/third_party/ffmpeg/libavcodec/ |
D | cfhdencdsp.c | 32 low[(0>>1) * low_stride] = av_clip_int16(input[0*in_stride] + input[1*in_stride]); in filter() 33 high[(0>>1) * high_stride] = av_clip_int16((5 * input[0*in_stride] - 11 * input[1*in_stride] + in filter() 38 low[(i>>1) * low_stride] = av_clip_int16(input[i*in_stride] + input[(i+1)*in_stride]); in filter() 39 … high[(i>>1) * high_stride] = av_clip_int16(((-input[(i-2)*in_stride] - input[(i-1)*in_stride] + in filter() 44 …low[((len-2)>>1) * low_stride] = av_clip_int16(input[((len-2)+0)*in_stride] + input[((len-2)+1)*… in filter() 45 …high[((len-2)>>1) * high_stride] = av_clip_int16((11* input[((len-2)+0)*in_stride] - 5 * input[((l… in filter()
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D | dss_sp.c | 454 coeffs[i] = av_clip_int16(tmp); in dss_sp_convert_coeffs() 457 coeffs[a_plus - i] = av_clip_int16(tmp); in dss_sp_convert_coeffs() 490 vector[i] = av_clip_int16(tmp); in dss_sp_gen_exc() 537 dst[a] = av_clip_int16(tmp); in dss_sp_shift_sq_sub() 559 dst[a] = av_clip_int16(tmp); in dss_sp_shift_sq_add() 635 p->vector_buf[i] = av_clip_int16(tmp); in dss_sp_sf_synthesis() 640 p->vector_buf[0] = av_clip_int16(tmp); in dss_sp_sf_synthesis() 656 noise[0] = av_clip_int16(tmp); in dss_sp_sf_synthesis() 660 noise[i] = av_clip_int16(tmp); in dss_sp_sf_synthesis() 666 dst[i] = av_clip_int16(tmp); in dss_sp_sf_synthesis() [all …]
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D | dpcm.c | 279 predictor[ch] = av_clip_int16(predictor[ch]); in dpcm_decode_frame() 298 predictor[ch] = av_clip_int16(predictor[ch]); in dpcm_decode_frame() 330 predictor[ch] = av_clip_int16(predictor[ch]); in dpcm_decode_frame() 358 s->sample[ch] = av_clip_int16(s->sample[ch]); in dpcm_decode_frame() 373 s->sample[ch] = av_clip_int16(s->sample[ch]); in dpcm_decode_frame() 399 s->sample[idx] = av_clip_int16(s->sample[idx]); in dpcm_decode_frame()
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D | bmvaudio.c | 71 *output_samples++ = av_clip_int16((scale[0] * (int8_t)*buf++) >> 5); in bmv_aud_decode_frame() 72 *output_samples++ = av_clip_int16((scale[1] * (int8_t)*buf++) >> 5); in bmv_aud_decode_frame()
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D | adpcm.c | 298 c->predictor = av_clip_int16(predictor); in adpcm_ima_expand_nibble() 321 c->predictor = av_clip_int16(predictor); in adpcm_ima_alp_expand_nibble() 337 c->predictor = av_clip_int16(predictor >> 4); in adpcm_ima_mtf_expand_nibble() 357 c->predictor = av_clip_int16(predictor); in adpcm_ima_cunning_expand_nibble() 380 c->predictor = av_clip_int16(predictor); in adpcm_ima_wav_expand_nibble() 406 c->predictor = av_clip_int16(predictor); in adpcm_ima_qt_expand_nibble() 420 c->sample1 = av_clip_int16(predictor); in adpcm_ms_expand_nibble() 465 c->predictor = av_clip_int16(c->predictor); in adpcm_ct_expand_nibble() 501 c->predictor = av_clip_int16(c->predictor); in adpcm_yamaha_expand_nibble() 510 c->predictor = av_clip_int16(c->predictor); in adpcm_mtaf_expand_nibble() [all …]
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D | acelp_filters.c | 72 if (av_clip_int16(v >> 15) != (v >> 15)) in ff_acelp_interpolate() 112 out[i] = av_clip_int16((tmp + 0x800) >> 12); in ff_acelp_high_pass_filter()
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D | g722.c | 129 cur_qtzd_reconst = av_clip_int16((band->s_predictor + cur_diff) * 2); in do_adaptive_prediction() 130 band->s_predictor = av_clip_int16(band->s_zero + in do_adaptive_prediction()
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D | g722dec.c | 128 *out_buf++ = av_clip_int16(xout[0] >> 11); in g722_decode_frame() 129 *out_buf++ = av_clip_int16(xout[1] >> 11); in g722_decode_frame()
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D | g729postfilter.c | 496 ga = (fact << 16) / av_clip_int16(32768 - FFABS(gt)); in apply_tilt_comp() 603 gain = av_clip_int16(gain); in ff_g729_adaptive_gain_control() 611 gain_prev = av_clip_int16(gain + gain_prev); in ff_g729_adaptive_gain_control() 612 speech[n] = av_clip_int16((speech[n] * gain_prev + 0x2000) >> 14); in ff_g729_adaptive_gain_control()
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D | gsmdec_template.c | 116 msr = av_clip_int16(data[i] + gsm_mult(msr, 28180)); in postprocess() 117 data[i] = av_clip_int16(msr * 2) & ~7; in postprocess()
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D | dsicinaudio.c | 114 delta = av_clip_int16(delta); in cinaudio_decode_frame()
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D | g723_1dec.c | 417 ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15); in comp_ppf_gains() 644 buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) + in gain_scale() 913 vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] + in generate_noise() 1001 int v = av_clip_int16(vector_ptr[j] * 2); in g723_1_decode_frame() 1002 vector_ptr[j] = av_clip_int16(v + acb_vector[j]); in g723_1_decode_frame() 1087 out[i] = av_clip_int16(2 * p->audio[LPC_ORDER + i]); in g723_1_decode_frame()
