Merge commit '4a2ef394'

* commit '4a2ef394':
  cosmetics: Add '0' to float constants ending in '.'.

Conflicts:
	libavcodec/ra288.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/aac_tablegen.h

@@ -35,7 +35,7 @@ void ff_aac_tableinit(void)
 {
     int i;
     for (i = 0; i < 428; i++)
-        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.);
+        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
 }
 #endif /* CONFIG_HARDCODED_TABLES */
 

libavcodec/aacdec.c

@@ -1224,7 +1224,7 @@ static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
             int run_end = band_type_run_end[idx];
             if (band_type[idx] == ZERO_BT) {
                 for (; i < run_end; i++, idx++)
-                    sf[idx] = 0.;
+                    sf[idx] = 0.0;
             } else if ((band_type[idx] == INTENSITY_BT) ||
                        (band_type[idx] == INTENSITY_BT2)) {
                 for (; i < run_end; i++, idx++) {
@@ -1968,7 +1968,7 @@ static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
         int idx  = 0;
         int cge  = 1;
         int gain = 0;
-        float gain_cache = 1.;
+        float gain_cache = 1.0;
         if (c) {
             cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
             gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;

libavcodec/aacps_tablegen.h

@@ -192,7 +192,7 @@ static void ps_tableinit(void)
     for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
         double f_center, theta;
         if (k < FF_ARRAY_ELEMS(f_center_34))
-            f_center = f_center_34[k] / 24.;
+            f_center = f_center_34[k] / 24.0;
         else
             f_center = k - 26.5f;
         for (m = 0; m < PS_AP_LINKS; m++) {

libavcodec/acelp_vectors.c

@@ -114,10 +114,10 @@ const float ff_b60_sinc[61] = {
  0.898529  ,  0.865051  ,  0.769257  ,  0.624054  ,  0.448639  ,  0.265289   ,
  0.0959167 , -0.0412598 , -0.134338  , -0.178986  , -0.178528  , -0.142609   ,
 -0.0849304 , -0.0205078 ,  0.0369568 ,  0.0773926 ,  0.0955200 ,  0.0912781  ,
- 0.0689392 ,  0.0357056 ,  0.        , -0.0305481 , -0.0504150 , -0.0570068  ,
+ 0.0689392 ,  0.0357056 ,  0.0       , -0.0305481 , -0.0504150 , -0.0570068  ,
 -0.0508423 , -0.0350037 , -0.0141602 ,  0.00665283,  0.0230713 ,  0.0323486  ,
  0.0335388 ,  0.0275879 ,  0.0167847 ,  0.00411987, -0.00747681, -0.0156860  ,
--0.0193481 , -0.0183716 , -0.0137634 , -0.00704956,  0.        ,  0.00582886 ,
+-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956,  0.0       ,  0.00582886 ,
  0.00939941,  0.0103760 ,  0.00903320,  0.00604248,  0.00238037, -0.00109863 ,
 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
  0.00103760,  0.00222778,  0.00277710,  0.00271606,  0.00213623,  0.00115967 ,

libavcodec/qcelpdec.c

@@ -94,7 +94,7 @@ static av_cold int qcelp_decode_init(AVCodecContext *avctx)
     avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
 
     for (i = 0; i < 10; i++)
-        q->prev_lspf[i] = (i + 1) / 11.;
+        q->prev_lspf[i] = (i + 1) / 11.0;
 
     return 0;
 }
@@ -162,7 +162,7 @@ static int decode_lspf(QCELPContext *q, float *lspf)
     } else {
         q->octave_count = 0;
 
-        tmp_lspf = 0.;
+        tmp_lspf = 0.0;
         for (i = 0; i < 5; i++) {
             lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
             lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
@@ -434,7 +434,7 @@ static const float *do_pitchfilter(float memory[303], const float v_in[160],
             v_lag = memory + 143 + 40 * i - lag[i];
             for (v_len = v_in + 40; v_in < v_len; v_in++) {
                 if (pfrac[i]) { // If it is a fractional lag...
-                    for (j = 0, *v_out = 0.; j < 4; j++)
+                    for (j = 0, *v_out = 0.0; j < 4; j++)
                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
                 } else
                     *v_out = *v_lag;

libavcodec/ra288.c

@@ -95,7 +95,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
     memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
 
