コード例 #1
0
ファイル: asciiU2S_main.c プロジェクト: anselm94/mimic
int main()
{
    int s;

    while (scanf("%d\n", &s) != EOF)
        printf("%d\n", (int) cst_ulaw_to_short((int) s));

    return 0;
}
コード例 #2
0
cst_wave *lpc_resynth(cst_lpcres *lpcres)
{
    cst_wave *w;
    int i,j,r,o,k;
    int ci,cr;
    float *outbuf, *lpccoefs;
    int pm_size_samps;

    /* Get a new wave to build the signal into */
    w = new_wave();
    cst_wave_resize(w,lpcres->num_samples,1);
    w->sample_rate = lpcres->sample_rate;
    /* outbuf is a circular buffer with past relevant samples in it */
    outbuf = cst_alloc(float,1+lpcres->num_channels);
    /* unpacked lpc coefficients */
    lpccoefs = cst_alloc(float,lpcres->num_channels);

    for (r=0,o=lpcres->num_channels,i=0; i < lpcres->num_frames; i++)
    {
	pm_size_samps = lpcres->sizes[i];

	/* Unpack the LPC coefficients */
	for (k=0; k<lpcres->num_channels; k++)
	{
	    lpccoefs[k] = (float)((((double)lpcres->frames[i][k])/65535.0)*
			   lpcres->lpc_range) + lpcres->lpc_min;
	}
	/* Note we don't zero the lead in from the previous part */
	/* seems like you should but it makes it worse if you do */
/*	memset(outbuf,0,sizeof(float)*(1+lpcres->num_channels)); */

	/* resynthesis the signal */
	for (j=0; j < pm_size_samps; j++,r++)
	{
            outbuf[o] = (float)cst_ulaw_to_short(lpcres->residual[r]);
	    cr = (o == 0 ? lpcres->num_channels : o-1);
	    for (ci=0; ci < lpcres->num_channels; ci++)
	    {
		outbuf[o] += lpccoefs[ci] * outbuf[cr];
		cr = (cr == 0 ? lpcres->num_channels : cr-1);
	    }
	    w->samples[r] = (short)(outbuf[o]);
	    o = (o == lpcres->num_channels ? 0 : o+1);
	}
    }

    cst_free(outbuf);
    cst_free(lpccoefs);

    return w;

}
コード例 #3
0
cst_wave *lpc_resynth_sfp(cst_lpcres *lpcres)
{
    /* The fixed point spike excited, without floats */
    cst_wave *w;
    int i,j,r,o,k;
    int ci,cr;
    int *outbuf, *lpccoefs;
    int pm_size_samps, ilpc_min, ilpc_range;
    //int pp = 0;

    /* Get a new wave to build the signal into */
    w = new_wave();
    cst_wave_resize(w,lpcres->num_samples,1);
    w->sample_rate = lpcres->sample_rate;
    /* outbuf is a circular buffer with past relevant samples in it */
    outbuf = cst_alloc(int,1+lpcres->num_channels);
    /* unpacked lpc coefficients */
    lpccoefs = cst_alloc(int,lpcres->num_channels);
    ilpc_min = (int)(lpcres->lpc_min*32768.0);
    /* assume range is never > abs(16) */
    ilpc_range = (int)(lpcres->lpc_range*2048.0);

    for (r=0,o=lpcres->num_channels,i=0; i < lpcres->num_frames; i++)
    {
	pm_size_samps = lpcres->sizes[i];

	/* Unpack the LPC coefficients */
	for (k=0; k<lpcres->num_channels; k++)
	    lpccoefs[k]=((lpcres->frames[i][k]/2*ilpc_range)/2048+ilpc_min)/2;

	/* resynthesis the signal */
	for (j=0; j < pm_size_samps; j++,r++)
	{
	    outbuf[o] = (int)cst_ulaw_to_short(lpcres->residual[r]);
	    cr = (o == 0 ? lpcres->num_channels : o-1);
	    for (ci=0; ci < lpcres->num_channels; ci++)
	    {
		outbuf[o] += (lpccoefs[ci]*outbuf[cr])/16384;
		cr = (cr == 0 ? lpcres->num_channels : cr-1);
	    }
	    w->samples[r] = (short)outbuf[o];
	    //pp = outbuf[o];
	    o = (o == lpcres->num_channels ? 0 : o+1);
	}
    }

    cst_free(outbuf);
    cst_free(lpccoefs);

    return w;