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D | adpcmenc.c | 222 c->prev_sample = av_clip_int16(c->prev_sample); in adpcm_ima_compress_sample() 241 c->prev_sample = av_clip_int16(c->prev_sample); in adpcm_ima_alp_compress_sample() 277 c->prev_sample = av_clip_int16(c->prev_sample); in adpcm_ima_qt_compress_sample() 303 c->sample1 = av_clip_int16(predictor); in adpcm_ms_compress_sample() 327 c->predictor = av_clip_int16(c->predictor); in adpcm_yamaha_compress_sample() 402 dec_sample = av_clip_int16(dec_sample);\ in adpcm_compress_trellis()
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/third_party/ffmpeg/libavfilter/ |
D | colorspacedsp_template.c | 89 rgb0[x << SS_W] = av_clip_int16((y00 * cy + crv * v + rnd) >> sh); in fn() 91 rgb0[2 * x + 1] = av_clip_int16((y01 * cy + crv * v + rnd) >> sh); in fn() 93 rgb0[2 * x + rgb_stride] = av_clip_int16((y10 * cy + crv * v + rnd) >> sh); in fn() 94 rgb0[2 * x + rgb_stride + 1] = av_clip_int16((y11 * cy + crv * v + rnd) >> sh); in fn() 98 rgb1[x << SS_W] = av_clip_int16((y00 * cy + cgu * u + in fn() 101 rgb1[2 * x + 1] = av_clip_int16((y01 * cy + cgu * u + in fn() 104 rgb1[2 * x + rgb_stride] = av_clip_int16((y10 * cy + cgu * u + in fn() 106 rgb1[2 * x + rgb_stride + 1] = av_clip_int16((y11 * cy + cgu * u + in fn() 111 rgb2[x << SS_W] = av_clip_int16((y00 * cy + cbu * u + rnd) >> sh); in fn() 113 rgb2[2 * x + 1] = av_clip_int16((y01 * cy + cbu * u + rnd) >> sh); in fn() [all …]
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D | colorspacedsp.c | 87 buf0[x] = av_clip_int16((m[0][0][0] * v0 + m[0][1][0] * v1 + in multiply3x3_c() 89 buf1[x] = av_clip_int16((m[1][0][0] * v0 + m[1][1][0] * v1 + in multiply3x3_c() 91 buf2[x] = av_clip_int16((m[2][0][0] * v0 + m[2][1][0] * v1 + in multiply3x3_c()
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D | af_earwax.c | 110 *out = av_clip_int16(sample >> 7); in scalarproduct() 168 dst[n] = av_clip_int16(srcl[n] + srcr[n]); in mix()
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/third_party/ffmpeg/libavresample/ |
D | audio_mix.c | 120 MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum))) in MIX_FUNC_GENERIC() 121 MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15)) in MIX_FUNC_GENERIC() 122 MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8)) in MIX_FUNC_GENERIC() 158 *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); in mix_2_to_1_s16p_flt_c() 159 *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); in mix_2_to_1_s16p_flt_c() 160 *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); in mix_2_to_1_s16p_flt_c() 161 *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); in mix_2_to_1_s16p_flt_c() 165 *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); in mix_2_to_1_s16p_flt_c() 702 CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) in ff_audio_mix_set_matrix()
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D | resample_template.c | 53 #define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15) 54 #define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
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D | dither.c | 128 dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i])); in quantize_c() 198 dst[i] = av_clip_int16(lrintf(sample)); in quantize_triangular_ns() 201 dst[i] = av_clip_int16(lrintf(sample + dither[i])); in quantize_triangular_ns()
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/third_party/ffmpeg/libavutil/ |
D | fixed_dsp.c | 88 … dst[i] = av_clip_int16(((((int64_t)s0*wj - (int64_t)s1*wi + 0x40000000) >> 31) + round) >> bits); in vector_fmul_window_scaled_c() 89 … dst[j] = av_clip_int16(((((int64_t)s0*wi + (int64_t)s1*wj + 0x40000000) >> 31) + round) >> bits); in vector_fmul_window_scaled_c()
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D | common.h | 136 #ifndef av_clip_int16 137 # define av_clip_int16 av_clip_int16_c macro
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/third_party/ffmpeg/libavutil/arm/ |
D | intmath.h | 57 #define av_clip_int16 av_clip_int16_arm macro
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/third_party/ffmpeg/libswresample/aarch64/ |
D | resample_init.c | 101 #define OUT(d, v) (v) = ((v) + (1<<(14)))>>15; (d) = av_clip_int16(v) in DECLARE_RESAMPLE_COMMON_TEMPLATE()
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/third_party/ffmpeg/libswresample/arm/ |
D | resample_init.c | 101 #define OUT(d, v) (v) = ((v) + (1<<(14)))>>15; (d) = av_clip_int16(v) in DECLARE_RESAMPLE_COMMON_TEMPLATE()
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/third_party/ffmpeg/libswresample/ |
D | audioconvert.c | 79 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi *… 85 CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi *…
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