Chiebot-Mirror
/
FFmpeg
mirror of https://github.com/FFmpeg/FFmpeg.git
8
0
Fork 0
Code Issues Projects Releases Wiki Activity

use av_clip_int16() where it makes sense

Browse Source 
Originally committed as revision 10078 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Aurelien Jacobs 18 years ago
parent 160ab30fcc
commit aee481cebe
14 changed files with 34 additions and 64 deletions
Whitespace
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. 36
      libavcodec/adpcm.c
  2. 6
      libavcodec/adx.c
  3. 6
      libavcodec/atrac3.c
  4. 2
      libavcodec/cook.c
  5. 10
      libavcodec/dpcm.c
  6. 2
      libavcodec/dsicinav.c
  7. 6
      libavcodec/liba52.c
  8. 3
      libavcodec/libvorbis.c
  9. 5
      libavcodec/mpegaudiodec.c
  10. 4
      libavcodec/ra144.c
  11. 2
      libavcodec/resample2.c
  12. 9
      libavcodec/sonic.c
  13. 2
      libavcodec/vmdav.c
  14. 5
      libavcodec/wmadec.c

36
libavcodec/adpcm.c
Unescape 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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape 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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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
Unescape View File

@ -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;
}

Write Preview
Loading…
Cancel
Save