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Code Issues Releases Activity

use av_clip_int16() where it makes sense

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Originally committed as revision 10078 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Aurelien Jacobs 2007-08-11 22:48:55 +00:00
parent 160ab30fcc
commit aee481cebe
14 changed files with 34 additions and 64 deletions
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36
libavcodec/adpcm.c
View File

@ -50,12 +50,6 @@
#define BLKSIZE 1024
#define CLAMP_TO_SHORT(value) \
if (value > 32767) \
value = 32767; \
else if (value < -32768) \
value = -32768; \
/* step_table[] and index_table[] are from the ADPCM reference source */
/* This is the index table: */
static const int index_table[16] = {
@ -215,7 +209,7 @@ static inline unsigned char adpcm_ima_compress_sample(ADPCMChannelStatus *c, sho
int delta = sample - c->prev_sample;
int nibble = FFMIN(7, abs(delta)*4/step_table[c->step_index]) + (delta<0)*8;
c->prev_sample = c->prev_sample + ((step_table[c->step_index] * yamaha_difflookup[nibble]) / 8);
CLAMP_TO_SHORT(c->prev_sample);
c->prev_sample = av_clip_int16(c->prev_sample);
c->step_index = av_clip(c->step_index + index_table[nibble], 0, 88);
return nibble;
}
@ -234,7 +228,7 @@ static inline unsigned char adpcm_ms_compress_sample(ADPCMChannelStatus *c, shor
nibble= av_clip(nibble, -8, 7)&0x0F;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
CLAMP_TO_SHORT(predictor);
predictor = av_clip_int16(predictor);
c->sample2 = c->sample1;
c->sample1 = predictor;
@ -259,7 +253,7 @@ static inline unsigned char adpcm_yamaha_compress_sample(ADPCMChannelStatus *c,
nibble = FFMIN(7, abs(delta)*4/c->step) + (delta<0)*8;
c->predictor = c->predictor + ((c->step * yamaha_difflookup[nibble]) / 8);
CLAMP_TO_SHORT(c->predictor);
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
@ -339,7 +333,7 @@ static void adpcm_compress_trellis(AVCodecContext *avctx, const short *samples,
#define STORE_NODE(NAME, STEP_INDEX)\
int d;\
uint32_t ssd;\
CLAMP_TO_SHORT(dec_sample);\
dec_sample = av_clip_int16(dec_sample);\
d = sample - dec_sample;\
ssd = nodes[j]->ssd + d*d;\
if(nodes_next[frontier-1] && ssd >= nodes_next[frontier-1]->ssd)\
@ -676,7 +670,7 @@ static inline short adpcm_ima_expand_nibble(ADPCMChannelStatus *c, char nibble,
if (sign) predictor -= diff;
else predictor += diff;
CLAMP_TO_SHORT(predictor);
predictor = av_clip_int16(predictor);
c->predictor = predictor;
c->step_index = step_index;
@ -689,7 +683,7 @@ static inline short adpcm_ms_expand_nibble(ADPCMChannelStatus *c, char nibble)
predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 256;
predictor += (signed)((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;
CLAMP_TO_SHORT(predictor);
predictor = av_clip_int16(predictor);
c->sample2 = c->sample1;
c->sample1 = predictor;
@ -725,7 +719,7 @@ static inline short adpcm_ct_expand_nibble(ADPCMChannelStatus *c, char nibble)
if(c->step > 32767)
c->step = 32767;
CLAMP_TO_SHORT(predictor);
predictor = av_clip_int16(predictor);
c->predictor = predictor;
return (short)predictor;
}
@ -766,7 +760,7 @@ static inline short adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, unsigned c
}
c->predictor += (c->step * yamaha_difflookup[nibble]) / 8;
CLAMP_TO_SHORT(c->predictor);
c->predictor = av_clip_int16(c->predictor);
c->step = (c->step * yamaha_indexscale[nibble]) >> 8;
c->step = av_clip(c->step, 127, 24567);
return c->predictor;
@ -795,7 +789,7 @@ static void xa_decode(short *out, const unsigned char *in,
t = (signed char)(d<<4)>>4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
CLAMP_TO_SHORT(s);
s = av_clip_int16(s);
*out = s;
out += inc;
s_2 = s_1;
@ -821,7 +815,7 @@ static void xa_decode(short *out, const unsigned char *in,
t = (signed char)d >> 4;
s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
CLAMP_TO_SHORT(s);
s = av_clip_int16(s);
*out = s;
out += inc;
s_2 = s_1;
@ -915,7 +909,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
if(cs->predictor & 0x8000)
cs->predictor -= 0x10000;
CLAMP_TO_SHORT(cs->predictor);
cs->predictor = av_clip_int16(cs->predictor);
cs->step_index = (*src++) & 0x7F;
@ -1187,8 +1181,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
next_right_sample = (next_right_sample +
(current_right_sample * coeff1r) +
(previous_right_sample * coeff2r) + 0x80) >> 8;
CLAMP_TO_SHORT(next_left_sample);
CLAMP_TO_SHORT(next_right_sample);
next_left_sample = av_clip_int16(next_left_sample);
next_right_sample = av_clip_int16(next_right_sample);
previous_left_sample = current_left_sample;
current_left_sample = next_left_sample;
@ -1318,7 +1312,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
c->status[i].step_index += table[delta & (~signmask)];
c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
c->status[i].predictor = av_clip(c->status[i].predictor, -32768, 32767);
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
*samples++ = c->status[i].predictor;
if (samples >= samples_end) {
@ -1392,7 +1386,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
sampledat = ((prev[ch][0]*factor1
+ prev[ch][1]*factor2) >> 11) + (sampledat>>exp);
CLAMP_TO_SHORT(sampledat);
sampledat = av_clip_int16(sampledat);
*samples = sampledat;
prev[ch][1] = prev[ch][0];
prev[ch][0] = *samples++;