     /* block 46 of G.728 spec */
-    sum = 32.;
+    sum = 32.0;
     for (i=0; i < 10; i++)
         sum -= gain_block[9-i] * ractx->gain_lpc[i];
 
@@ -111,7 +111,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
 
     sum = avpriv_scalarproduct_float_c(buffer, buffer, 5);
 
-    sum = FFMAX(sum, 5. / (1<<24));
+    sum = FFMAX(sum, 5.0 / (1<<24));
 
     /* shift and store */
     memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
@@ -157,7 +157,7 @@ static void do_hybrid_window(RA288Context *ractx,
     }
 
     /* Multiply by the white noise correcting factor (WNCF). */
-    *out *= 257./256.;
+    *out *= 257.0 / 256.0;
 }
 
 /**

libavcodec/sipr.c

@@ -240,7 +240,7 @@ static void eval_ir(const float *Az, int pitch_lag, float *freq,
     float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
     int i;
 
-    tmp1[0] = 1.;
+    tmp1[0] = 1.0;
     for (i = 0; i < LP_FILTER_ORDER; i++) {
         tmp1[i+1] = Az[i] * ff_pow_0_55[i];
         tmp2[i  ] = Az[i] * ff_pow_0_7 [i];

libavcodec/twinvq.c

@@ -423,12 +423,12 @@ static inline float mulawinv(float y, float clip, float mu)
  * {
  *    static float test; // Ugh, force gcc to do the division first...
  *
- *    test = a / 400.;
+ *    test = a / 400.0;
  *    return b * test + 0.5;
  * }
  * @endcode
  *
- * @note if this function is replaced by just ROUNDED_DIV(a * b, 400.), the
+ * @note if this function is replaced by just ROUNDED_DIV(a * b, 400.0), the
  * stddev between the original file (before encoding with Yamaha encoder) and
  * the decoded output increases, which leads one to believe that the encoder
  * expects exactly this broken calculation.
@@ -516,12 +516,12 @@ static void dec_gain(TwinContext *tctx, GetBitContext *gb, enum FrameType ftype,
 
     if (ftype == FT_LONG) {
         for (i = 0; i < tctx->avctx->channels; i++)
-            out[i] = (1. / (1 << 13)) *
+            out[i] = (1.0 / (1 << 13)) *
                      mulawinv(step * 0.5 + step * get_bits(gb, GAIN_BITS),
                               AMP_MAX, MULAW_MU);
     } else {
         for (i = 0; i < tctx->avctx->channels; i++) {
-            float val = (1. / (1 << 23)) *
+            float val = (1.0 / (1 << 23)) *
                         mulawinv(step * 0.5 + step * get_bits(gb, GAIN_BITS),
                                  AMP_MAX, MULAW_MU);
 
@@ -582,7 +582,7 @@ static void decode_lsp(TwinContext *tctx, int lpc_idx1, uint8_t *lpc_idx2,
     rearrange_lsp(mtab->n_lsp, lsp, 0.0001);
 
     for (i = 0; i < mtab->n_lsp; i++) {
-        float tmp1 = 1.      - cb3[lpc_hist_idx * mtab->n_lsp + i];
+        float tmp1 = 1.0     - cb3[lpc_hist_idx * mtab->n_lsp + i];
         float tmp2 = hist[i] * cb3[lpc_hist_idx * mtab->n_lsp + i];
         hist[i] = lsp[i];
         lsp[i]  = lsp[i] * tmp1 + tmp2;
@@ -713,13 +713,13 @@ static void dec_bark_env(TwinContext *tctx, const uint8_t *in, int use_hist,
     for (i = 0; i < fw_cb_len; i++)
         for (j = 0; j < bark_n_coef; j++, idx++) {
             float tmp2 = mtab->fmode[ftype].bark_cb[fw_cb_len * in[j] + i] *
-                         (1. / 4096);
-            float st   = use_hist ? (1. - val) * tmp2 + val * hist[idx] + 1.
-                                  : tmp2 + 1.;
+                         (1.0 / 4096);
+            float st   = use_hist ? (1.0 - val) * tmp2 + val * hist[idx] + 1.0
+                                  : tmp2 + 1.0;
 
