/*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(¶ms); /* 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
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