libmad学习进阶4 -----基于atlas音频驱动架构的MP3播放器

 /*modify by hfl 20140216*/ #define ALSA_PCM_NEW_HW_PARAMS_API # include <stdio.h> # include <unistd.h> # include <sys/stat.h> # include <sys/mman.h> # include "mad.h" #include<sys/types.h> #include<sys/stat.h> #include<fcntl.h> #include<stdlib.h> #include <sys/ioctl.h> #include <sys/soundcard.h> #include <alsa/asoundlib.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.  */  //#define printf     static Get_file_length(char *PATH); static int init_dsp(); static int Uninit_dsp(); static int decode(unsigned char const *, unsigned long); static enum mad_flow outputplay(void *data,     struct mad_header const *header,     struct mad_pcm *pcm); int main(int argc, char *argv[]) { printf("The main is start!\n");   struct stat stat;   void *fdm;   int  fd;   //char buffer1[80000];   printf("###The input file is %s  ! the arc=%d###\n",argv[1],argc);   if (argc == 1)    {    printf("The argc is wrong!\n");     return 1;  } #if 0   if (fstat(STDIN_FILENO, &stat) == -1 ||       stat.st_size == 0)     return 2; #endif  fd =open(argv[1],O_RDWR); if(-1==fd) {    printf("sorry,The file open is faild!\n"); } else {  printf("The file open is sucessed!\n"); } //read(fd,buffer1,sizeof(buffer1)); //printf("%s", buffer1); stat.st_size = Get_file_length(argv[1]); printf("The file size is %d\n",stat.st_size ); printf("The Map is begin ok!\n"); fdm = mmap(0, stat.st_size, PROT_READ, MAP_SHARED, fd, 0);   if (fdm == MAP_FAILED) { printf("mmap is failed\n");     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; }; int id; int flag=0; snd_pcm_t *handle;  snd_pcm_uframes_t frames =1024;  int fd=0; /*初始化音频设备*/ int init_dsp(int rate,int channels) {   int rc;   snd_pcm_hw_params_t *params;   int dir;    /* Open PCM device for playback. */   rc = snd_pcm_open(&handle, "default",                     SND_PCM_STREAM_PLAYBACK, 0);   if (rc < 0) {     fprintf(stderr,             "unable to open pcm device: %s\n",             snd_strerror(rc));     exit(1);   }   /* Allocate a hardware parameters object. */   snd_pcm_hw_params_alloca(&params);   /* Fill it in with default values. */   snd_pcm_hw_params_any(handle, params);   /* Set the desired hardware parameters. */   /* Interleaved mode */   snd_pcm_hw_params_set_access(handle, params,                       SND_PCM_ACCESS_RW_INTERLEAVED);   /* Signed 16-bit little-endian format */   snd_pcm_hw_params_set_format(handle, params,                               SND_PCM_FORMAT_S16_LE);   /* Two channels (stereo) */   printf("channel=%d\n", channels);   snd_pcm_hw_params_set_channels(handle, params, channels);   /* 44100 bits/second sampling rate (CD quality) */  // val = 16000;   snd_pcm_hw_params_set_rate_near(handle, params,                                   & rate, &dir); printf("rate=%d\n",rate);   /* Set period size to 32 frames. */   /*一次送人的帧太少，会下溢冲（至少15帧)*/  // snd_pcm_hw_params_set_period_size_near(handle,  params, &frames, &dir);   /* Write the parameters to the driver */   rc = snd_pcm_hw_params(handle, params);   if (rc < 0) {     fprintf(stderr,             "unable to set hw parameters: %s\n",             snd_strerror(rc));     exit(1);   } printf( "The Dsp init is atlas ok!