6
libavcodec/adx.c
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@ -46,8 +46,6 @@ typedef struct {
#define SCALE1 0x7298
#define SCALE2 0x3350
#define CLIP(s) if (s>32767) s=32767; else if (s<-32768) s=-32768
/* 18 bytes <-> 32 samples */
#ifdef CONFIG_ENCODERS
@ -110,7 +108,7 @@ static void adx_decode(short *out,const unsigned char *in,PREV *prev)
// d>>=4; if (d&8) d-=16;
d = ((signed char)d >> 4);
s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
CLIP(s0);
s0 = av_clip_int16(s0);
*out++=s0;
s2 = s1;
s1 = s0;
@ -119,7 +117,7 @@ static void adx_decode(short *out,const unsigned char *in,PREV *prev)
//d&=15; if (d&8) d-=16;
d = ((signed char)(d<<4) >> 4);
s0 = (BASEVOL*d*scale + SCALE1*s1 - SCALE2*s2)>>14;
CLIP(s0);
s0 = av_clip_int16(s0);
*out++=s0;
s2 = s1;
s1 = s0;

6
libavcodec/atrac3.c
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@ -895,13 +895,13 @@ static int atrac3_decode_frame(AVCodecContext *avctx,
if (q->channels == 1) {
/* mono */
for (i = 0; i<1024; i++)
samples[i] = av_clip(round(q->outSamples[i]), -32768, 32767);
samples[i] = av_clip_int16(round(q->outSamples[i]));
*data_size = 1024 * sizeof(int16_t);
} else {
/* stereo */
for (i = 0; i < 1024; i++) {
samples[i*2] = av_clip(round(q->outSamples[i]), -32768, 32767);
samples[i*2+1] = av_clip(round(q->outSamples[1024+i]), -32768, 32767);
samples[i*2] = av_clip_int16(round(q->outSamples[i]));
samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
}
*data_size = 2048 * sizeof(int16_t);
}

2
libavcodec/cook.c
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@ -906,7 +906,7 @@ saturate_output_float (COOKContext *q, int chan, int16_t *out)
*/
for (j = 0; j < q->samples_per_channel; j++) {
out[chan + q->nb_channels * j] =
av_clip(lrintf(output[j]), -32768, 32767);
av_clip_int16(lrintf(output[j]));
}
}

