| /* |
| * 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); |
| } |