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- /*
- * Copyright (c) 2004 Gildas Bazin
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- *
- * This file is part of Libav.
- *
- * Libav is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * Libav is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
- #include "config.h"
-
- #include "libavutil/attributes.h"
- #include "libavutil/intreadwrite.h"
-
- #include "dcadsp.h"
- #include "dcamath.h"
-
- static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
- const int32_t vq_num[DCA_SUBBANDS],
- const int8_t hf_vq[1024][32], intptr_t vq_offset,
- int32_t scale[DCA_SUBBANDS][2],
- intptr_t start, intptr_t end)
- {
- int i, j;
-
- for (j = start; j < end; j++) {
- const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
- for (i = 0; i < 8; i++)
- dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
- }
- }
-
- static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
- int decifactor)
- {
- float *out2 = out + 2 * decifactor - 1;
- int num_coeffs = 256 / decifactor;
- int j, k;
-
- /* One decimated sample generates 2*decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float v0 = 0.0;
- float v1 = 0.0;
- for (j = 0; j < num_coeffs; j++, coefs++) {
- v0 += in[-j] * *coefs;
- v1 += in[j + 1 - num_coeffs] * *coefs;
- }
- *out++ = v0;
- *out2-- = v1;
- }
- }
-
- static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
- SynthFilterContext *synth, FFTContext *imdct,
- float synth_buf_ptr[512],
- int *synth_buf_offset, float synth_buf2[32],
- const float window[512], float *samples_out,
- float raXin[32], float scale)
- {
- int i;
- int subindex;
-
- for (i = sb_act; i < 32; i++)
- raXin[i] = 0.0;
-
- /* Reconstructed channel sample index */
- for (subindex = 0; subindex < 8; subindex++) {
- /* Load in one sample from each subband and clear inactive subbands */
- for (i = 0; i < sb_act; i++) {
- unsigned sign = (i - 1) & 2;
- uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
- AV_WN32A(&raXin[i], v);
- }
-
- synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
- synth_buf2, window, samples_out, raXin,
- scale);
- samples_out += 32;
- }
- }
-
- static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
- {
- int64_t step = (int64_t)step_size * scale;
- int shift, i;
- int32_t step_scale;
-
- if (step > (1 << 23))
- shift = av_log2(step >> 23) + 1;
- else
- shift = 0;
- step_scale = (int32_t)(step >> shift);
-
- for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
- samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
- }
-
- static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
- {
- dca_lfe_fir(out, in, coefs, 32);
- }
-
- static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
- {
- dca_lfe_fir(out, in, coefs, 64);
- }
-
- av_cold void ff_dcadsp_init(DCADSPContext *s)
- {
- s->lfe_fir[0] = dca_lfe_fir0_c;
- s->lfe_fir[1] = dca_lfe_fir1_c;
- s->qmf_32_subbands = dca_qmf_32_subbands;
- s->decode_hf = decode_hf_c;
- s->dequantize = dequantize_c;
-
- if (ARCH_AARCH64)
- ff_dcadsp_init_aarch64(s);
- if (ARCH_ARM)
- ff_dcadsp_init_arm(s);
- if (ARCH_X86)
- ff_dcadsp_init_x86(s);
- }
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