\n"); return 0; } static int Uninit_dsp() {    //fclose(fdout);    snd_pcm_drain(handle);    snd_pcm_close(handle);    printf("play end \n"); } /*  * 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; printf("1111");   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); } static int Get_file_length(char *PATH) {        FILE *fp;      fp=fopen(PATH,"r");     if(!fp)    {    printf("sorry,The file open is faild!\n");    }    else     {     printf("The file open is sucessed!\n");    }    fseek(fp, 0L,SEEK_END);    return (ftell(fp)); } /*  * 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;   static FILE *fdout;   char buf[1];   /* pcm->samplerate contains the sampling frequency */  fdout= fopen("mypcm.pcm","ab+");   if(!fdout) {  printf("open is failed\n"); }   else printf("out open is ok\n");   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++);     buf[0]=(sample >> 0) & 0xff;     printf("%d\t",buf[0]);     fwrite(buf,1,1,fdout);     buf[0]=(sample >> 8) & 0xff;   printf("%d\t",buf[0]);     fwrite(buf,1,1,fdout);     if (nchannels == 2) {      sample = scale(*right_ch++);      buf[0]=(sample >> 0) & 0xff; fwrite(buf,1,1,fdout);      buf[0]=(sample >> 8) & 0xff; fwrite(buf,1,1,fdout);     }   } fclose(fdout);   return MAD_FLOW_CONTINUE; } static enum mad_flow outputplay(void *data,     struct mad_header const *header,     struct mad_pcm *pcm) {   unsigned int nchannels;   long int nsamples,samplerate;   mad_fixed_t const *left_ch, *right_ch;    static int i=0;   char buf[1];   static char buffer[1024*2*2];   /* pcm->samplerate contains the sampling frequency */   nchannels = pcm->channels;   nsamples  = pcm->length;/* 这个不是采样位，而一帧的数据长度12*3(采样)*32(子带)=1152*/   left_ch   = pcm->samples[0];   right_ch  = pcm->samples[1];   samplerate=pcm->samplerate;   if(!flag) {   printf("channels=%d, samples2=%ld,flag=%d\n", nchannels,samplerate,flag); printf("init dsp is begin\n");   init_dsp(samplerate,nchannels);   memset(buffer,0,sizeof(buffer));    flag++; } #if 1   while (nsamples--) {     signed int sample;     /* output sample(s) in 16-bit signed little-endian PCM */     sample = scale(*left_ch++);     buf[0]=(sample >> 0) & 0xff;     memcpy(buffer+i,buf,1);     i++;    // printf("i=%d,%d,%d\t",i,buf[i-1],buf[0]);     buf[0]=(sample >> 8) & 0xff;   memcpy(buffer+i,buf,1);   i++;     if (nchannels == 2) {      sample = scale(*right_ch++);      buf[0]=(sample >> 0) & 0xff;  memcpy(buffer+i,buf,1);             i++;      buf[0]=(sample >> 8) & 0xff; memcpy(buffer+i,buf,1);           i++;     } if(i==frames*2*nchannels) { i=0; snd_pcm_writei(handle, buffer, frames); }   }   #endif   //snd_pcm_writei(handle, buffer, frames);   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);     Uninit_dsp();   /* 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 */, outputplay,   error, 0 /* message */);   /* start decoding */   result = mad_decoder_run(&decoder, MAD_DECODER_MODE_SYNC);   /* release the decoder */   mad_decoder_finish(&decoder);   return result;

}

以上是基于alas音频驱动的mp3播放器。这里要注意alas送数据是以帧为单位送数据。而oss是以字节为单位，所以先要攒包到frame,再送数据。snd_pcm_writei(handle, buffer, frames); 要注意frames和字节的换算关系:size=frame*(每个采样率所占字节数）*声道数。同时frames不能太小，太小会解码器数据不够f而下溢出。frames只是32。本代码为1M，为的防止概率性同步不上问题

注意alsa架构要链接到alsa库，注意修改makefile编译选项。

CFLAGS = -Wall -march=i486 -g -O  -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -I/usr/include/alsa -lasound

编译命令：sudo make minimad

即可