mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Paul B Mahol <onemda@gmail.com>pull/215/head
Peter Ross
9 years ago
committed by
Paul B Mahol
parent
365b0c13e4
commit
86e493a6ff
11 changed files with 611 additions and 88 deletions
@ -0,0 +1,86 @@ |
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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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#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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#include "dsd.h" |
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static av_cold void dsd_ctables_tableinit(void) |
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{ |
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int t, e, m, sign; |
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double acc[CTABLES]; |
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for (e = 0; e < 256; ++e) { |
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memset(acc, 0, sizeof(acc)); |
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for (m = 0; m < 8; ++m) { |
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sign = (((e >> (7 - m)) & 1) * 2 - 1); |
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for (t = 0; t < CTABLES; ++t) |
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acc[t] += sign * htaps[t * 8 + m]; |
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} |
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for (t = 0; t < CTABLES; ++t) |
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ctables[CTABLES - 1 - t][e] = acc[t]; |
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} |
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} |
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av_cold void ff_init_dsd_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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void ff_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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@ -0,0 +1,53 @@ |
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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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#ifndef AVCODEC_DSD_H |
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#define AVCODEC_DSD_H |
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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 HTAPS 48 /** number of FIR constants */ |
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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 DSDContext { |
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unsigned char buf[FIFOSIZE]; |
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unsigned pos; |
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} DSDContext; |
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void ff_init_dsd_data(void); |
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void ff_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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#endif /* AVCODEC_DSD_H */ |
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@ -0,0 +1,374 @@ |
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/*
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* Direct Stream Transfer (DST) decoder |
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* Copyright (c) 2014 Peter Ross <pross@xvid.org> |
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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 Transfer (DST) decoder |
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* ISO/IEC 14496-3 Part 3 Subpart 10: Technical description of lossless coding of oversampled audio |
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*/ |
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#include "libavutil/avassert.h" |
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#include "libavutil/intreadwrite.h" |
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#include "internal.h" |
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#include "get_bits.h" |
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#include "avcodec.h" |
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#include "golomb.h" |
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#include "mathops.h" |
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#include "dsd.h" |
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#define DST_MAX_CHANNELS 6 |
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#define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS) |
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#define DSD_FS44(sample_rate) (sample_rate * 8 / 44100) |
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#define DST_SAMPLES_PER_FRAME(sample_rate) (588 * DSD_FS44(sample_rate)) |
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static const int8_t fsets_code_pred_coeff[3][3] = { |
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{ -8 }, |
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{ -16, 8 }, |
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{ -9, -5, 6 }, |
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}; |
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static const int8_t probs_code_pred_coeff[3][3] = { |
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{ -8 }, |
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{ -16, 8 }, |
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{ -24, 24, -8 }, |
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}; |
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typedef struct ArithCoder { |
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unsigned int a; |
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unsigned int c; |
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} ArithCoder; |
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typedef struct Table { |
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unsigned int elements; |
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unsigned int length[DST_MAX_ELEMENTS]; |
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int coeff[DST_MAX_ELEMENTS][128]; |
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} Table; |
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typedef struct DSTContext { |
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AVClass *class; |
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GetBitContext gb; |
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ArithCoder ac; |
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Table fsets, probs; |
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DECLARE_ALIGNED(64, uint8_t, status)[DST_MAX_CHANNELS][16]; |
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DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256]; |
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DSDContext dsdctx[DST_MAX_CHANNELS]; |
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} DSTContext; |
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static av_cold int decode_init(AVCodecContext *avctx) |
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{ |
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DSTContext *s = avctx->priv_data; |
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int i; |
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if (avctx->channels > DST_MAX_CHANNELS) { |
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avpriv_request_sample(avctx, "Channel count %d", avctx->channels); |
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return AVERROR_PATCHWELCOME; |
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} |
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
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for (i = 0; i < avctx->channels; i++) |
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memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf)); |
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ff_init_dsd_data(); |
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return 0; |
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} |
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static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels) |
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{ |
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int ch; |
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t->elements = 1; |
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map[0] = 0; |
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if (!get_bits1(gb)) { |
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for (ch = 1; ch < channels; ch++) { |
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int bits = av_log2(t->elements) + 1; |
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map[ch] = get_bits(gb, bits); |
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if (map[ch] == t->elements) { |
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t->elements++; |
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if (t->elements >= DST_MAX_ELEMENTS) |
