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b46f19100b
See b2bed9325
.
975 lines
30 KiB
C
975 lines
30 KiB
C
/*
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* DCA encoder
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* Copyright (C) 2008-2012 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/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "avcodec.h"
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#include "dca.h"
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#include "dcadata.h"
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#include "dcaenc.h"
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#include "internal.h"
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#include "put_bits.h"
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#define MAX_CHANNELS 6
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#define DCA_MAX_FRAME_SIZE 16384
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#define DCA_HEADER_SIZE 13
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#define DCA_LFE_SAMPLES 8
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#define DCA_SUBBANDS 32
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#define SUBFRAMES 1
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#define SUBSUBFRAMES 2
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#define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
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#define AUBANDS 25
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typedef struct DCAContext {
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PutBitContext pb;
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int frame_size;
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int frame_bits;
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int fullband_channels;
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int channels;
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int lfe_channel;
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int samplerate_index;
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int bitrate_index;
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int channel_config;
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const int32_t *band_interpolation;
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const int32_t *band_spectrum;
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int lfe_scale_factor;
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softfloat lfe_quant;
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int32_t lfe_peak_cb;
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int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
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int32_t subband[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];
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int32_t quantized[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS];
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int32_t peak_cb[DCA_SUBBANDS][MAX_CHANNELS];
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int32_t downsampled_lfe[DCA_LFE_SAMPLES];
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int32_t masking_curve_cb[SUBSUBFRAMES][256];
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int abits[DCA_SUBBANDS][MAX_CHANNELS];
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int scale_factor[DCA_SUBBANDS][MAX_CHANNELS];
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softfloat quant[DCA_SUBBANDS][MAX_CHANNELS];
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int32_t eff_masking_curve_cb[256];
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int32_t band_masking_cb[32];
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int32_t worst_quantization_noise;
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int32_t worst_noise_ever;
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int consumed_bits;
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} DCAContext;
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static int32_t cos_table[2048];
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static int32_t band_interpolation[2][512];
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static int32_t band_spectrum[2][8];
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static int32_t auf[9][AUBANDS][256];
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static int32_t cb_to_add[256];
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static int32_t cb_to_level[2048];
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static int32_t lfe_fir_64i[512];
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/* Transfer function of outer and middle ear, Hz -> dB */
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static double hom(double f)
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{
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double f1 = f / 1000;
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return -3.64 * pow(f1, -0.8)
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+ 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4))
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- 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7))
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- 0.0006 * (f1 * f1) * (f1 * f1);
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}
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static double gammafilter(int i, double f)
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{
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double h = (f - fc[i]) / erb[i];
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h = 1 + h * h;
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h = 1 / (h * h);
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return 20 * log10(h);
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}
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static int encode_init(AVCodecContext *avctx)
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{
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DCAContext *c = avctx->priv_data;
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uint64_t layout = avctx->channel_layout;
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int i, min_frame_bits;
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c->fullband_channels = c->channels = avctx->channels;
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c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6);
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c->band_interpolation = band_interpolation[1];
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c->band_spectrum = band_spectrum[1];
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c->worst_quantization_noise = -2047;
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c->worst_noise_ever = -2047;
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if (!layout) {
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av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
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"encoder will guess the layout, but it "
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"might be incorrect.\n");
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layout = av_get_default_channel_layout(avctx->channels);
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}
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switch (layout) {
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case AV_CH_LAYOUT_MONO: c->channel_config = 0; break;
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case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break;
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case AV_CH_LAYOUT_2_2: c->channel_config = 8; break;
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case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break;
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case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break;
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default:
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av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n");
