예제 #1
0
파일: sidc.c 프로젝트: sorki/sidc
static inline void maybe_do_fft( void)
{
   if( ++grab_cnt < FFTWID) return;
   grab_cnt = 0;

   process_fft( &left);
   if( CF_chans == 2) process_fft( &right);

   if( ++frame_cnt == output_int)
   {
      frame_cnt = 0;
      output_record();
   }

   if( ++uspec_cnt == uspec_max)
   {
      uspec_cnt = 0;
      utility_spectrum();
   }
}
예제 #2
0
파일: mad.c 프로젝트: mvd7793/RED-SIREN
enum mad_flow output(void *data,
                     struct mad_header const *header,
                     struct mad_pcm *pcm)
{
    register int nsamples = pcm->length;
    mad_fixed_t const *left_ch = pcm->samples[0], *right_ch = pcm->samples[1];
    
    static unsigned char stream[1152*4]; /* 1152 because that's what mad has as a max; *4 because
                                    there are 4 distinct bytes per sample (in 2 channel case) */
    static unsigned int rate = 0;
    static int channels = 0;
    static struct audio_dither dither;

    register char * ptr = stream;
    register signed int sample;
    register mad_fixed_t tempsample;


    /* Semaphore operations */
    static struct sembuf read_sops = {0, -1, 0};
    static struct sembuf write_sops = {1, 1, 0};

    if(options.opt & MPG321_ENABLE_BUFFER)
    {
	    semop(semarray,&read_sops,1);
	    ptr = (Output_Queue+mad_decoder_position)->data;
	    memcpy(&((Output_Queue+mad_decoder_position)->header),header,sizeof(struct mad_header));
			 
    }else{
	    /* We need to know information about the file before we can open the playdevice
	       in some cases. So, we do it here. */
	    if (!playdevice)
	    {
	    	    channels = MAD_NCHANNELS(header);
	    	    rate = header->samplerate;
	    	    open_ao_playdevice(header);        
	    }
	    else if ((channels != MAD_NCHANNELS(header) || rate != header->samplerate) && playdevice_is_live())
	    {
	    	    ao_close(playdevice);
	    	    channels = MAD_NCHANNELS(header);
	    	    rate = header->samplerate;
	    	    open_ao_playdevice(header);        
	    }        
    }

    static int peak_rate = 0;
    static unsigned short int peak = 0;


    if (pcm->channels == 2)
    {
        while (nsamples--)
        {
            tempsample = mad_f_mul(*left_ch++, options.volume);
            sample = (signed int) audio_linear_dither(16, tempsample, &dither);

	    peak = abs(sample) > peak ? abs(sample) : peak;

#ifndef WORDS_BIGENDIAN
            *ptr++ = (unsigned char) (sample >> 0);
            *ptr++ = (unsigned char) (sample >> 8);
#else
            *ptr++ = (unsigned char) (sample >> 8);
            *ptr++ = (unsigned char) (sample >> 0);
#endif
            
            tempsample = mad_f_mul(*right_ch++, options.volume);
            sample = (signed int) audio_linear_dither(16, tempsample, &dither);

	    peak = abs(sample) > peak ? abs(sample) : peak;

#ifndef WORDS_BIGENDIAN
            *ptr++ = (unsigned char) (sample >> 0);
            *ptr++ = (unsigned char) (sample >> 8);
#else
            *ptr++ = (unsigned char) (sample >> 8);
            *ptr++ = (unsigned char) (sample >> 0);
#endif
        }
	
	process_fft(stream, pcm->length * 4);
	if(options.opt & MPG321_ENABLE_BUFFER)
	{
		(Output_Queue+mad_decoder_position)->length = pcm->length * 4;
	}else
	{
		ao_play(playdevice, stream, pcm->length * 4);
	}
    }
    
    else if (options.opt & MPG321_FORCE_STEREO)
예제 #3
0
파일: mad.c 프로젝트: e3c/mpg321
enum mad_flow output(void *data,
                     struct mad_header const *header,
                     struct mad_pcm *pcm)
{
    register int nsamples = pcm->length;
    mad_fixed_t const *left_ch = pcm->samples[0], *right_ch = pcm->samples[1];

    static unsigned char stream[1152*4]; /* 1152 because that's what mad has as a max; *4 because
                                    there are 4 distinct bytes per sample (in 2 channel case) */
    static unsigned int rate = 0;
    static int channels = 0;
    static struct audio_dither dither;

    register char * ptr = stream;
    register signed int sample;
    register mad_fixed_t tempsample;

    /* We need to know information about the file before we can open the playdevice
       in some cases. So, we do it here. */
    if (!playdevice)
    {
        channels = MAD_NCHANNELS(header);
        rate = header->samplerate;
        open_ao_playdevice(header);
    }

    else if ((channels != MAD_NCHANNELS(header) || rate != header->samplerate) && playdevice_is_live())
    {
        ao_close(playdevice);
        channels = MAD_NCHANNELS(header);
        rate = header->samplerate;
        open_ao_playdevice(header);
    }

    static int peak_rate = 0;
    static unsigned short int peak = 0;

    if (pcm->channels == 2)
    {
        while (nsamples--)
        {
            tempsample = (mad_fixed_t)((*left_ch++ * (double)options.volume)/MAD_F_ONE);
            sample = (signed int) audio_linear_dither(16, tempsample, &dither);

            peak = abs(sample) > peak ? abs(sample) : peak;

#ifndef WORDS_BIGENDIAN
            *ptr++ = (unsigned char) (sample >> 0);
            *ptr++ = (unsigned char) (sample >> 8);
#else
            *ptr++ = (unsigned char) (sample >> 8);
            *ptr++ = (unsigned char) (sample >> 0);
#endif

            tempsample = (mad_fixed_t)((*right_ch++ * (double)options.volume)/MAD_F_ONE);
            sample = (signed int) audio_linear_dither(16, tempsample, &dither);

            peak = abs(sample) > peak ? abs(sample) : peak;

#ifndef WORDS_BIGENDIAN
            *ptr++ = (unsigned char) (sample >> 0);
            *ptr++ = (unsigned char) (sample >> 8);
#else
            *ptr++ = (unsigned char) (sample >> 8);
            *ptr++ = (unsigned char) (sample >> 0);
#endif
        }

        process_fft(stream, pcm->length * 4);
        ao_play(playdevice, stream, pcm->length * 4);
    }

    else if (options.opt & MPG321_FORCE_STEREO)