* Maps G.723_24 code word to reconstructed scale factor normalized log
* magnitude values.
*/
-static short _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};
+static short _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};
/* Maps G.723_24 code word to log of scale factor multiplier. */
-static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
+static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
/*
* Maps G.723_24 code words to a set of values whose long and short
* term averages are computed and then compared to give an indication
* how stationary (steady state) the signal is.
*/
-static short _fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};
+static short _fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};
static short qtab_723_24[3] = {8, 218, 331};
*/
int
g723_24_encoder(
- int sl,
- int in_coding,
- struct g72x_state *state_ptr)
+ int sl,
+ int in_coding,
+ struct g72x_state *state_ptr)
{
- short sei, sezi, se, sez; /* ACCUM */
- short d; /* SUBTA */
- short y; /* MIX */
- short sr; /* ADDB */
- short dqsez; /* ADDC */
- short dq, i;
-
- switch (in_coding) { /* linearize input sample to 14-bit PCM */
- case AUDIO_ENCODING_ALAW:
- sl = alaw2linear(sl) >> 2;
- break;
- case AUDIO_ENCODING_ULAW:
- sl = ulaw2linear(sl) >> 2;
- break;
- case AUDIO_ENCODING_LINEAR:
- sl = ((short)sl) >> 2; /* sl of 14-bit dynamic range */
- break;
- default:
- return (-1);
- }
-
- sezi = predictor_zero(state_ptr);
- sez = sezi >> 1;
- sei = sezi + predictor_pole(state_ptr);
- se = sei >> 1; /* se = estimated signal */
-
- d = sl - se; /* d = estimation diff. */
-
- /* quantize prediction difference d */
- y = step_size(state_ptr); /* quantizer step size */
- i = quantize(d, y, qtab_723_24, 3); /* i = ADPCM code */
- dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */
-
- sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */
-
- dqsez = sr + sez - se; /* pole prediction diff. */
-
- update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
-
- return (i);
+ short sei, sezi, se, sez; /* ACCUM */
+ short d; /* SUBTA */
+ short y; /* MIX */
+ short sr; /* ADDB */
+ short dqsez; /* ADDC */
+ short dq, i;
+
+ switch (in_coding) { /* linearize input sample to 14-bit PCM */
+ case AUDIO_ENCODING_ALAW:
+ sl = alaw2linear(sl) >> 2;
+ break;
+ case AUDIO_ENCODING_ULAW:
+ sl = ulaw2linear(sl) >> 2;
+ break;
+ case AUDIO_ENCODING_LINEAR:
+ sl = ((short)sl) >> 2; /* sl of 14-bit dynamic range */
+ break;
+ default:
+ return (-1);
+ }
+
+ sezi = predictor_zero(state_ptr);
+ sez = sezi >> 1;
+ sei = sezi + predictor_pole(state_ptr);
+ se = sei >> 1; /* se = estimated signal */
+
+ d = sl - se; /* d = estimation diff. */
+
+ /* quantize prediction difference d */
+ y = step_size(state_ptr); /* quantizer step size */
+ i = quantize(d, y, qtab_723_24, 3); /* i = ADPCM code */
+ dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */
+
+ sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */
+
+ dqsez = sr + sez - se; /* pole prediction diff. */
+
+ update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
+
+ return (i);
}
/*
*/
int
g723_24_decoder(
- int i,
- int out_coding,
- struct g72x_state *state_ptr)
+ int i,
+ int out_coding,
+ struct g72x_state *state_ptr)
{
- short sezi, sei, sez, se; /* ACCUM */
- short y; /* MIX */
- short sr; /* ADDB */
- short dq;
- short dqsez;
-
- i &= 0x07; /* mask to get proper bits */
- sezi = predictor_zero(state_ptr);
- sez = sezi >> 1;
- sei = sezi + predictor_pole(state_ptr);
- se = sei >> 1; /* se = estimated signal */
-
- y = step_size(state_ptr); /* adaptive quantizer step size */
- dq = reconstruct(i & 0x04, _dqlntab[i], y); /* unquantize pred diff */
-
- sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */
-
- dqsez = sr - se + sez; /* pole prediction diff. */
-
- update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
-
- switch (out_coding) {
- case AUDIO_ENCODING_ALAW:
- return (tandem_adjust_alaw(sr, se, y, i, 4, qtab_723_24));
- case AUDIO_ENCODING_ULAW:
- return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24));
- case AUDIO_ENCODING_LINEAR:
- return (sr << 2); /* sr was of 14-bit dynamic range */
- default:
- return (-1);
- }
+ short sezi, sei, sez, se; /* ACCUM */
+ short y; /* MIX */
+ short sr; /* ADDB */
+ short dq;
+ short dqsez;
+
+ i &= 0x07; /* mask to get proper bits */
+ sezi = predictor_zero(state_ptr);
+ sez = sezi >> 1;
+ sei = sezi + predictor_pole(state_ptr);
+ se = sei >> 1; /* se = estimated signal */
+
+ y = step_size(state_ptr); /* adaptive quantizer step size */
+ dq = reconstruct(i & 0x04, _dqlntab[i], y); /* unquantize pred diff */
+
+ sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */
+
+ dqsez = sr - se + sez; /* pole prediction diff. */
+
+ update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
+
+ switch (out_coding) {
+ case AUDIO_ENCODING_ALAW:
+ return (tandem_adjust_alaw(sr, se, y, i, 4, qtab_723_24));
+ case AUDIO_ENCODING_ULAW:
+ return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24));
+ case AUDIO_ENCODING_LINEAR:
+ return (sr << 2); /* sr was of 14-bit dynamic range */
+ default:
+ return (-1);
+ }
}
projects / wxWidgets.git / blobdiff