}
コード例 #4
0
cst_wave *reconstruct_wave(cst_wave *sig, cst_sts *sts, cst_track *lpc)
{
    cst_lpcres *lpcres;
    int i,j,r;
    int start;
    int num_samples;
/*    FILE *ofd; */

    for (num_samples = 0, i=0; i < lpc->num_frames; i++)
	num_samples += sts[i].size;

    lpcres = new_lpcres();
    lpcres_resize_frames(lpcres,lpc->num_frames);
    lpcres->num_channels = lpc->num_channels-1;
    start = (int)((float)sig->sample_rate * lpc->times[0]/2);
    num_samples += start;
    for (i=0; i<lpc->num_frames; i++)
    {
	lpcres->frames[i] = sts[i].frame;
	lpcres->sizes[i] = sts[i].size;
    }
    lpcres_resize_samples(lpcres,num_samples);
    lpcres->lpc_min = lpc_min;
    lpcres->lpc_range = lpc_range;
    lpcres->sample_rate = sig->sample_rate;
    for (r=start,i=0; i<lpc->num_frames; i++)
	for (j=0; j<sts[i].size; j++,r++)
	    lpcres->residual[r] = sts[i].residual[j];

#if 0
    /* Debug dump */
    ofd = fopen("lpc_resid.lpc","w");
    for (s=0,i=0; i<lpcres->num_frames; i++)
    {
	fprintf(ofd,"%d %d %d\n",i,0,lpcres->sizes[i]);
	for (j=0; j < lpcres->num_channels; j++)
	    fprintf(ofd,"%d ",lpcres->frames[i][j]);
	fprintf(ofd,"\n");
	for (j=0; j < lpcres->sizes[i]; j++,s++)
	    fprintf(ofd,"%d ",lpcres->residual[s]);
	fprintf(ofd,"\n");
    }
    fclose(ofd);
    ofd = fopen("lpc_resid.res","w");
    for (i=0; i < r; i++)
	fprintf(ofd,"%d\n",cst_ulaw_to_short(lpcres->residual[i]));
    fclose(ofd);
#endif

    return lpc_resynth(lpcres);
}
コード例 #5
0
static int play_wave_from_socket(snd_header *header,int audiostream)
{
    /* Read audio from stream and play it to audio device, converting */
    /* it to pcm if required                                          */
    int num_samples;
    int sample_width;
    cst_audiodev *audio_device;
    int q,i,n,r;
    unsigned char bytes[CST_AUDIOBUFFSIZE];
    short shorts[CST_AUDIOBUFFSIZE];
    cst_file fff;

    fff = cst_fopen("/tmp/awb.wav",CST_OPEN_WRITE|CST_OPEN_BINARY);

    if ((audio_device = audio_open(header->sample_rate,1,
				   (header->encoding == CST_SND_SHORT) ?
				   CST_AUDIO_LINEAR16 : CST_AUDIO_LINEAR8)) == NULL)
    {
	cst_errmsg("play_wave_from_socket: can't open audio device\n");
	return -1;
    }

    if (header->encoding == CST_SND_SHORT)
	sample_width = 2;
    else
	sample_width = 1;

    num_samples = header->data_size / sample_width;
    /* we naively let the num_channels sort itself out */
    for (i=0; i < num_samples; i += r/2)
    {
	if (num_samples > i+CST_AUDIOBUFFSIZE)
	    n = CST_AUDIOBUFFSIZE;
	else
	    n = num_samples-i;
	if (header->encoding == CST_SND_ULAW)
	{
	    r = read(audiostream,bytes,n);
	    for (q=0; q<r; q++)
		shorts[q] = cst_ulaw_to_short(bytes[q]);
	    r *= 2;
	}
	else /* if (header->encoding == CST_SND_SHORT) */
	{
	    r = read(audiostream,shorts,n*2);
	    if (CST_LITTLE_ENDIAN)
		for (q=0; q<r/2; q++)
		    shorts[q] = SWAPSHORT(shorts[q]);
	}
	
	if (r <= 0)
	{   /* I'm not getting any data from the server */
	    audio_close(audio_device);
	    return CST_ERROR_FORMAT;
	}
	
	for (q=r; q > 0; q-=n)
	{
	    n = audio_write(audio_device,shorts,q);
	    cst_fwrite(fff,shorts,2,q);
	    if (n <= 0)
	    {
		audio_close(audio_device);
		return CST_ERROR_FORMAT;
	    }
	}
    }
    audio_close(audio_device);
    cst_fclose(fff);

    return CST_OK_FORMAT;

}