             hist[idx] = tmp2;
-            if (st < -1.)
-                st = 1.;
+            if (st < -1.0)
+                st = 1.0;
 
             memset_float(out, st * gain, mtab->fmode[ftype].bark_tab[idx]);
             out += mtab->fmode[ftype].bark_tab[idx];
@@ -789,12 +789,12 @@ static void read_and_decode_spectrum(TwinContext *tctx, GetBitContext *gb,
         }
 
         if (ftype == FT_LONG) {
-            float pgain_step = 25000. / ((1 << mtab->pgain_bit) - 1);
+            float pgain_step = 25000.0 / ((1 << mtab->pgain_bit) - 1);
             int p_coef       = get_bits(gb, tctx->mtab->ppc_period_bit);
             int g_coef       = get_bits(gb, tctx->mtab->pgain_bit);
-            float v          = 1. / 8192 *
+            float v          = 1.0 / 8192 *
                                mulawinv(pgain_step * g_coef + pgain_step / 2,
-                                        25000., PGAIN_MU);
+                                        25000.0, PGAIN_MU);
 
             decode_ppc(tctx, p_coef, ppc_shape + i * mtab->ppc_shape_len, v,
                        chunk);
@@ -881,7 +881,7 @@ static av_cold int init_mdct_win(TwinContext *tctx)
     int size_s          = mtab->size / mtab->fmode[FT_SHORT].sub;
     int size_m          = mtab->size / mtab->fmode[FT_MEDIUM].sub;
     int channels        = tctx->avctx->channels;
-    float norm          = channels == 1 ? 2. : 1.;
+    float norm          = channels == 1 ? 2.0 : 1.0;
 
     for (i = 0; i < 3; i++) {
         int bsize = tctx->mtab->size / tctx->mtab->fmode[i].sub;

libavcodec/vorbisenc.c

@@ -189,7 +189,7 @@ static int ready_codebook(vorbis_enc_codebook *cb)
                 cb->pow2[i] += cb->dimensions[i * cb->ndimensions + j] * cb->dimensions[i * cb->ndimensions + j];
                 div *= vals;
             }
-            cb->pow2[i] /= 2.;
+            cb->pow2[i] /= 2.0;
         }
     }
     return 0;
@@ -728,7 +728,7 @@ static void floor_fit(vorbis_enc_context *venc, vorbis_enc_floor *fc,
 {
     int range = 255 / fc->multiplier + 1;
     int i;
-    float tot_average = 0.;
+    float tot_average = 0.0;
     float averages[MAX_FLOOR_VALUES];
     for (i = 0; i < fc->values; i++) {
         averages[i] = get_floor_average(fc, coeffs, i);
@@ -881,7 +881,7 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
     assert(rc->type == 2);
     assert(real_ch == 2);
     for (p = 0; p < partitions; p++) {
-        float max1 = 0., max2 = 0.;
+        float max1 = 0.0, max2 = 0.0;
         int s = rc->begin + p * psize;
         for (k = s; k < s + psize; k += 2) {
             max1 = FFMAX(max1, fabs(coeffs[          k / real_ch]));
@@ -968,7 +968,7 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
     int i, channel;
     const float * win = venc->win[0];
     int window_len = 1 << (venc->log2_blocksize[0] - 1);
-    float n = (float)(1 << venc->log2_blocksize[0]) / 4.;
+    float n = (float)(1 << venc->log2_blocksize[0]) / 4.0;
     // FIXME use dsp
 
     if (!venc->have_saved && !samples)

libavformat/swfenc.c

@@ -234,7 +234,7 @@ static int swf_write_header(AVFormatContext *s)
     }
 
     if (!swf->audio_enc)
-        swf->samples_per_frame = (44100. * rate_base) / rate;
+        swf->samples_per_frame = (44100.0 * rate_base) / rate;
     else
         swf->samples_per_frame = (swf->audio_enc->sample_rate * rate_base) / rate;