mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Peter Ross <pross@xvid.org> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>pull/66/head
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/*
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* Generate a header file for hardcoded DSD tables |
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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 <stdlib.h> |
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#define CONFIG_HARDCODED_TABLES 0 |
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#include "dsd_tablegen.h" |
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#include "tableprint.h" |
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#include <inttypes.h> |
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int main(void) |
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{ |
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dsd_ctables_tableinit(); |
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write_fileheader(); |
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printf("static const double ctables[CTABLES][256] = {\n"); |
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write_float_2d_array(ctables, CTABLES, 256); |
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printf("};\n"); |
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return 0; |
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} |
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/*
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* Header file for hardcoded DSD tables |
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* based on BSD licensed dsd2pcm by Sebastian Gesemann |
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* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved. |
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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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#ifndef AVCODEC_DSD_TABLEGEN_H |
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#define AVCODEC_DSD_TABLEGEN_H |
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#include <stdint.h> |
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#include "libavutil/attributes.h" |
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#define HTAPS 48 /** number of FIR constants */ |
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#define CTABLES ((HTAPS + 7) / 8) /** number of "8 MACs" lookup tables */ |
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#if CONFIG_HARDCODED_TABLES |
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#define dsd_ctables_tableinit() |
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#include "libavcodec/dsd_tables.h" |
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#else |
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#include "libavutil/common.h" |
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/*
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* Properties of this 96-tap lowpass filter when applied on a signal |
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* with sampling rate of 44100*64 Hz: |
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* |
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* () has a delay of 17 microseconds. |
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* |
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* () flat response up to 48 kHz |
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* |
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* () if you downsample afterwards by a factor of 8, the |
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* spectrum below 70 kHz is practically alias-free. |
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* |
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* () stopband rejection is about 160 dB |
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* |
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* The coefficient tables ("ctables") take only 6 Kibi Bytes and |
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* should fit into a modern processor's fast cache. |
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*/ |
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/**
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* The 2nd half (48 coeffs) of a 96-tap symmetric lowpass filter |
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*/ |
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static const double htaps[HTAPS] = { |
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0.09950731974056658, 0.09562845727714668, 0.08819647126516944, |
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0.07782552527068175, 0.06534876523171299, 0.05172629311427257, |
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0.0379429484910187, 0.02490921351762261, 0.0133774746265897, |
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0.003883043418804416, -0.003284703416210726, -0.008080250212687497, |
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-0.01067241812471033, -0.01139427235000863, -0.0106813877974587, |
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-0.009007905078766049, -0.006828859761015335, -0.004535184322001496, |
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-0.002425035959059578, -0.0006922187080790708, 0.0005700762133516592, |
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0.001353838005269448, 0.001713709169690937, 0.001742046839472948, |
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0.001545601648013235, 0.001226696225277855, 0.0008704322683580222, |
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0.0005381636200535649, 0.000266446345425276, 7.002968738383528e-05, |
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-5.279407053811266e-05, -0.0001140625650874684, -0.0001304796361231895, |
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-0.0001189970287491285, -9.396247155265073e-05, -6.577634378272832e-05, |
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-4.07492895872535e-05, -2.17407957554587e-05, -9.163058931391722e-06, |
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-2.017460145032201e-06, 1.249721855219005e-06, 2.166655190537392e-06, |
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1.930520892991082e-06, 1.319400334374195e-06, 7.410039764949091e-07, |
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3.423230509967409e-07, 1.244182214744588e-07, 3.130441005359396e-08 |
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}; |
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static float ctables[CTABLES][256]; |
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static av_cold void dsd_ctables_tableinit(void) |
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{ |
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int t, e, m, k; |
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double acc; |
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for (t = 0; t < CTABLES; ++t) { |
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k = FFMIN(HTAPS - t * 8, 8); |
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for (e = 0; e < 256; ++e) { |
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acc = 0.0; |
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for (m = 0; m < k; ++m) |
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acc += (((e >> (7 - m)) & 1) * 2 - 1) * htaps[t * 8 + m]; |
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ctables[CTABLES - 1 - t][e] = (float)acc; |
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} |
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} |
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} |
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#endif /* CONFIG_HARDCODED_TABLES */ |
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#endif /* AVCODEC_DSD_TABLEGEN_H */ |
@ -0,0 +1,167 @@ |
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/*
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* Direct Stream Digital (DSD) decoder |
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* based on BSD licensed dsd2pcm by Sebastian Gesemann |
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* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved. |
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* Copyright (c) 2014 Peter Ross |
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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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/**
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* @file |
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* Direct Stream Digital (DSD) decoder |
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*/ |
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#include "libavcodec/internal.h" |
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#include "libavcodec/mathops.h" |
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#include "avcodec.h" |
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#include "dsd_tablegen.h" |
