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-rw-r--r--libavcodec/mpegaudio.c106
1 files changed, 53 insertions, 53 deletions
diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c
index 7a0b0a31ce..c673ebc67c 100644
--- a/libavcodec/mpegaudio.c
+++ b/libavcodec/mpegaudio.c
@@ -16,12 +16,12 @@
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-
+
/**
* @file mpegaudio.c
* The simplest mpeg audio layer 2 encoder.
*/
-
+
#include "avcodec.h"
#include "bitstream.h"
#include "mpegaudio.h"
@@ -49,7 +49,7 @@ typedef struct MpegAudioContext {
int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
/* code to group 3 scale factors */
- unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
+ unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
} MpegAudioContext;
@@ -79,7 +79,7 @@ static int MPA_encode_init(AVCodecContext *avctx)
/* encoding freq */
s->lsf = 0;
for(i=0;i<3;i++) {
- if (mpa_freq_tab[i] == freq)
+ if (mpa_freq_tab[i] == freq)
break;
if ((mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
@@ -94,7 +94,7 @@ static int MPA_encode_init(AVCodecContext *avctx)
/* encoding bitrate & frequency */
for(i=0;i<15;i++) {
- if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
+ if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
if (i == 15){
@@ -104,14 +104,14 @@ static int MPA_encode_init(AVCodecContext *avctx)
s->bitrate_index = i;
/* compute total header size & pad bit */
-
+
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
s->frame_size = ((int)a) * 8;
/* frame fractional size to compute padding */
s->frame_frac = 0;
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
-
+
/* select the right allocation table */
table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
@@ -120,7 +120,7 @@ static int MPA_encode_init(AVCodecContext *avctx)
s->alloc_table = alloc_tables[table];
#ifdef DEBUG
- av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
+ av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
bitrate, freq, s->frame_size, table, s->frame_frac_incr);
#endif
@@ -163,14 +163,14 @@ static int MPA_encode_init(AVCodecContext *avctx)
v = 2;
else if (v < 3)
v = 3;
- else
+ else
v = 4;
scale_diff_table[i] = v;
}
for(i=0;i<17;i++) {
v = quant_bits[i];
- if (v < 0)
+ if (v < 0)
v = -v;
else
v = v * 3;
@@ -191,7 +191,7 @@ static void idct32(int *out, int *tab)
const int *xp = costab32;
for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
-
+
t = tab + 30;
t1 = tab + 2;
do {
@@ -209,30 +209,30 @@ static void idct32(int *out, int *tab)
t[3] += t[3-8];
t -= 8;
} while (t != t1);
-
+
t = tab;
t1 = tab + 32;
do {
- t[ 3] = -t[ 3];
- t[ 6] = -t[ 6];
-
- t[11] = -t[11];
- t[12] = -t[12];
- t[13] = -t[13];
- t[15] = -t[15];
+ t[ 3] = -t[ 3];
+ t[ 6] = -t[ 6];
+
+ t[11] = -t[11];
+ t[12] = -t[12];
+ t[13] = -t[13];
+ t[15] = -t[15];
t += 16;
} while (t != t1);
-
+
t = tab;
t1 = tab + 8;
do {
int x1, x2, x3, x4;
-
+
x3 = MUL(t[16], FIX(SQRT2*0.5));
x4 = t[0] - x3;
x3 = t[0] + x3;
-
+
x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
x1 = MUL((t[8] - x2), xp[0]);
x2 = MUL((t[8] + x2), xp[1]);
@@ -255,11 +255,11 @@ static void idct32(int *out, int *tab)
xr = MUL(t[4],xp[1]);
t[ 4] = (t[24] - xr);
t[24] = (t[24] + xr);
-
+
xr = MUL(t[20],xp[2]);
t[20] = (t[8] - xr);
t[ 8] = (t[8] + xr);
-
+
xr = MUL(t[12],xp[3]);
t[12] = (t[16] - xr);
t[16] = (t[16] + xr);
@@ -271,19 +271,19 @@ static void idct32(int *out, int *tab)
xr = MUL(tab[30-i*4],xp[0]);
tab[30-i*4] = (tab[i*4] - xr);
tab[ i*4] = (tab[i*4] + xr);
-
+
xr = MUL(tab[ 2+i*4],xp[1]);
tab[ 2+i*4] = (tab[28-i*4] - xr);
tab[28-i*4] = (tab[28-i*4] + xr);
-
+
xr = MUL(tab[31-i*4],xp[0]);
tab[31-i*4] = (tab[1+i*4] - xr);
tab[ 1+i*4] = (tab[1+i*4] + xr);
-
+
xr = MUL(tab[ 3+i*4],xp[1]);
tab[ 3+i*4] = (tab[29-i*4] - xr);
tab[29-i*4] = (tab[29-i*4] + xr);
-
+
xp += 2;
}
@@ -352,7 +352,7 @@ static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
out += 32;
/* handle the wrap around */
if (offset < 0) {
- memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
+ memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
s->samples_buf[ch], (512 - 32) * 2);
offset = SAMPLES_BUF_SIZE - 512;
}
@@ -363,14 +363,14 @@ static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
}
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
- unsigned char scale_factors[SBLIMIT][3],
+ unsigned char scale_factors[SBLIMIT][3],
int sb_samples[3][12][SBLIMIT],
int sblimit)
{
int *p, vmax, v, n, i, j, k, code;
int index, d1, d2;
