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Diffstat (limited to '')
| -rw-r--r-- | media/ffvpx/libavcodec/flacdsp.c | 127 |
1 files changed, 127 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/flacdsp.c b/media/ffvpx/libavcodec/flacdsp.c new file mode 100644 index 0000000000..42e231db53 --- /dev/null +++ b/media/ffvpx/libavcodec/flacdsp.c @@ -0,0 +1,127 @@ +/* + * 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/internal.h" +#include "libavutil/samplefmt.h" +#include "flacdsp.h" +#include "config.h" + +#define SAMPLE_SIZE 16 +#define PLANAR 0 +#include "flacdsp_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" + +#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) { + SUINT c = coeffs[0]; + SUINT 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] += (SUINT)(s0 >> qlevel); + s1 += c*d; + decoded[j + 1] += (SUINT)(s1 >> qlevel); + } + if (i < len) { + int sum = 0; + for (j = 0; j < pred_order; j++) + sum += coeffs[j] * (SUINT)decoded[j]; + decoded[j] = decoded[j] + (unsigned)(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) +{ + c->lpc16 = flac_lpc_16_c; + c->lpc32 = flac_lpc_32_c; + + 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); +#elif ARCH_X86 + ff_flacdsp_init_x86(c, fmt, channels); +#endif +} |