/* * Copyright (c) 2007-2008 Ian Caulfield * 2009 Ramiro Polla * * 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 "mlpdsp.h" #include "mlp.h" static void mlp_filter_channel(int32_t *state, const int32_t *coeff, int firorder, int iirorder, unsigned int filter_shift, int32_t mask, int blocksize, int32_t *sample_buffer) { int32_t *firbuf = state; int32_t *iirbuf = state + MAX_BLOCKSIZE + MAX_FIR_ORDER; const int32_t *fircoeff = coeff; const int32_t *iircoeff = coeff + MAX_FIR_ORDER; int i; for (i = 0; i < blocksize; i++) { int32_t residual = *sample_buffer; unsigned int order; int64_t accum = 0; int32_t result; for (order = 0; order < firorder; order++) accum += (int64_t) firbuf[order] * fircoeff[order]; for (order = 0; order < iirorder; order++) accum += (int64_t) iirbuf[order] * iircoeff[order]; accum = accum >> filter_shift; result = (accum + residual) & mask; *--firbuf = result; *--iirbuf = result - accum; *sample_buffer = result; sample_buffer += MAX_CHANNELS; } } void ff_mlp_rematrix_channel(int32_t *samples, const int32_t *coeffs, const uint8_t *bypassed_lsbs, const int8_t *noise_buffer, int index, unsigned int dest_ch, uint16_t blockpos, unsigned int maxchan, int matrix_noise_shift, int access_unit_size_pow2, int32_t mask) { unsigned int src_ch, i; int index2 = 2 * index + 1; for (i = 0; i < blockpos; i++) { int64_t accum = 0; for (src_ch = 0; src_ch <= maxchan; src_ch++) accum += (int64_t) samples[src_ch] * coeffs[src_ch]; if (matrix_noise_shift) { index &= access_unit_size_pow2 - 1; accum += noise_buffer[index] << (matrix_noise_shift + 7); index += index2; } samples[dest_ch] = ((accum >> 14) & mask) + *bypassed_lsbs; bypassed_lsbs += MAX_CHANNELS; samples += MAX_CHANNELS; } } static int32_t (*mlp_select_pack_output(uint8_t *ch_assign, int8_t *output_shift, uint8_t max_matrix_channel, int is32))(int32_t, uint16_t, int32_t (*)[], void *, uint8_t*, int8_t *, uint8_t, int) { return ff_mlp_pack_output; } int32_t ff_mlp_pack_output(int32_t lossless_check_data, uint16_t blockpos, int32_t (*sample_buffer)[MAX_CHANNELS], void *data, uint8_t *ch_assign, int8_t *output_shift, uint8_t max_matrix_channel, int is32) { unsigned int i, out_ch = 0; int32_t *data_32 = data; int16_t *data_16 = data; for (i = 0; i < blockpos; i++) { for (out_ch = 0; out_ch <= max_matrix_channel; out_ch++) { int mat_ch = ch_assign[out_ch]; int32_t sample = sample_buffer[i][mat_ch] * (1 << output_shift[mat_ch]); lossless_check_data ^= (sample & 0xffffff) << mat_ch; if (is32) *data_32++ = sample << 8; else *data_16++ = sample >> 8; } } return lossless_check_data; } av_cold void ff_mlpdsp_init(MLPDSPContext *c) { c->mlp_filter_channel = mlp_filter_channel; c->mlp_rematrix_channel = ff_mlp_rematrix_channel; c->mlp_select_pack_output = mlp_select_pack_output; c->mlp_pack_output = ff_mlp_pack_output; if (ARCH_ARM) ff_mlpdsp_init_arm(c); if (ARCH_X86) ff_mlpdsp_init_x86(c); }