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by Sérgio Benjamim
FFmpeg 2.7.1 source for ppsspp. |
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/*
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* Copyright (c) 2004 Gildas Bazin
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* Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h" |
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#include "libavutil/attributes.h" |
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#include "libavutil/intreadwrite.h" |
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#include "dcadsp.h" |
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static void decode_hf_c(float dst[DCA_SUBBANDS][8], |
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const int32_t vq_num[DCA_SUBBANDS], |
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const int8_t hf_vq[1024][32], intptr_t vq_offset, |
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int32_t scale[DCA_SUBBANDS][2], |
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intptr_t start, intptr_t end) |
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{
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int i, l; |
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for (l = start; l < end; l++) { |
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/* 1 vector -> 32 samples but we only need the 8 samples
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* for this subsubframe. */
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const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset]; |
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float fscale = scale[l][0] * (1 / 16.0); |
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for (i = 0; i < 8; i++) |
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dst[l][i] = ptr[i] * fscale; |
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}
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}
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static inline void dca_lfe_fir(float *out, const float *in, const float *coefs, |
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int decifactor) |
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{
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float *out2 = out + 2 * decifactor - 1; |
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int num_coeffs = 256 / decifactor; |
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int j, k; |
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/* One decimated sample generates 2*decifactor interpolated ones */
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for (k = 0; k < decifactor; k++) { |
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float v0 = 0.0; |
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float v1 = 0.0; |
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for (j = 0; j < num_coeffs; j++, coefs++) { |
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v0 += in[-j] * *coefs; |
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v1 += in[j + 1 - num_coeffs] * *coefs; |
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}
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*out++ = v0; |
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*out2-- = v1; |
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}
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}
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static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act, |
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SynthFilterContext *synth, FFTContext *imdct, |
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float synth_buf_ptr[512], |
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int *synth_buf_offset, float synth_buf2[32], |
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const float window[512], float *samples_out, |
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float raXin[32], float scale) |
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{
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int i; |
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int subindex; |
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for (i = sb_act; i < 32; i++) |
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raXin[i] = 0.0; |
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/* Reconstructed channel sample index */
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for (subindex = 0; subindex < 8; subindex++) { |
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/* Load in one sample from each subband and clear inactive subbands */
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for (i = 0; i < sb_act; i++) { |
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unsigned sign = (i - 1) & 2; |
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uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30; |
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AV_WN32A(&raXin[i], v); |
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}
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synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset, |
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synth_buf2, window, samples_out, raXin, |
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scale); |
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samples_out += 32; |
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}
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}
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static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs) |
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{
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dca_lfe_fir(out, in, coefs, 32); |
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}
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static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs) |
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{
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dca_lfe_fir(out, in, coefs, 64); |
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}
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av_cold void ff_dcadsp_init(DCADSPContext *s) |
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{
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s->lfe_fir[0] = dca_lfe_fir0_c; |
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s->lfe_fir[1] = dca_lfe_fir1_c; |
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s->qmf_32_subbands = dca_qmf_32_subbands; |
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s->decode_hf = decode_hf_c; |
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if (ARCH_ARM) |
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ff_dcadsp_init_arm(s); |
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if (ARCH_X86) |
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ff_dcadsp_init_x86(s); |
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}
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