[FFmpeg-cvslog] cosmetics: Add '0' to float constants ending in '.'.
Diego Biurrun
git at videolan.org
Fri Jul 26 10:49:55 CEST 2013
ffmpeg | branch: master | Diego Biurrun <diego at biurrun.de> | Tue Jul 23 23:48:45 2013 +0200| [4a2ef39442bf7f0150db07a1fbfcf8286e4d44a3] | committer: Diego Biurrun
cosmetics: Add '0' to float constants ending in '.'.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4a2ef39442bf7f0150db07a1fbfcf8286e4d44a3
---
libavcodec/aac_tablegen.h | 2 +-
libavcodec/aacdec.c | 4 ++--
libavcodec/aacps_tablegen.h | 2 +-
libavcodec/acelp_vectors.c | 4 ++--
libavcodec/qcelpdec.c | 6 +++---
libavcodec/ra288.c | 6 +++---
libavcodec/sipr.c | 2 +-
libavcodec/twinvq.c | 28 ++++++++++++++--------------
libavcodec/vorbisenc.c | 8 ++++----
libavformat/swfenc.c | 2 +-
10 files changed, 32 insertions(+), 32 deletions(-)
diff --git a/libavcodec/aac_tablegen.h b/libavcodec/aac_tablegen.h
index a45de9a..8a05ec5 100644
--- a/libavcodec/aac_tablegen.h
+++ b/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 */
diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
index e44bb5a..659be55 100644
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
@@ -1172,7 +1172,7 @@ 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++) {
@@ -1916,7 +1916,7 @@ 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;
diff --git a/libavcodec/aacps_tablegen.h b/libavcodec/aacps_tablegen.h
index 0c610ed..701812b 100644
--- a/libavcodec/aacps_tablegen.h
+++ b/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++) {
diff --git a/libavcodec/acelp_vectors.c b/libavcodec/acelp_vectors.c
index a85e45f..0c660ac 100644
--- a/libavcodec/acelp_vectors.c
+++ b/libavcodec/acelp_vectors.c
@@ -94,10 +94,10 @@ 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 ,
diff --git a/libavcodec/qcelpdec.c b/libavcodec/qcelpdec.c
index ead7d90..3772e26 100644
--- a/libavcodec/qcelpdec.c
+++ b/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;
diff --git a/libavcodec/ra288.c b/libavcodec/ra288.c
index 5cccbe9..0b750ac 100644
--- a/libavcodec/ra288.c
+++ b/libavcodec/ra288.c
@@ -90,7 +90,7 @@ 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];
@@ -104,7 +104,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
- sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.);
+ sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);
sum = FFMAX(sum, 1);
@@ -150,7 +150,7 @@ static void do_hybrid_window(RA288Context *ractx,
}
/* Multiply by the white noise correcting factor (WNCF). */
- *out *= 257./256.;
+ *out *= 257.0 / 256.0;
}
/**
diff --git a/libavcodec/sipr.c b/libavcodec/sipr.c
index 2d35841..83a4217 100644
--- a/libavcodec/sipr.c
+++ b/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];
diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c
index 7b56140..887a88c 100644
--- a/libavcodec/twinvq.c
+++ b/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);
@@ -883,7 +883,7 @@ 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;
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index db0394a..e47f996 100644
--- a/libavcodec/vorbisenc.c
+++ b/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)
diff --git a/libavformat/swfenc.c b/libavformat/swfenc.c
index 31f405d..93487cc 100644
--- a/libavformat/swfenc.c
+++ b/libavformat/swfenc.c
@@ -229,7 +229,7 @@ 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;
More information about the ffmpeg-cvslog
mailing list