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return AVERROR_PATCHWELCOME;
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}
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if (c->lfe_channel)
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c->fullband_channels--;
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for (i = 0; i < 9; i++) {
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if (sample_rates[i] == avctx->sample_rate)
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break;
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}
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if (i == 9)
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return AVERROR(EINVAL);
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c->samplerate_index = i;
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if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) {
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av_log(avctx, AV_LOG_ERROR, "Bit rate %i not supported.", avctx->bit_rate);
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return AVERROR(EINVAL);
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}
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for (i = 0; dca_bit_rates[i] < avctx->bit_rate; i++)
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;
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c->bitrate_index = i;
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avctx->bit_rate = dca_bit_rates[i];
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c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32);
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min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72;
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if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3))
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return AVERROR(EINVAL);
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c->frame_size = (c->frame_bits + 7) / 8;
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avctx->frame_size = 32 * SUBBAND_SAMPLES;
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if (!cos_table[0]) {
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int j, k;
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for (i = 0; i < 2048; i++) {
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cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
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cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i));
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}
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/* FIXME: probably incorrect */
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for (i = 0; i < 256; i++) {
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lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
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lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * lfe_fir_64[i]);
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}
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for (i = 0; i < 512; i++) {
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band_interpolation[0][i] = (int32_t)(0x1000000000ULL * fir_32bands_perfect[i]);
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band_interpolation[1][i] = (int32_t)(0x1000000000ULL * fir_32bands_nonperfect[i]);
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}
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for (i = 0; i < 9; i++) {
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for (j = 0; j < AUBANDS; j++) {
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for (k = 0; k < 256; k++) {
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double freq = sample_rates[i] * (k + 0.5) / 512;
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auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq)));
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}
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}
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}
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for (i = 0; i < 256; i++) {
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double add = 1 + pow(10, -0.01 * i);
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cb_to_add[i] = (int32_t)(100 * log10(add));
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}
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for (j = 0; j < 8; j++) {
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double accum = 0;
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for (i = 0; i < 512; i++) {
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double reconst = fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1);
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accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
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}
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band_spectrum[0][j] = (int32_t)(200 * log10(accum));
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}
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for (j = 0; j < 8; j++) {
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double accum = 0;
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for (i = 0; i < 512; i++) {
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double reconst = fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1);
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accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512);
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}
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band_spectrum[1][j] = (int32_t)(200 * log10(accum));
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}
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}
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return 0;
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}
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static inline int32_t cos_t(int x)
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{
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return cos_table[x & 2047];
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}
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static inline int32_t sin_t(int x)
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{
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return cos_t(x - 512);
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}
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static inline int32_t half32(int32_t a)
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{
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return (a + 1) >> 1;
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}
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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 + 0x80000000ULL;
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return r >> 32;
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}
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static void subband_transform(DCAContext *c, const int32_t *input)
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{
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int ch, subs, i, k, j;
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for (ch = 0; ch < c->fullband_channels; ch++) {
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/* History is copied because it is also needed for PSY */
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int32_t hist[512];
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int hist_start = 0;
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for (i = 0; i < 512; i++)
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hist[i] = c->history[i][ch];
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for (subs = 0; subs < SUBBAND_SAMPLES; subs++) {
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int32_t accum[64];
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int32_t resp;
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int band;
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/* Calculate the convolutions at once */
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for (i = 0; i < 64; i++)
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accum[i] = 0;
