mirror of
https://git.ffmpeg.org/ffmpeg.git
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85cd1eb12f
Reviewed-by: Carl Eugen Hoyos Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
588 lines
18 KiB
C
588 lines
18 KiB
C
/*
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* DCA encoder
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* Copyright (C) 2008 Alexander E. Patrakov
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* 2010 Benjamin Larsson
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* 2011 Xiang Wang
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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 "libavutil/common.h"
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#include "libavutil/avassert.h"
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#include "libavutil/audioconvert.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "put_bits.h"
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#include "dcaenc.h"
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#include "dcadata.h"
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#undef NDEBUG
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#define MAX_CHANNELS 6
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#define DCA_SUBBANDS_32 32
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#define DCA_MAX_FRAME_SIZE 16383
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#define DCA_HEADER_SIZE 13
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#define DCA_SUBBANDS 32 ///< Subband activity count
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#define QUANTIZER_BITS 16
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#define SUBFRAMES 1
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#define SUBSUBFRAMES 4
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#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
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#define LFE_BITS 8
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#define LFE_INTERPOLATION 64
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#define LFE_PRESENT 2
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#define LFE_MISSING 0
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static const int8_t dca_lfe_index[] = {
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1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
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};
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static const int8_t dca_channel_reorder_lfe[][9] = {
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{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, 2, -1, -1, -1, -1, -1 },
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{ 1, 2, 0, -1, 3, -1, -1, -1, -1 },
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{ 0, 1, -1, 2, 3, -1, -1, -1, -1 },
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{ 1, 2, 0, -1, 3, 4, -1, -1, -1 },
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{ 2, 3, -1, 0, 1, 4, 5, -1, -1 },
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{ 1, 2, 0, -1, 3, 4, 5, -1, -1 },
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{ 0, -1, 4, 5, 2, 3, 1, -1, -1 },
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{ 3, 4, 1, -1, 0, 2, 5, 6, -1 },
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{ 2, 3, -1, 5, 7, 0, 1, 4, 6 },
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{ 3, 4, 1, -1, 0, 2, 5, 7, 6 },
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};
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static const int8_t dca_channel_reorder_nolfe[][9] = {
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{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
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{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
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{ 0, 1, 2, -1, -1, -1, -1, -1, -1 },
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{ 1, 2, 0, 3, -1, -1, -1, -1, -1 },
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{ 0, 1, 2, 3, -1, -1, -1, -1, -1 },
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{ 1, 2, 0, 3, 4, -1, -1, -1, -1 },
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{ 2, 3, 0, 1, 4, 5, -1, -1, -1 },
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{ 1, 2, 0, 3, 4, 5, -1, -1, -1 },
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{ 0, 4, 5, 2, 3, 1, -1, -1, -1 },
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{ 3, 4, 1, 0, 2, 5, 6, -1, -1 },
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{ 2, 3, 5, 7, 0, 1, 4, 6, -1 },
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{ 3, 4, 1, 0, 2, 5, 7, 6, -1 },
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};
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typedef struct {
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PutBitContext pb;
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int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
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int start[MAX_CHANNELS];
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int frame_size;
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int prim_channels;
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int lfe_channel;
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int sample_rate_code;
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int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
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int lfe_scale_factor;
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int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
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int a_mode; ///< audio channels arrangement
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int num_channel;
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int lfe_state;
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int lfe_offset;
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const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
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int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
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int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
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} DCAContext;
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static int32_t cos_table[128];
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static inline int32_t mul32(int32_t a, int32_t b)
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{
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int64_t r = (int64_t) a * b;
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/* round the result before truncating - improves accuracy */
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return (r + 0x80000000) >> 32;
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}
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/* Integer version of the cosine modulated Pseudo QMF */
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static void qmf_init(void)
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{
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int i;
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int32_t c[17], s[17];
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s[0] = 0; /* sin(index * PI / 64) * 0x7fffffff */
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c[0] = 0x7fffffff; /* cos(index * PI / 64) * 0x7fffffff */
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for (i = 1; i <= 16; i++) {
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s[i] = 2 * (mul32(c[i - 1], 105372028) + mul32(s[i - 1], 2144896908));
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c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
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}
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for (i = 0; i < 16; i++) {
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cos_table[i ] = c[i] >> 3; /* avoid output overflow */
