2 * Copyright (c) 2004 Gildas Bazin
3 * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include "libavutil/attributes.h"
25 #include "libavutil/intreadwrite.h"
30 static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
31 const int32_t vq_num[DCA_SUBBANDS],
32 const int8_t hf_vq[1024][32], intptr_t vq_offset,
33 int32_t scale[DCA_SUBBANDS][2],
34 intptr_t start, intptr_t end)
38 for (j = start; j < end; j++) {
39 const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
40 for (i = 0; i < 8; i++)
41 dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
45 static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
48 float *out2 = out + 2 * decifactor - 1;
49 int num_coeffs = 256 / decifactor;
52 /* One decimated sample generates 2*decifactor interpolated ones */
53 for (k = 0; k < decifactor; k++) {
56 for (j = 0; j < num_coeffs; j++, coefs++) {
57 v0 += in[-j] * *coefs;
58 v1 += in[j + 1 - num_coeffs] * *coefs;
65 static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
66 SynthFilterContext *synth, FFTContext *imdct,
67 float synth_buf_ptr[512],
68 int *synth_buf_offset, float synth_buf2[32],
69 const float window[512], float *samples_out,
70 float raXin[32], float scale)
75 for (i = sb_act; i < 32; i++)
78 /* Reconstructed channel sample index */
79 for (subindex = 0; subindex < 8; subindex++) {
80 /* Load in one sample from each subband and clear inactive subbands */
81 for (i = 0; i < sb_act; i++) {
82 unsigned sign = (i - 1) & 2;
83 uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
84 AV_WN32A(&raXin[i], v);
87 synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
88 synth_buf2, window, samples_out, raXin,
94 static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
96 int64_t step = (int64_t)step_size * scale;
100 if (step > (1 << 23))
101 shift = av_log2(step >> 23) + 1;
104 step_scale = (int32_t)(step >> shift);
106 for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
107 samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
110 static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
112 dca_lfe_fir(out, in, coefs, 32);
115 static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
117 dca_lfe_fir(out, in, coefs, 64);
120 av_cold void ff_dcadsp_init(DCADSPContext *s)
122 s->lfe_fir[0] = dca_lfe_fir0_c;
123 s->lfe_fir[1] = dca_lfe_fir1_c;
124 s->qmf_32_subbands = dca_qmf_32_subbands;
125 s->decode_hf = decode_hf_c;
126 s->dequantize = dequantize_c;
129 ff_dcadsp_init_aarch64(s);
131 ff_dcadsp_init_arm(s);
133 ff_dcadsp_init_x86(s);