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for (k = 0, i = hist_start, j = 0;
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i < 512; k = (k + 1) & 63, i++, j++)
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accum[k] += mul32(hist[i], c->band_interpolation[j]);
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for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++)
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accum[k] += mul32(hist[i], c->band_interpolation[j]);
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for (k = 16; k < 32; k++)
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accum[k] = accum[k] - accum[31 - k];
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for (k = 32; k < 48; k++)
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accum[k] = accum[k] + accum[95 - k];
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for (band = 0; band < 32; band++) {
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resp = 0;
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for (i = 16; i < 48; i++) {
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int s = (2 * band + 1) * (2 * (i + 16) + 1);
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resp += mul32(accum[i], cos_t(s << 3)) >> 3;
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}
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c->subband[subs][band][ch] = ((band + 1) & 2) ? -resp : resp;
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}
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/* Copy in 32 new samples from input */
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for (i = 0; i < 32; i++)
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hist[i + hist_start] = input[(subs * 32 + i) * c->channels + ch];
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hist_start = (hist_start + 32) & 511;
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}
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}
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}
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static void lfe_downsample(DCAContext *c, const int32_t *input)
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{
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/* FIXME: make 128x LFE downsampling possible */
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int i, j, lfes;
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int32_t hist[512];
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int32_t accum;
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int hist_start = 0;
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for (i = 0; i < 512; i++)
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hist[i] = c->history[i][c->channels - 1];
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for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) {
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/* Calculate the convolution */
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accum = 0;
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for (i = hist_start, j = 0; i < 512; i++, j++)
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accum += mul32(hist[i], lfe_fir_64i[j]);
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for (i = 0; i < hist_start; i++, j++)
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accum += mul32(hist[i], lfe_fir_64i[j]);
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c->downsampled_lfe[lfes] = accum;
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/* Copy in 64 new samples from input */
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for (i = 0; i < 64; i++)
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hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + c->channels - 1];
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hist_start = (hist_start + 64) & 511;
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}
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}
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typedef struct {
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int32_t re;
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int32_t im;
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} cplx32;
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static void fft(const int32_t in[2 * 256], cplx32 out[256])
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{
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cplx32 buf[256], rin[256], rout[256];
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int i, j, k, l;
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/* do two transforms in parallel */
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for (i = 0; i < 256; i++) {
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/* Apply the Hann window */
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rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1));
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rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1));
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}
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/* pre-rotation */
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for (i = 0; i < 256; i++) {
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buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re)
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- mul32(sin_t(4 * i + 2), rin[i].im);
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buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im)
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+ mul32(sin_t(4 * i + 2), rin[i].re);
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}
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for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) {
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for (k = 0; k < 256; k += j) {
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for (i = k; i < k + j / 2; i++) {
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cplx32 sum, diff;
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int t = 8 * l * i;
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sum.re = buf[i].re + buf[i + j / 2].re;
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sum.im = buf[i].im + buf[i + j / 2].im;
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diff.re = buf[i].re - buf[i + j / 2].re;
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diff.im = buf[i].im - buf[i + j / 2].im;
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buf[i].re = half32(sum.re);
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buf[i].im = half32(sum.im);
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buf[i + j / 2].re = mul32(diff.re, cos_t(t))
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- mul32(diff.im, sin_t(t));
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buf[i + j / 2].im = mul32(diff.im, cos_t(t))
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+ mul32(diff.re, sin_t(t));
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}
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}
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}
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/* post-rotation */
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for (i = 0; i < 256; i++) {
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int b = ff_reverse[i];
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rout[i].re = mul32(buf[b].re, cos_t(4 * i))
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- mul32(buf[b].im, sin_t(4 * i));
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rout[i].im = mul32(buf[b].im, cos_t(4 * i))
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+ mul32(buf[b].re, sin_t(4 * i));
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}
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for (i = 0; i < 256; i++) {
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/* separate the results of the two transforms */
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cplx32 o1, o2;
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o1.re = rout[i].re - rout[255 - i].re;