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return AVERROR_INVALIDDATA; |
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} else if (map[ch] > t->elements) { |
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return AVERROR_INVALIDDATA; |
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} |
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} |
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} else { |
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memset(map, 0, sizeof(*map) * DST_MAX_CHANNELS); |
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} |
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return 0; |
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} |
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static av_always_inline int get_sr_golomb_dst(GetBitContext *gb, unsigned int k) |
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{ |
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int v = get_ur_golomb(gb, k, get_bits_left(gb), 0); |
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if (v && get_bits1(gb)) |
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v = -v; |
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return v; |
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} |
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static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements, |
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int coeff_bits, int is_signed, int offset) |
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{ |
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int i; |
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for (i = 0; i < elements; i++) { |
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dst[i] = (is_signed ? get_sbits(gb, coeff_bits) : get_bits(gb, coeff_bits)) + offset; |
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} |
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} |
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static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3], |
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int length_bits, int coeff_bits, int is_signed, int offset) |
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{ |
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unsigned int i, j, k; |
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for (i = 0; i < t->elements; i++) { |
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t->length[i] = get_bits(gb, length_bits) + 1; |
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if (!get_bits1(gb)) { |
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read_uncoded_coeff(gb, t->coeff[i], t->length[i], coeff_bits, is_signed, offset); |
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} else { |
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int method = get_bits(gb, 2), lsb_size; |
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if (method == 3) |
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return AVERROR_INVALIDDATA; |
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read_uncoded_coeff(gb, t->coeff[i], method + 1, coeff_bits, is_signed, offset); |
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lsb_size = get_bits(gb, 3); |
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for (j = method + 1; j < t->length[i]; j++) { |
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int c, x = 0; |
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for (k = 0; k < method + 1; k++) |
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x += code_pred_coeff[method][k] * t->coeff[i][j - k - 1]; |
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c = get_sr_golomb_dst(gb, lsb_size); |
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if (x >= 0) |
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c -= (x + 4) / 8; |
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else |
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c += (-x + 3) / 8; |
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t->coeff[i][j] = c; |
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} |
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} |
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} |
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return 0; |
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} |
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static void ac_init(ArithCoder *ac, GetBitContext *gb) |
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{ |
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ac->a = 4095; |
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ac->c = get_bits(gb, 12); |
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} |
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static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e) |
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{ |
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unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1); |
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unsigned int q = k * p; |
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unsigned int a_q = ac->a - q; |
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*e = ac->c < a_q; |
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if (*e) { |
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ac->a = a_q; |
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} else { |
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ac->a = q; |
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ac->c -= a_q; |
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} |
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if (ac->a < 2048) { |
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int n = 11 - av_log2(ac->a); |
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ac->a <<= n; |
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ac->c = (ac->c << n) | get_bits(gb, n); |
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} |
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} |
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static uint8_t prob_dst_x_bit(int c) |
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{ |
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return (ff_reverse[c & 127] >> 1) + 1; |
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} |
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static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets) |
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{ |
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int i, j, k, l; |
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for (i = 0; i < fsets->elements; i++) { |
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int length = fsets->length[i]; |
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for (j = 0; j < 16; j++) { |
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int total = av_clip(length - j * 8, 0, 8); |
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for (k = 0; k < 256; k++) { |
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int v = 0; |
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for (l = 0; l < total; l++) |
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v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l]; |
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table[i][j][k] = v; |
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} |
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} |
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} |
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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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unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate); |
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unsigned map_ch_to_felem[DST_MAX_CHANNELS]; |
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unsigned map_ch_to_pelem[DST_MAX_CHANNELS]; |
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unsigned i, ch, same_map, dst_x_bit; |
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unsigned half_prob[DST_MAX_CHANNELS]; |
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const int channels = avctx->channels; |
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DSTContext *s = avctx->priv_data; |
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GetBitContext *gb = &s->gb; |
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ArithCoder *ac = &s->ac; |
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AVFrame *frame = data; |
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uint8_t *dsd; |
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float *pcm; |
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int ret; |
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if (avpkt->size <= 1) |
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return AVERROR_INVALIDDATA; |
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frame->nb_samples = samples_per_frame / 8; |