unsigned char *sf = &scale_factors[0][0];
-
+
for(j=0;j<sblimit;j++) {
for(i=0;i<3;i++) {
/* find the max absolute value */
@@ -385,7 +385,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
/* compute the scale factor index using log 2 computations */
if (vmax > 0) {
n = av_log2(vmax);
- /* n is the position of the MSB of vmax. now
+ /* n is the position of the MSB of vmax. now
use at most 2 compares to find the index */
index = (21 - n) * 3 - 3;
if (index >= 0) {
@@ -399,7 +399,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
}
#if 0
- printf("%2d:%d in=%x %x %d\n",
+ printf("%2d:%d in=%x %x %d\n",
j, i, vmax, scale_factor_table[index], index);
#endif
/* store the scale factor */
@@ -411,7 +411,7 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
are close enough to each other */
d1 = scale_diff_table[sf[0] - sf[1] + 64];
d2 = scale_diff_table[sf[1] - sf[2] + 64];
-
+
/* handle the 25 cases */
switch(d1 * 5 + d2) {
case 0*5+0:
@@ -468,9 +468,9 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
assert(0); //cant happen
code = 0; /* kill warning */
}
-
+
#if 0
- printf("%d: %2d %2d %2d %d %d -> %d\n", j,
+ printf("%d: %2d %2d %2d %d %d -> %d\n", j,
sf[0], sf[1], sf[2], d1, d2, code);
#endif
scale_code[j] = code;
@@ -498,7 +498,7 @@ static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
/* Try to maximize the smr while using a number of bits inferior to
the frame size. I tried to make the code simpler, faster and
smaller than other encoders :-) */
-static void compute_bit_allocation(MpegAudioContext *s,
+static void compute_bit_allocation(MpegAudioContext *s,
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
int *padding)
@@ -512,7 +512,7 @@ static void compute_bit_allocation(MpegAudioContext *s,
memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
-
+
/* compute frame size and padding */
max_frame_size = s->frame_size;
s->frame_frac += s->frame_frac_incr;
@@ -547,13 +547,13 @@ static void compute_bit_allocation(MpegAudioContext *s,
}
}
#if 0
- printf("current=%d max=%d max_sb=%d alloc=%d\n",
+ printf("current=%d max=%d max_sb=%d alloc=%d\n",
current_frame_size, max_frame_size, max_sb,
bit_alloc[max_sb]);
-#endif
+#endif
if (max_sb < 0)
break;
-
+
/* find alloc table entry (XXX: not optimal, should use
pointer table) */
alloc = s->alloc_table;
@@ -568,7 +568,7 @@ static void compute_bit_allocation(MpegAudioContext *s,
} else {
/* increments bit allocation */
b = bit_alloc[max_ch][max_sb];
- incr = total_quant_bits[alloc[b + 1]] -
+ incr = total_quant_bits[alloc[b + 1]] -
total_quant_bits[alloc[b]];
}
@@ -637,11 +637,11 @@ static void encode_frame(MpegAudioContext *s,
}
j += 1 << bit_alloc_bits;
}
-
+
/* scale codes */
for(i=0;i<s->sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
- if (bit_alloc[ch][i])
+ if (bit_alloc[ch][i])
put_bits(p, 2, s->scale_code[ch][i]);
}
}
@@ -669,7 +669,7 @@ static void encode_frame(MpegAudioContext *s,
}
}
}
-
+
/* quantization & write sub band samples */
for(k=0;k<3;k++) {
@@ -699,7 +699,7 @@ static void encode_frame(MpegAudioContext *s,
e = s->scale_factors[ch][i][k];
shift = scale_factor_shift[e];
mult = scale_factor_mult[e];
-
+
/* normalize to P bits */
if (shift < 0)
q1 = sample << (-shift);
@@ -716,17 +716,17 @@ static void encode_frame(MpegAudioContext *s,
bits = quant_bits[qindex];
if (bits < 0) {
/* group the 3 values to save bits */
- put_bits(p, -bits,
+ put_bits(p, -bits,
q[0] + steps * (q[1] + steps * q[2]));
#if 0
- printf("%d: gr1 %d\n",
+ printf("%d: gr1 %d\n",
i, q[0] + steps * (q[1] + steps * q[2]));
#endif
} else {
#if 0
- printf("%d: gr3 %d %d %d\n",
+ printf("%d: gr3 %d %d %d\n",
i, q[0], q[1], q[2]);
-#endif
+#endif
put_bits(p, bits, q[0]);
put_bits(p, bits, q[1]);
put_bits(p, bits, q[2]);
@@ -734,7 +734,7 @@ static void encode_frame(MpegAudioContext *s,
}
}
/* next subband in alloc table */
- j += 1 << bit_alloc_bits;
+ j += 1 << bit_alloc_bits;
}
}
}
@@ -761,7 +761,7 @@ static int MPA_encode_frame(AVCodecContext *avctx,
}
for(i=0;i<s->nb_channels;i++) {
- compute_scale_factors(s->scale_code[i], s->scale_factors[i],
+ compute_scale_factors(s->scale_code[i], s->scale_factors[i],
s->sb_samples[i], s->sblimit);
}
for(i=0;i<s->nb_channels;i++) {
@@ -772,7 +772,7 @@ static int MPA_encode_frame(AVCodecContext *avctx,
init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
encode_frame(s, bit_alloc, padding);
-
+
s->nb_samples += MPA_FRAME_SIZE;
return pbBufPtr(&s->pb) - s->pb.buf;
}
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