10
libavcodec/dpcm.c
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@ -46,8 +46,6 @@ typedef struct DPCMContext {
const int *sol_table;//for SOL_DPCM
} DPCMContext;
#define SATURATE_S16(x) if (x < -32768) x = -32768; \
else if (x > 32767) x = 32767;
#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
static int interplay_delta_table[] = {
@ -190,7 +188,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx,
/* decode the samples */
for (in = 8, out = 0; in < buf_size; in++, out++) {
predictor[channel_number] += s->roq_square_array[buf[in]];
SATURATE_S16(predictor[channel_number]);
predictor[channel_number] = av_clip_int16(predictor[channel_number]);
output_samples[out] = predictor[channel_number];
/* toggle channel */
@ -213,7 +211,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx,
while (in < buf_size) {
predictor[channel_number] += interplay_delta_table[buf[in++]];
SATURATE_S16(predictor[channel_number]);
predictor[channel_number] = av_clip_int16(predictor[channel_number]);
output_samples[out++] = predictor[channel_number];
/* toggle channel */
@ -248,7 +246,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx,
diff >>= shift[channel_number];
predictor[channel_number] += diff;
SATURATE_S16(predictor[channel_number]);
predictor[channel_number] = av_clip_int16(predictor[channel_number]);
output_samples[out++] = predictor[channel_number];
/* toggle channel */
@ -277,7 +275,7 @@ static int dpcm_decode_frame(AVCodecContext *avctx,
n = buf[in++];
if (n & 0x80) s->sample[channel_number] -= s->sol_table[n & 0x7F];
else s->sample[channel_number] += s->sol_table[n & 0x7F];
SATURATE_S16(s->sample[channel_number]);
s->sample[channel_number] = av_clip_int16(s->sample[channel_number]);
output_samples[out++] = s->sample[channel_number];
/* toggle channel */
channel_number ^= s->channels - 1;

2
libavcodec/dsicinav.c
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@ -325,7 +325,7 @@ static int cinaudio_decode_frame(AVCodecContext *avctx,
}
while (buf_size > 0) {
cin->delta += cinaudio_delta16_table[*src++];
cin->delta = av_clip(cin->delta, -32768, 32767);
cin->delta = av_clip_int16(cin->delta);
*samples++ = cin->delta;
--buf_size;
}

6
libavcodec/liba52.c
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@ -123,11 +123,7 @@ static int a52_decode_init(AVCodecContext *avctx)
/**** the following two functions comes from a52dec */
static inline int blah (int32_t i)
{
if (i > 0x43c07fff)
return 32767;
else if (i < 0x43bf8000)
return -32768;
return i - 0x43c00000;
return av_clip_int16(i - 0x43c00000);
}
static inline void float_to_int (float * _f, int16_t * s16, int nchannels)

3
libavcodec/libvorbis.c
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@ -307,8 +307,7 @@ static inline int conv(int samples, float **pcm, char *buf, int channels) {
val = mono[j] * 32767.f;
if(val > 32767) val = 32767 ;
if(val < -32768) val = -32768 ;
val = av_clip_int16(val);
*ptr = val ;
ptr += channels;

5
libavcodec/mpegaudiodec.c
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@ -822,10 +822,7 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
#if FRAC_BITS <= 15
/* NOTE: can cause a loss in precision if very high amplitude
sound */
if (v > 32767)
v = 32767;
else if (v < -32768)
v = -32768;
v = av_clip_int16(v);
#endif
synth_buf[j] = v;
}

4
libavcodec/ra144.c
View File

@ -486,9 +486,7 @@ static int ra144_decode_frame(AVCodecContext * avctx,
shptr=glob->output_buffer;
while (shptr<glob->output_buffer+BLOCKSIZE) {
s=*(shptr++)<<2;
*data=s;
if (s>32767) *data=32767;
if (s<-32767) *data=-32768;
*data=av_clip_int16(s);
data++;
}
b+=30;

2
libavcodec/resample2.c
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@ -279,7 +279,7 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
dst[dst_index] = av_clip(lrintf(val), -32768, 32767);
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;

9
libavcodec/sonic.c
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@ -926,14 +926,7 @@ static int sonic_decode_frame(AVCodecContext *avctx,
// internal -> short
for (i = 0; i < s->frame_size; i++)
{
if (s->int_samples[i] > 32767)
samples[i] = 32767;
else if (s->int_samples[i] < -32768)
samples[i] = -32768;
else
samples[i] = s->int_samples[i];
}
samples[i] = av_clip_int16(s->int_samples[i]);
align_get_bits(&gb);

2
libavcodec/vmdav.c
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@ -458,7 +458,7 @@ static void vmdaudio_decode_audio(VmdAudioContext *s, unsigned char *data,
s->predictors[chan] -= vmdaudio_table[buf[i] & 0x7F];
else
s->predictors[chan] += vmdaudio_table[buf[i]];
s->predictors[chan] = av_clip(s->predictors[chan], -32768, 32767);
s->predictors[chan] = av_clip_int16(s->predictors[chan]);
out[i] = s->predictors[chan];
chan ^= stereo;
}

5
libavcodec/wmadec.c
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@ -740,10 +740,7 @@ static int wma_decode_frame(WMACodecContext *s, int16_t *samples)
for(i=0;i<n;i++) {
a = lrintf(*iptr++);
if (a > 32767)
a = 32767;
else if (a < -32768)
a = -32768;
a = av_clip_int16(a);
*ptr = a;
ptr += incr;
}