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
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return ret; |
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dsd = frame->data[0]; |
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pcm = (float *)frame->data[0]; |
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if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0) |
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return ret; |
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if (!get_bits1(gb)) { |
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skip_bits1(gb); |
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if (get_bits(gb, 6)) |
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return AVERROR_INVALIDDATA; |
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memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * avctx->channels)); |
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goto dsd; |
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} |
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/* Segmentation (10.4, 10.5, 10.6) */ |
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if (!get_bits1(gb)) { |
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avpriv_request_sample(avctx, "Not Same Segmentation"); |
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return AVERROR_PATCHWELCOME; |
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} |
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|
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if (!get_bits1(gb)) { |
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avpriv_request_sample(avctx, "Not Same Segmentation For All Channels"); |
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return AVERROR_PATCHWELCOME; |
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} |
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|
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if (!get_bits1(gb)) { |
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avpriv_request_sample(avctx, "Not End Of Channel Segmentation"); |
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return AVERROR_PATCHWELCOME; |
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} |
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/* Mapping (10.7, 10.8, 10.9) */ |
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same_map = get_bits1(gb); |
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if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, avctx->channels)) < 0) |
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return ret; |
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|
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if (same_map) { |
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s->probs.elements = s->fsets.elements; |
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memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem)); |
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} else { |
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avpriv_request_sample(avctx, "Not Same Mapping"); |
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if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, avctx->channels)) < 0) |
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return ret; |
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} |
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|
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/* Half Probability (10.10) */ |
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for (ch = 0; ch < avctx->channels; ch++) |
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half_prob[ch] = get_bits1(gb); |
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/* Filter Coef Sets (10.12) */ |
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read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0); |
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/* Probability Tables (10.13) */ |
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read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1); |
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|
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/* Arithmetic Coded Data (10.11) */ |
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|
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if (get_bits1(gb)) |
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return AVERROR_INVALIDDATA; |
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ac_init(ac, gb); |
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|
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build_filter(s->filter, &s->fsets); |
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|
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memset(s->status, 0xAA, sizeof(s->status)); |
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memset(dsd, 0, frame->nb_samples * 4 * avctx->channels); |
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ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit); |
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|
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for (i = 0; i < samples_per_frame; i++) { |
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for (ch = 0; ch < channels; ch++) { |
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const unsigned felem = map_ch_to_felem[ch]; |
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const int16_t (*filter)[256] = s->filter[felem]; |
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uint8_t *status = s->status[ch]; |
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int prob, residual, v; |
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#define F(x) filter[(x)][status[(x)]] |
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const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) + |
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F( 4) + F( 5) + F( 6) + F( 7) + |
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F( 8) + F( 9) + F(10) + F(11) + |
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F(12) + F(13) + F(14) + F(15); |
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#undef F |
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|
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if (!half_prob[ch] || i >= s->fsets.length[felem]) { |
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unsigned pelem = map_ch_to_pelem[ch]; |
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unsigned index = FFABS(predict) >> 3; |
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prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)]; |
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} else { |
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prob = 128; |
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} |
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|
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ac_get(ac, gb, prob, &residual); |
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v = ((predict >> 15) ^ residual) & 1; |
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dsd[((i >> 3) * channels + ch) << 2] |= v << (7 - (i & 0x7 )); |
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|
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AV_WN64A(status + 8, (AV_RN64A(status + 8) << 1) | ((AV_RN64A(status) >> 63) & 1)); |
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AV_WN64A(status, (AV_RN64A(status) << 1) | v); |
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} |
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} |
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|
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dsd: |
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for (i = 0; i < avctx->channels; i++) { |
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ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0, |
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frame->data[0] + i * 4, |
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avctx->channels * 4, pcm + i, avctx->channels); |
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} |
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|
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*got_frame_ptr = 1; |
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|
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return avpkt->size; |
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} |
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|
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AVCodec ff_dst_decoder = { |
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.name = "dst", |
||||
.long_name = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"), |
||||
.type = AVMEDIA_TYPE_AUDIO, |
||||
.id = AV_CODEC_ID_DST, |
||||
.priv_data_size = sizeof(DSTContext), |
||||
.init = decode_init, |
||||
.decode = decode_frame, |
||||
.capabilities = AV_CODEC_CAP_DR1, |
||||
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, |
||||
AV_SAMPLE_FMT_NONE }, |
||||
}; |
||||
Loading…
Reference in new issue