/* * Copyright (c) 2012 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 "libavutil/attributes.h" #include "libavutil/samplefmt.h" #include "flacdsp.h" #include "config.h" #define SAMPLE_SIZE 16 #define PLANAR 0 #include "flacdsp_template.c" #include "flacdsp_lpc_template.c" #undef PLANAR #define PLANAR 1 #include "flacdsp_template.c" #undef SAMPLE_SIZE #undef PLANAR #define SAMPLE_SIZE 32 #define PLANAR 0 #include "flacdsp_template.c" #include "flacdsp_lpc_template.c" #undef PLANAR #define PLANAR 1 #include "flacdsp_template.c" static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32], int pred_order, int qlevel, int len) { int i, j; for (i = pred_order; i < len - 1; i += 2, decoded += 2) { int c = coeffs[0]; int d = decoded[0]; int s0 = 0, s1 = 0; for (j = 1; j < pred_order; j++) { s0 += c*d; d = decoded[j]; s1 += c*d; c = coeffs[j]; } s0 += c*d; d = decoded[j] += s0 >> qlevel; s1 += c*d; decoded[j + 1] += s1 >> qlevel; } if (i < len) { int sum = 0; for (j = 0; j < pred_order; j++) sum += coeffs[j] * decoded[j]; decoded[j] += sum >> qlevel; } } static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32], int pred_order, int qlevel, int len) { int i, j; for (i = pred_order; i < len; i++, decoded++) { int64_t sum = 0; for (j = 0; j < pred_order; j++) sum += (int64_t)coeffs[j] * decoded[j]; decoded[j] += sum >> qlevel; } } av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps) { if (bps > 16) { c->lpc = flac_lpc_32_c; c->lpc_encode = flac_lpc_encode_c_32; } else { c->lpc = flac_lpc_16_c; c->lpc_encode = flac_lpc_encode_c_16; } switch (fmt) { case AV_SAMPLE_FMT_S32: c->decorrelate[0] = flac_decorrelate_indep_c_32; c->decorrelate[1] = flac_decorrelate_ls_c_32; c->decorrelate[2] = flac_decorrelate_rs_c_32; c->decorrelate[3] = flac_decorrelate_ms_c_32; break; case AV_SAMPLE_FMT_S32P: c->decorrelate[0] = flac_decorrelate_indep_c_32p; c->decorrelate[1] = flac_decorrelate_ls_c_32p; c->decorrelate[2] = flac_decorrelate_rs_c_32p; c->decorrelate[3] = flac_decorrelate_ms_c_32p; break; case AV_SAMPLE_FMT_S16: c->decorrelate[0] = flac_decorrelate_indep_c_16; c->decorrelate[1] = flac_decorrelate_ls_c_16; c->decorrelate[2] = flac_decorrelate_rs_c_16; c->decorrelate[3] = flac_decorrelate_ms_c_16; break; case AV_SAMPLE_FMT_S16P: c->decorrelate[0] = flac_decorrelate_indep_c_16p; c->decorrelate[1] = flac_decorrelate_ls_c_16p; c->decorrelate[2] = flac_decorrelate_rs_c_16p; c->decorrelate[3] = flac_decorrelate_ms_c_16p; break; } if (ARCH_ARM) ff_flacdsp_init_arm(c, fmt, channels, bps); if (ARCH_X86) ff_flacdsp_init_x86(c, fmt, channels, bps); }