让终端支持播放mp3,移植mp3解码库libmad和madplay到嵌入式linux

libmad简介

MAD (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,完全是定点计算,非常适合没有浮点支持的平台上使用。使用 libmad 提供的一系列 API,就可以非常简单地实现 MP3 数据解码工作。


移植涉及到的库:
zlib-1.2.3.tar.gz
libid3tag-0.15.1b.tar.gz
libmad-0.15.1b.tar.gz
---------------------------------------------------------------------------------------------------------------
madplay介绍:
 madplay基于libmad的基础上做了一个播放器,该播放器除了目前不支持网络播放以为,其余功能都支持。如快进播放,seek播放,暂停,恢复等
最后移植了一个基于libmad的应用madplay,可以直接用它来播放mp3.
madplay-0.15.2b.tar.gz

 

介绍完了,当然移植不是一番风顺的,折腾了一天。中间遇到不少问题,还好,最后都一一解决了。

网上提供的代码。交叉编译,由于环境不一样,会出现各种问题。

这里记录一下过程。

首先是交叉编译zlib-1.2.3.tar.gz

用交叉编译工具编译zlib,并且把库生成到交叉编译环境的库目录下
./configure --prefix=/home/ban/madplay/source   
修改Makefile.
CC=arm-linux-gnueabihf-gcc
AR=arm-linux-gnueabihf-ar rc
RANLIB=arm-linux-gnueabihf-ranlib
make 
make install
安装完成后,在/home/ban/madplay/source/ 中将生产lib跟include2个文件夹。

这步一般不会有啥问题,但是,默认成功的是静态库啊, 虽然配置上是说默认配置生成动态库,但是确实没有。

实际是有的,指定--shared即可。

或者仔细查makefile,把相关的编译语句找出来,我手动调用gcc - shared -fPIC - $(OBJS)生成了.so

这里还需要注意的是,指定好自己的--prifix,因为后续的编译,好多是要依赖这个的。

接下来编译libid3tag-0.15.1b.tar.gz

这时候要注意了,如果上一步编译不过,或者没有指定--prifix, 这里就麻烦了。

由于我需要的是动态库,发现configer后,竟没带-FPIC参数,还要去改makefile才行。

 

./configure --host=arm-linux-gnueabihf  --disable-debugging --prefix=/home/ban/madplay/source CPPFLAGS=-I/home/ban/madplay/source/include LDFLAGS=-L/home/ban/madplay/source/lib
make 
make install

 

编译libmad
./configure --host=arm-linux-gnueabihf  --disable-debugging --prefix=/home/ban/madplay/source CPPFLAGS=-I/home/ban/madplay/source/include LDFLAGS=-L/home/ban/madplay/source/lib
make
make install 

 

出现错误:
cc1: error: unrecognized command line option “-fforce-mem”
原因是高版本的gcc,已经将-fforce-mem去除了,解决方法:
 sed -i '/-fforce-mem/d' configure

再执行:
./configure --host=arm-linux-gnueabihf --prefix=/usr/local/libmad_arm --enable-shared --enable-static --enable-fpm=arm --


with-gnu-ld=arm-linux-gnueabihf-ld --build=arm
出现错误:
/tmp/ccf2FxyW.s:1299: Error: selected processor does not support Thumb mode `rsc r0,r0,#0'
/tmp/ccf2FxyW.s:1435: Error: selected processor does not support Thumb mode `rsc r8,r8,#0'
/tmp/ccf2FxyW.s:1857: Error: selected processor does not support Thumb mode `rsc r0,r0,#0'
/tmp/ccf2FxyW.s:1996: Error: selected processor does not support Thumb mode `rsc r0,r0,#0
百度一下发现这是libmad的一个bug.
解决方法是:
vim  fixed.h

