/* * Copyright (c) 2004 Gildas Bazin * Copyright (c) 2010 Mans Rullgard <mans@mansr.com> * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; 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" static void decode_hf_c(float dst[DCA_SUBBANDS][8], 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, l; for (l = start; l < end; l++) { /* 1 vector -> 32 samples but we only need the 8 samples * for this subsubframe. */ const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset]; float fscale = scale[l][0] * (1 / 16.0); for (i = 0; i < 8; i++) dst[l][i] = ptr[i] * fscale; } } 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[32][8], 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 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; if (ARCH_ARM) ff_dcadsp_init_arm(s); if (ARCH_X86) ff_dcadsp_init_x86(s); }