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cos_table[i + 16] = s[16 - i] >> 3;
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cos_table[i + 32] = -s[i] >> 3;
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cos_table[i + 48] = -c[16 - i] >> 3;
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cos_table[i + 64] = -c[i] >> 3;
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cos_table[i + 80] = -s[16 - i] >> 3;
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cos_table[i + 96] = s[i] >> 3;
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cos_table[i + 112] = c[16 - i] >> 3;
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}
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}
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static int32_t band_delta_factor(int band, int sample_num)
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{
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int index = band * (2 * sample_num + 1);
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if (band == 0)
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return 0x07ffffff;
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else
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return cos_table[index & 127];
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}
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static void add_new_samples(DCAContext *c, const int32_t *in,
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int count, int channel)
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{
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int i;
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/* Place new samples into the history buffer */
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for (i = 0; i < count; i++) {
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c->history[channel][c->start[channel] + i] = in[i];
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av_assert0(c->start[channel] + i < 512);
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}
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c->start[channel] += count;
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if (c->start[channel] == 512)
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c->start[channel] = 0;
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av_assert0(c->start[channel] < 512);
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}
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static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
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int channel)
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{
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int band, i, j, k;
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int32_t resp;
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int32_t accum[DCA_SUBBANDS_32] = {0};
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add_new_samples(c, in, DCA_SUBBANDS_32, channel);
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/* Calculate the dot product of the signal with the (possibly inverted)
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reference decoder's response to this vector:
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(0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
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so that -1.0 cancels 1.0 from the previous step */
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for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
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accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
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for (i = 0; i < c->start[channel]; k++, j++, i++)
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accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
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resp = 0;
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/* TODO: implement FFT instead of this naive calculation */
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for (band = 0; band < DCA_SUBBANDS_32; band++) {
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for (j = 0; j < 32; j++)
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resp += mul32(accum[j], band_delta_factor(band, j));
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out[band] = (band & 2) ? (-resp) : resp;
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}
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}
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static int32_t lfe_fir_64i[512];
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static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
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{
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int i, j;
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int channel = c->prim_channels;
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int32_t accum = 0;
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add_new_samples(c, in, LFE_INTERPOLATION, channel);
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for (i = c->start[channel], j = 0; i < 512; i++, j++)
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accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
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for (i = 0; i < c->start[channel]; i++, j++)
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accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
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return accum;
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}
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static void init_lfe_fir(void)
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{
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static int initialized = 0;
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int i;
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if (initialized)
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return;
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for (i = 0; i < 512; i++)
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lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
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initialized = 1;
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}
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static void put_frame_header(DCAContext *c)
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{
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/* SYNC */
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put_bits(&c->pb, 16, 0x7ffe);
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put_bits(&c->pb, 16, 0x8001);
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/* Frame type: normal */
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put_bits(&c->pb, 1, 1);
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/* Deficit sample count: none */
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put_bits(&c->pb, 5, 31);
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/* CRC is not present */
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put_bits(&c->pb, 1, 0);
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/* Number of PCM sample blocks */
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put_bits(&c->pb, 7, PCM_SAMPLES-1);
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/* Primary frame byte size */
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put_bits(&c->pb, 14, c->frame_size-1);
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/* Audio channel arrangement: L + R (stereo) */
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put_bits(&c->pb, 6, c->num_channel);
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/* Core audio sampling frequency */
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put_bits(&c->pb, 4, c->sample_rate_code);
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/* Transmission bit rate: 1411.2 kbps */