#  define MAD_F_MLN(hi, lo)  \
    asm ("rsbs  %0, %2, #0\n\t"  \
         "rsc   %1, %3, #0"  \
         : "=r" (lo), "=r" (hi)  \
         : "0" (lo), "1" (hi)  \
         : "cc")
改为
#ifdef __thumb__
/* In Thumb-2, the RSB-immediate instruction is only allowed with a zero
operand. If needed this code can also support Thumb-1 
(simply append "s" to the end of the second two instructions). */
# define MAD_F_MLN(hi, lo) \
asm ("rsbs %0, %0, #0\n\t" \
"       sbc %1, %1, %1\n\t" \
        "sub %1, %1, %2" \
        : "+&r" (lo), "=&r" (hi) \
        : "r" (hi) \
        : "cc")
#else /* ! __thumb__ */
# define MAD_F_MLN(hi, lo) \
        asm ("rsbs %0, %2, #0\n\t" \
        "rsc %1, %3, #0" \
         : "=r" (lo), "=r" (hi) \
          : "=&r" (lo), "=r" (hi) \
          : "0" (lo), "1" (hi) \
          : "cc")
#endif /* __thumb__ */
再make,编译通过了!

编译madplay
./configure --host=arm-linux-gnueabihf CC=arm-linux-gnueabihf-gcc --disable-debugging --with-alsa CPPFLAGS=-I/home/ban/madplay/source/include LDFLAGS=-L/home/ban/madplay/source/lib
make 
make install
完成以后把生成的可执行文件madplay下载到开发板中
执行./madplay filename.mp3

这个需要注意的是,如果不指定--with-alsa,即便编译成功了,放到板子上也是跑不起的,提示找不到dev/dsp,这个让我折腾了好久,竟发现,配置上没启用alsa啊,

但板子上带的是alsa架构的linux音频驱动。

 

总体上操作是就这么几步,但是,你会发现,如果照这个步骤来,仍是有错。

具体细节。,根据编译提示的错误,基本都能定为到。比如,找不到上几步编译出的库,就去改makefile吧,添加进去路径

或者仍拷贝到 --prifix指定的目录中。
 

最后再说一点儿,编译网上这种开源库,最好设置下交叉工具链的环境变量为全局的,且用root权限。否则,坑真的好多。

 

附截图:

让终端支持播放mp3,移植mp3解码库libmad和madplay到嵌入式linux_第1张图片

 

让终端支持播放mp3,移植mp3解码库libmad和madplay到嵌入式linux_第2张图片

如果不用这个现成的播放器madplay,只测试下libmad是否成功,

可以编译测试下 libmad提供的一个简单demo,这个demo 不是播放mp3的,而是把mp3解码成 pcm文件 。

测试如下:

./testmad.out out1.pcm     

显示出了信息,且在当前路径下产生了out1.pcm文件。

9522 frames decoded (0:04:08.7), +1.7 dB peak amplitude, 4202 clipped samples

 

/*
 * 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 
# include 
# include 
# include 

# 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 main(int argc, char *argv[])
{
  struct stat stat;
  void *fdm;

  if (argc != 1)
    return 1;

  if (fstat(STDIN_FILENO, &stat) == -1 ||
      stat.st_size == 0)
    return 2;

  fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, STDIN_FILENO, 0);
  if (fdm == MAP_FAILED)
    return 3;

  decode(fdm, stat.st_size);

  if (munmap(fdm, stat.st_size) == -1)
    return 4;

  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;

  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;
  mad_fixed_t const *left_ch, *right_ch;

  /* pcm->samplerate contains the sampling frequency */

  nchannels = pcm->channels;
  nsamples  = pcm->length;
  left_ch   = pcm->samples[0];
  right_ch  = pcm->samples[1];

  while (nsamples--) {
    signed int sample;

    /* output sample(s) in 16-bit signed little-endian PCM */

    sample = scale(*left_ch++);
    putchar((sample >> 0) & 0xff);
    putchar((sample >> 8) & 0xff);

    if (nchannels == 2) {
      sample = scale(*right_ch++);
      putchar((sample >> 0) & 0xff);
      putchar((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)
{
  struct buffer *buffer = data;

  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;
}

 

 

 

 

 

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