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Taken from libdca svn: svn://svn.videolan.org/libdca/trunk.
r84 | gbazin | 2008-06-01 16:13:33 +0000 (Sun, 01 Jun 2008) | 3 lines
* libdca/parse.c: Change output normalisation factor from 3/2 to sqrt(2).
Thanks to Alexander E. Patrakov for finding that this is the proper normalisation factor.
Fixed a bug where the output bias wasn't applied when downmixing wasn't being done.
--- libdca.orig/libdca/parse.c
+++ libdca/libdca/parse.c
@@ -59,12 +59,11 @@
static int decode_blockcode (int code, int levels, int *values);
static void qmf_32_subbands (dca_state_t * state, int chans,
- double samples_in[32][8], sample_t *samples_out,
- double rScale, sample_t bias);
+ double samples_in[32][8], sample_t *samples_out);
static void lfe_interpolation_fir (int nDecimationSelect, int nNumDeciSample,
double *samples_in, sample_t *samples_out,
- double rScale, sample_t bias );
+ sample_t bias);
static void pre_calc_cosmod( dca_state_t * state );
@@ -123,7 +122,9 @@
bitstream_get (state, 1);
*frame_length = (bitstream_get (state, 7) + 1) * 32;
+ if (*frame_length < 6 * 32) return 0;
frame_size = bitstream_get (state, 14) + 1;
+ if (frame_size < 96) return 0;
if (!state->word_mode) frame_size = frame_size * 8 / 14 * 2;
/* Audio channel arrangement */
@@ -981,14 +982,7 @@
/* 32 subbands QMF */
for (k = 0; k < state->prim_channels; k++)
{
- /*static double pcm_to_float[8] =
- {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
-
- qmf_32_subbands (state, k,
- subband_samples[k],
- &state->samples[256*k],
- /*WTF ???*/ 32768.0*3/2/*pcm_to_float[state->source_pcm_res]*/,
- 0/*state->bias*/);
+ qmf_32_subbands (state, k, subband_samples[k], &state->samples[256*k]);
}
/* Down/Up mixing */
@@ -1000,6 +994,10 @@
{
dca_downmix (state->samples, state->amode, state->output, state->bias,
state->clev, state->slev);
+ } else if (state->bias)
+ {
+ for ( k = 0; k < 256*state->prim_channels; k++ )
+ state->samples[k] += state->bias;
}
/* Generate LFE samples for this subsubframe FIXME!!! */
@@ -1011,8 +1009,7 @@
lfe_interpolation_fir (state->lfe, 2 * state->lfe,
state->lfe_data + lfe_samples +
2 * state->lfe * subsubframe,
- &state->samples[256*i_channels],
- 8388608.0, state->bias);
+ &state->samples[256*i_channels], state->bias);
/* Outputs 20bits pcm samples */
}
@@ -1142,9 +1139,9 @@
}
static void qmf_32_subbands (dca_state_t * state, int chans,
- double samples_in[32][8], sample_t *samples_out,
- double scale, sample_t bias)
+ double samples_in[32][8], sample_t *samples_out)
{
+ static const double scale = 1.4142135623730951 /* sqrt(2) */ * 32768.0;
const double *prCoeff;
int i, j, k;
double raXin[32];
@@ -1211,7 +1208,7 @@
/* Create 32 PCM output samples */
for (i=0;i<32;i++)
- samples_out[nChIndex++] = subband_fir_hist2[i] / scale + bias;
+ samples_out[nChIndex++] = subband_fir_hist2[i] / scale;
/* Update working arrays */
for (i=511;i>=32;i--)
@@ -1225,7 +1222,7 @@
static void lfe_interpolation_fir (int nDecimationSelect, int nNumDeciSample,
double *samples_in, sample_t *samples_out,
- double scale, sample_t bias)
+ sample_t bias)
{
/* samples_in: An array holding decimated samples.
* Samples in current subframe starts from samples_in[0],
@@ -1235,6 +1232,7 @@
* samples_out: An array holding interpolated samples
*/
+ static const double scale = 8388608.0;
int nDeciFactor, k, J;
const double *prCoeff;