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#define FIFOSIZE 16 /** must be a power of two */ |
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#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */ |
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#if FIFOSIZE * 8 < HTAPS * 2 |
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#error "FIFOSIZE too small" |
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#endif |
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/**
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* Per-channel buffer |
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*/ |
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typedef struct { |
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unsigned char buf[FIFOSIZE]; |
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unsigned pos; |
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} DSDContext; |
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static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf, |
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const unsigned char *src, ptrdiff_t src_stride, |
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float *dst, ptrdiff_t dst_stride) |
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{ |
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unsigned pos, i; |
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unsigned char* p; |
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double sum; |
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pos = s->pos; |
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while (samples-- > 0) { |
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s->buf[pos] = lsbf ? ff_reverse[*src] : *src; |
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src += src_stride; |
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p = s->buf + ((pos - CTABLES) & FIFOMASK); |
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*p = ff_reverse[*p]; |
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sum = 0.0; |
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for (i = 0; i < CTABLES; i++) { |
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unsigned char a = s->buf[(pos - i) & FIFOMASK]; |
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unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK]; |
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sum += ctables[i][a] + ctables[i][b]; |
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} |
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*dst = (float)sum; |
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dst += dst_stride; |
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pos = (pos + 1) & FIFOMASK; |
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} |
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s->pos = pos; |
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} |
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static av_cold void init_static_data(void) |
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{ |
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static int done = 0; |
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if (done) |
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return; |
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dsd_ctables_tableinit(); |
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done = 1; |
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} |
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static av_cold int decode_init(AVCodecContext *avctx) |
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{ |
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DSDContext * s; |
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int i; |
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init_static_data(); |
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s = av_malloc(sizeof(DSDContext) * avctx->channels); |
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if (!s) |
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return AVERROR(ENOMEM); |
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for (i = 0; i < avctx->channels; i++) { |
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s[i].pos = 0; |
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memset(s[i].buf, 0x69, sizeof(s[i].buf)); |
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/* 0x69 = 01101001
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* This pattern "on repeat" makes a low energy 352.8 kHz tone |
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* and a high energy 1.0584 MHz tone which should be filtered |
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* out completely by any playback system --> silence |
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*/ |
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} |
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avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
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avctx->priv_data = s; |
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return 0; |
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} |
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static int decode_frame(AVCodecContext *avctx, void *data, |
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int *got_frame_ptr, AVPacket *avpkt) |
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{ |
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DSDContext * s = avctx->priv_data; |
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AVFrame *frame = data; |
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int ret, i; |
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int lsbf = avctx->codec_id == AV_CODEC_ID_DSD_LSBF || avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR; |
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int src_next; |
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int src_stride; |
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frame->nb_samples = avpkt->size / avctx->channels; |
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if (avctx->codec_id == AV_CODEC_ID_DSD_LSBF_PLANAR || avctx->codec_id == AV_CODEC_ID_DSD_MSBF_PLANAR) { |
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src_next = frame->nb_samples; |
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src_stride = 1; |
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} else { |
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src_next = 1; |
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src_stride = avctx->channels; |
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} |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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for (i = 0; i < avctx->channels; i++) { |
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float * dst = ((float **)frame->extended_data)[i]; |
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dsd2pcm_translate(&s[i], frame->nb_samples, lsbf, |
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avpkt->data + i * src_next, src_stride, |
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dst, 1); |
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} |
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*got_frame_ptr = 1; |
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return frame->nb_samples * avctx->channels; |
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} |
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#define DSD_DECODER(id_, name_, long_name_) \ |
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AVCodec ff_##name_##_decoder = { \
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.name = #name_, \
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.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
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.type = AVMEDIA_TYPE_AUDIO, \
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.id = AV_CODEC_ID_##id_, \
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.init = decode_init, \
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.decode = decode_frame, \
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, \
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AV_SAMPLE_FMT_NONE }, \
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}; |
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DSD_DECODER(DSD_LSBF, dsd_lsbf, "DSD (Direct Stream Digital), least significant bit first") |
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DSD_DECODER(DSD_MSBF, dsd_msbf, "DSD (Direct Stream Digital), most significant bit first") |
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DSD_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar, "DSD (Direct Stream Digital), most significant bit first, planar") |
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DSD_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar, "DSD (Direct Stream Digital), least significant bit first, planar") |
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