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put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
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/* Embedded down mix: disabled */
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put_bits(&c->pb, 1, 0);
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/* Embedded dynamic range flag: not present */
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put_bits(&c->pb, 1, 0);
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/* Embedded time stamp flag: not present */
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put_bits(&c->pb, 1, 0);
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/* Auxiliary data flag: not present */
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put_bits(&c->pb, 1, 0);
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/* HDCD source: no */
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put_bits(&c->pb, 1, 0);
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/* Extension audio ID: N/A */
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put_bits(&c->pb, 3, 0);
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/* Extended audio data: not present */
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put_bits(&c->pb, 1, 0);
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/* Audio sync word insertion flag: after each sub-frame */
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put_bits(&c->pb, 1, 0);
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/* Low frequency effects flag: not present or interpolation factor=64 */
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put_bits(&c->pb, 2, c->lfe_state);
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/* Predictor history switch flag: on */
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put_bits(&c->pb, 1, 1);
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/* No CRC */
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/* Multirate interpolator switch: non-perfect reconstruction */
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put_bits(&c->pb, 1, 0);
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/* Encoder software revision: 7 */
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put_bits(&c->pb, 4, 7);
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/* Copy history: 0 */
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put_bits(&c->pb, 2, 0);
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/* Source PCM resolution: 16 bits, not DTS ES */
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put_bits(&c->pb, 3, 0);
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/* Front sum/difference coding: no */
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put_bits(&c->pb, 1, 0);
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/* Surrounds sum/difference coding: no */
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put_bits(&c->pb, 1, 0);
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/* Dialog normalization: 0 dB */
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put_bits(&c->pb, 4, 0);
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}
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static void put_primary_audio_header(DCAContext *c)
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{
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static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
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static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
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int ch, i;
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/* Number of subframes */
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put_bits(&c->pb, 4, SUBFRAMES - 1);
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/* Number of primary audio channels */
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put_bits(&c->pb, 3, c->prim_channels - 1);
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/* Subband activity count */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
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/* High frequency VQ start subband */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
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/* Joint intensity coding index: 0, 0 */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 3, 0);
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/* Transient mode codebook: A4, A4 (arbitrary) */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 2, 0);
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/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 3, 6);
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/* Bit allocation quantizer select: linear 5-bit */
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, 3, 6);
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/* Quantization index codebook select: dummy data
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to avoid transmission of scale factor adjustment */
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for (i = 1; i < 11; i++)
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for (ch = 0; ch < c->prim_channels; ch++)
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put_bits(&c->pb, bitlen[i], thr[i]);
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/* Scale factor adjustment index: not transmitted */
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}
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/**
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* 8-23 bits quantization
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* @param sample
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* @param bits
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*/
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static inline uint32_t quantize(int32_t sample, int bits)
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{
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av_assert0(sample < 1 << (bits - 1));
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av_assert0(sample >= -(1 << (bits - 1)));
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return sample & ((1 << bits) - 1);
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}
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static inline int find_scale_factor7(int64_t max_value, int bits)
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{
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int i = 0, j = 128, q;
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max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
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while (i < j) {
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q = (i + j) >> 1;
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if (max_value < scale_factor_quant7[q])
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j = q;
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else
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i = q + 1;
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}
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av_assert1(i < 128);
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return i;
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}
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static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
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int scale_factor)
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{
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sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
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put_bits(&c->pb, bits, quantize((int) sample, bits));
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}
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static void put_subframe(DCAContext *c,
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int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
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int subframe)
