diff options
Diffstat (limited to 'src/libdca-2-normalisation-factor-sqrt2+output-bias.patch')
| -rw-r--r-- | src/libdca-2-normalisation-factor-sqrt2+output-bias.patch | 112 |
1 files changed, 112 insertions, 0 deletions
diff --git a/src/libdca-2-normalisation-factor-sqrt2+output-bias.patch b/src/libdca-2-normalisation-factor-sqrt2+output-bias.patch new file mode 100644 index 0000000..2d4cdb5 --- /dev/null +++ b/src/libdca-2-normalisation-factor-sqrt2+output-bias.patch @@ -0,0 +1,112 @@ +This file is part of MXE. +See index.html for further information. + +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; + |