简单mp3的实现
2012-05-18 11:16:19 来源:WEB开发网核心提示: 使用libmad播放mp3格式的文件,gcc play.c -o play -lmad 运行./play < ***.mp3 ,简单mp3的实现,通过将解码后的pcm直接写到声卡,实现播放#include <stdio.h>#include <stdlib.h>#include &
使用libmad播放mp3格式的文件。gcc play.c -o play -lmad 运行./play < ***.mp3 。通过将解码后的pcm直接写到声卡,实现播放
#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <string.h> #include <fcntl.h> #include <sys/ioctl.h> #include <sys/soundcard.h> # include <sys/stat.h> # include <sys/mman.h> # include "mad.h" int soundfd; void set_dsp() { int format = AFMT_S16_LE; int channels = 2; //CHANNELS int rate = 44100; //HZ if((soundfd = open("/dev/dsp" , O_WRONLY)) < 0) { fprintf(stderr , "can't open sound device!\n"); exit(-1); } ioctl(soundfd , SNDCTL_DSP_SPEED , &rate); ioctl(soundfd , SNDCTL_DSP_SETFMT, &format); ioctl(soundfd , SNDCTL_DSP_CHANNELS , &channels); return ; } void write_dsp(int c) { write(soundfd , (char *)&c , 1); return ; } /* * 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; set_dsp(); 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++); write_dsp((sample >> 0) & 0xff); write_dsp((sample >> 8) & 0xff); if (nchannels == 2) { sample = scale(*right_ch++); write_dsp((sample >> 0) & 0xff); write_dsp((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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