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{
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int i, sub, ss, ch, max_value;
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int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
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/* Subsubframes count */
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put_bits(&c->pb, 2, SUBSUBFRAMES -1);
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/* Partial subsubframe sample count: dummy */
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put_bits(&c->pb, 3, 0);
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/* Prediction mode: no ADPCM, in each channel and subband */
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for (ch = 0; ch < c->prim_channels; ch++)
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for (sub = 0; sub < DCA_SUBBANDS; sub++)
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put_bits(&c->pb, 1, 0);
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/* Prediction VQ addres: not transmitted */
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/* Bit allocation index */
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for (ch = 0; ch < c->prim_channels; ch++)
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for (sub = 0; sub < DCA_SUBBANDS; sub++)
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put_bits(&c->pb, 5, QUANTIZER_BITS+3);
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if (SUBSUBFRAMES > 1) {
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/* Transition mode: none for each channel and subband */
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for (ch = 0; ch < c->prim_channels; ch++)
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for (sub = 0; sub < DCA_SUBBANDS; sub++)
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put_bits(&c->pb, 1, 0); /* codebook A4 */
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}
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/* Determine scale_factor */
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for (ch = 0; ch < c->prim_channels; ch++)
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for (sub = 0; sub < DCA_SUBBANDS; sub++) {
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max_value = 0;
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for (i = 0; i < 8 * SUBSUBFRAMES; i++)
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max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
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c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
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}
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if (c->lfe_channel) {
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max_value = 0;
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for (i = 0; i < 4 * SUBSUBFRAMES; i++)
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max_value = FFMAX(max_value, FFABS(lfe_data[i]));
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c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
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}
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/* Scale factors: the same for each channel and subband,
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encoded according to Table D.1.2 */
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for (ch = 0; ch < c->prim_channels; ch++)
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for (sub = 0; sub < DCA_SUBBANDS; sub++)
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put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
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|
|
/* Joint subband scale factor codebook select: not transmitted */
|
|
/* Scale factors for joint subband coding: not transmitted */
|
|
/* Stereo down-mix coefficients: not transmitted */
|
|
/* Dynamic range coefficient: not transmitted */
|
|
/* Stde information CRC check word: not transmitted */
|
|
/* VQ encoded high frequency subbands: not transmitted */
|
|
|
|
/* LFE data */
|
|
if (c->lfe_channel) {
|
|
for (i = 0; i < 4 * SUBSUBFRAMES; i++)
|
|
put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
|
|
put_bits(&c->pb, 8, c->lfe_scale_factor);
|
|
}
|
|
|
|
/* Audio data (subsubframes) */
|
|
|
|
for (ss = 0; ss < SUBSUBFRAMES ; ss++)
|
|
for (ch = 0; ch < c->prim_channels; ch++)
|
|
for (sub = 0; sub < DCA_SUBBANDS; sub++)
|
|
for (i = 0; i < 8; i++)
|
|
put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
|
|
|
|
/* DSYNC */
|
|
put_bits(&c->pb, 16, 0xffff);
|
|
}
|
|
|
|
static void put_frame(DCAContext *c,
|
|
int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
|
|
uint8_t *frame)
|
|
{
|
|
int i;
|
|
init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
|
|
|
|
put_primary_audio_header(c);
|
|
for (i = 0; i < SUBFRAMES; i++)
|
|
put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
|
|
|
|
flush_put_bits(&c->pb);
|
|
c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
|
|
|
|
init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
|
|
put_frame_header(c);
|
|
flush_put_bits(&c->pb);
|
|
}
|
|
|
|
static int encode_frame(AVCodecContext *avctx, uint8_t *frame,
|
|
int buf_size, void *data)
|
|
{
|
|
int i, k, channel;
|
|
DCAContext *c = avctx->priv_data;
|
|
int16_t *samples = data;
|
|
int real_channel = 0;
|
|
|
|
for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
|
|
for (channel = 0; channel < c->prim_channels + 1; channel++) {
|
|
/* Get 32 PCM samples */
|
|
for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
|
|
c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
|
|
}
|
|
/* Put subband samples into the proper place */
|
|
real_channel = c->channel_order_tab[channel];
|
|
if (real_channel >= 0) {
|
|
qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (c->lfe_channel) {
|
|
for (i = 0; i < PCM_SAMPLES / 2; i++) {
|
|
for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
|
|
c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
|
|
c->lfe_data[i] = lfe_downsample(c, c->pcm);
|
|
}
|
|
}
|
|
|
|
put_frame(c, c->subband, frame);
|
|
|
|
return c->frame_size;
|
|
}
|
|
|
|
static int encode_init(AVCodecContext *avctx)
|
|
{
|
|
DCAContext *c = avctx->priv_data;
|
|
int i;
|
|
|
|
c->prim_channels = avctx->channels;
|
|
c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
|
|
|
|
switch (avctx->channel_layout) {
|
|
case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
|
|
case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
|
|
case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
|
|
case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
|
|
case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
|
|
default:
|
|
av_log(avctx, AV_LOG_ERROR,
|
|
"Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
|
|
return AVERROR_PATCHWELCOME;
|
|
}
|
|
|
|
if (c->lfe_channel) {
|
|
init_lfe_fir();
|
|
c->prim_channels--;
|
|
c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
|
|
c->lfe_state = LFE_PRESENT;
|
|
c->lfe_offset = dca_lfe_index[c->a_mode];
|
|
} else {
|
|
c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
|
|
c->lfe_state = LFE_MISSING;
|
|
}
|
|
|
|
for (i = 0; i < 16; i++) {
|
|
if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
|
|
break;
|
|
}
|
|
if (i == 16) {
|
|
av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
|
|
for (i = 0; i < 16; i++)
|
|
av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
|
|
av_log(avctx, AV_LOG_ERROR, "supported.\n");
|
|
return -1;
|
|
}
|
|
c->sample_rate_code = i;
|
|
|
|
avctx->frame_size = 32 * PCM_SAMPLES;
|
|
|
|
if (!cos_table[127])
|
|
qmf_init();
|
|
return 0;
|
|
}
|
|
|
|
AVCodec ff_dca_encoder = {
|
|
.name = "dca",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = CODEC_ID_DTS,
|
|
.priv_data_size = sizeof(DCAContext),
|
|
.init = encode_init,
|
|
.encode = encode_frame,
|
|
.capabilities = CODEC_CAP_EXPERIMENTAL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
|
|
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
|
|
};
|