. Be careful there, as
* some functions of mad's timer module receive some of their
* mad_timer_t arguments by value!
*/
FrameCount++;
mad_timer_add(&Timer,Frame.header.duration);
/* Between the frame decoding and samples synthesis we can
* perform some operations on the audio data. We do this only
* if some processing was required. Detailed explanations are
* given in the ApplyFilter() function.
*/
//if(DoFilter)
// ApplyFilter(&Frame);
/* Once decoded the frame is synthesized to PCM samples. No errors
* are reported by mad_synth_frame();
*/
mad_synth_frame(&Synth,&Frame);
/* Synthesized samples must be converted from mad's fixed
* point number to the consumer format. Here we use unsigned
* 16 bit big endian integers on two channels. Integer samples
* are temporarily stored in a buffer that is flushed when
* full.
*/
for(i=0;i<Synth.pcm.length;i++)
{
signed short Sample;
/* Left channel */
Sample=MadFixedToSshort(Synth.pcm.samples[0]);
*(OutputPtr++)=Sample>>0;
*(OutputPtr++)=Sample>>8;
/* Right channel. If the decoded stream is monophonic then
* the right output channel is the same as the left one.
*/
if(MAD_NCHANNELS(&Frame.header)==2)
Sample=MadFixedToSshort(Synth.pcm.samples[1]);
*(OutputPtr++)=Sample>>0;
*(OutputPtr++)=Sample>>8;
/* Flush the output buffer if it is full. */
if(OutputPtr==OutputBufferEnd)
{
//if(fwrite(OutputBuffer,1,OUTPUT_BUFFER_SIZE,OutputFp)!=OUTPUT_BUFFER_SIZE)
if(write(sfd, OutputBuffer, OUTPUT_BUFFER_SIZE)!=OUTPUT_BUFFER_SIZE)
{
fprintf(stderr,"%s: PCM write error (%s).\n",
ProgName,strerror(errno));
Status=2;
break;
}
OutputPtr=OutputBuffer;
}
}
}while(1);
/* The input file was completely read; the memory allocated by our
* reading module must be reclaimed.
*/
BstdFileDestroy(BstdFile);
/* Mad is no longer used, the structures that were initialized must
* now be cleared.
*/
mad_synth_finish(&Synth);
mad_frame_finish(&Frame);
mad_stream_finish(&Stream);
/* If the output buffer is not empty and no error occured during
* the last write, then flush it.
*/
if(OutputPtr!=OutputBuffer && Status!=2)
{
size_t BufferSize=OutputPtr-OutputBuffer;
//if(fwrite(OutputBuffer,1,BufferSize,OutputFp)!=BufferSize)
if(write(sfd,OutputBuffer,BufferSize)!=BufferSize)
{
fprintf(stderr,"%s: PCM write error (%s).\n",
ProgName,strerror(errno));
Status=2;
}
}
/* Accounting report if no error occured. */
if(!Status)
{
char Buffer[80];
/* The duration timer is converted to a human readable string
* with the versatile, but still constrained mad_timer_string()
* function, in a fashion not unlike strftime(). The main
* difference is that the timer is broken into several
* values according some of it's arguments. The units and
* fracunits arguments specify the intended conversion to be
* executed.
*
* The conversion unit (MAD_UNIT_MINUTES in our example) also
* specify the order and kind of conversion specifications
* that can be used in the format string.
*
* It is best to examine mad's timer.c source-code for details
* of the available units, fraction of units, their meanings,
* the format arguments, etc.
*/
mad_timer_string(Timer,Buffer,"%lu:%02lu.%03u",
MAD_UNITS_MINUTES,MAD_UNITS_MILLISECONDS,0);
fprintf(stderr,"%s: %lu frames decoded (%s).\n",
ProgName,FrameCount,Buffer);
}
/* That's the end of the world (in the H. G. Wells way). */
return(Status);
}
/****************************************************************************
* Program entry point. *
****************************************************************************/
int main(int argc, char *argv[])
{
int Status;
int i=0;
FILE *file_fd; //文件
long file_len; //文件长度
long loops;
unsigned char audio_buffer[BUF_SIZE];
unsigned char *file_name;
printf("This is a wav&mp3 play program.\n");
printf("Please add a file name and the audio rate!\nsuch as \"./test_music test.mp3\" \n\n\n");
//delay(100000);
ProgName=argv[0];
file_name = argv[1];
file_len = strlen(file_name);
if(file_len < 3)
{
printf("file is too short!\n");
exit(0);
}
printf("File name : %s \n",file_name);
//delay(100000);
//if file format is mp3//
if(file_name[file_len-3]=='m' && file_name[file_len-2]=='p' && file_name[file_len-1]=='3')
{
printf("Play mp3!\n");
// fd = open(file_name, O_RDONLY);
file_fd = fopen(file_name, "r");
if ((sfd = open("/dev/iis", O_WRONLY)) < 0)
{
printf("can not open device!!!\n");
return 5;
}
/* Decode stdin to stdout. */
Status=MpegAudioDecoder(file_fd);
if(Status)
fprintf(stderr,"%s: an error occured during decoding.\n",ProgName);
/* All done. */
fclose(file_fd);
close(sfd);
return(Status);
}
else
{
/*打开音频设备，准备play*/
if ((sfd = open(DEVICE_NAME, O_WRONLY)) == -1)
{
printf("open error\n");
return -1;
}
/*打开并计算文件长度*/
file_fd = fopen(file_name, "r");
fseek(file_fd,0,SEEK_END); //定位到文件末
file_len = ftell(file_fd); //文件长度
loops = file_len/(BUF_SIZE);
/*重新定位到文件头*/
fclose(file_fd);
file_fd = fopen(file_name, "r");
/*播放wav文件*/
for(i=0;i<loops;i++)
{
fread(audio_buffer, BUF_SIZE, 1, file_fd);
//printf("fread_ret=%d,loops=%d,i=%d\n",fread_ret,loops,i);
write(sfd,audio_buffer,BUF_SIZE);
}
/*关闭设备和文件*/
fclose(file_fd);
close(sfd);
}
return 0;
}
/*
* Local Variables:
* tab-width: 4
* End:
*/
/****************************************************************************
* End of file madllc.c *
****************************************************************************/
好，准备好这些文件后，开始动手，编译连接libmad，生成应用程序madlld。
arm-elf-gcc -static -Wl,-elf2flt -o madlld madlld.c bstdfile.c ./libmad.a
同样放到根文件系统bin下面，开始测试：
哈哈，总算成功拉！
这是源码code大礼包！
https://static.assets-stash.eet-china.com/album/old-resources/2010/4/24/f2c44bca-d69a-4e25-9a8b-2dc226c37f06.zip
来张照片合影下
这是手稿
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