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o1.im = rout[i].im + rout[255 - i].im;
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o2.re = rout[i].im - rout[255 - i].im;
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o2.im = -rout[i].re - rout[255 - i].re;
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/* combine them into one long transform */
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out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1))
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+ mul32( o1.im - o2.im, sin_t(2 * i + 1));
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out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1))
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+ mul32(-o1.re + o2.re, sin_t(2 * i + 1));
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}
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}
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static int32_t get_cb(int32_t in)
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{
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int i, res;
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res = 0;
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if (in < 0)
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in = -in;
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for (i = 1024; i > 0; i >>= 1) {
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if (cb_to_level[i + res] >= in)
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res += i;
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}
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return -res;
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}
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static int32_t add_cb(int32_t a, int32_t b)
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{
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if (a < b)
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FFSWAP(int32_t, a, b);
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if (a - b >= 256)
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return a;
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return a + cb_to_add[a - b];
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}
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static void adjust_jnd(int samplerate_index,
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const int32_t in[512], int32_t out_cb[256])
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{
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int32_t power[256];
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cplx32 out[256];
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int32_t out_cb_unnorm[256];
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int32_t denom;
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const int32_t ca_cb = -1114;
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const int32_t cs_cb = 928;
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int i, j;
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fft(in, out);
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for (j = 0; j < 256; j++) {
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power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im));
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out_cb_unnorm[j] = -2047; /* and can only grow */
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}
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for (i = 0; i < AUBANDS; i++) {
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denom = ca_cb; /* and can only grow */
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for (j = 0; j < 256; j++)
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denom = add_cb(denom, power[j] + auf[samplerate_index][i][j]);
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for (j = 0; j < 256; j++)
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out_cb_unnorm[j] = add_cb(out_cb_unnorm[j],
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-denom + auf[samplerate_index][i][j]);
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}
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for (j = 0; j < 256; j++)
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out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb);
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}
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typedef void (*walk_band_t)(DCAContext *c, int band1, int band2, int f,
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int32_t spectrum1, int32_t spectrum2, int channel,
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int32_t * arg);
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static void walk_band_low(DCAContext *c, int band, int channel,
|
|
walk_band_t walk, int32_t *arg)
|
|
{
|
|
int f;
|
|
|
|
if (band == 0) {
|
|
for (f = 0; f < 4; f++)
|
|
walk(c, 0, 0, f, 0, -2047, channel, arg);
|
|
} else {
|
|
for (f = 0; f < 8; f++)
|
|
walk(c, band, band - 1, 8 * band - 4 + f,
|
|
c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg);
|
|
}
|
|
}
|
|
|
|
static void walk_band_high(DCAContext *c, int band, int channel,
|
|
walk_band_t walk, int32_t *arg)
|
|
{
|
|
int f;
|
|
|
|
if (band == 31) {
|
|
for (f = 0; f < 4; f++)
|
|
walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg);
|
|
} else {
|
|
for (f = 0; f < 8; f++)
|
|
walk(c, band, band + 1, 8 * band + 4 + f,
|
|
c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg);
|
|
}
|
|
}
|
|
|
|
static void update_band_masking(DCAContext *c, int band1, int band2,
|
|
int f, int32_t spectrum1, int32_t spectrum2,
|
|
int channel, int32_t * arg)
|
|
{
|
|
int32_t value = c->eff_masking_curve_cb[f] - spectrum1;
|
|
|
|
if (value < c->band_masking_cb[band1])
|
|
c->band_masking_cb[band1] = value;
|
|
}
|
|
|
|
static void calc_masking(DCAContext *c, const int32_t *input)
|
|
{
|
|
int i, k, band, ch, ssf;
|
|
int32_t data[512];
|
|
|
|
for (i = 0; i < 256; i++)
|
|
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
|
|
c->masking_curve_cb[ssf][i] = -2047;
|
|
|
|
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++) {
|
|
for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++)
|
|
data[i] = c->history[k][ch];
|
|
for (k -= 512; i < 512; i++, k++)
|
|
data[i] = input[k * c->channels + ch];
|
|
adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
|
|
}
|
|
for (i = 0; i < 256; i++) {
|
|
int32_t m = 2048;
|
|
|
|
for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
|
|
if (c->masking_curve_cb[ssf][i] < m)
|
|
m = c->masking_curve_cb[ssf][i];
|
|
c->eff_masking_curve_cb[i] = m;
|
|
}
|
|
|
|
for (band = 0; band < 32; band++) {
|
|
c->band_masking_cb[band] = 2048;
|
|
walk_band_low(c, band, 0, update_band_masking, NULL);
|
|
walk_band_high(c, band, 0, update_band_masking, NULL);
|
|
}
|
|
}
|
|
|
|
static void find_peaks(DCAContext *c)
|
|
{
|
|
int band, ch;
|
|
|
|
for (band = 0; band < 32; band++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++) {
|
|
int sample;
|
|
int32_t m = 0;
|
|
|
|
for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
|
|
int32_t s = abs(c->subband[sample][band][ch]);
|
|
if (m < s)
|
|
m = s;
|
|
}
|
|
c->peak_cb[band][ch] = get_cb(m);
|
|
}
|
|
|
|
if (c->lfe_channel) {
|
|
int sample;
|
|
int32_t m = 0;
|
|
|
|
for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
|
|
if (m < abs(c->downsampled_lfe[sample]))
|
|
m = abs(c->downsampled_lfe[sample]);
|
|
c->lfe_peak_cb = get_cb(m);
|
|
}
|
|
}
|
|
|
|
static const int snr_fudge = 128;
|
|
#define USED_1ABITS 1
|
|
#define USED_NABITS 2
|
|
#define USED_26ABITS 4
|
|
|
|
static int init_quantization_noise(DCAContext *c, int noise)
|
|
{
|
|
int ch, band, ret = 0;
|
|
|
|
c->consumed_bits = 132 + 493 * c->fullband_channels;
|
|
if (c->lfe_channel)
|
|
c->consumed_bits += 72;
|
|
|
|
/* attempt to guess the bit distribution based on the prevoius frame */
|
|
for (ch = 0; ch < c->fullband_channels; ch++) {
|
|
for (band = 0; band < 32; band++) {
|
|
int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
|
|
|
|
if (snr_cb >= 1312) {
|
|
c->abits[band][ch] = 26;
|
|
ret |= USED_26ABITS;
|
|
} else if (snr_cb >= 222) {
|
|
c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
|
|
ret |= USED_NABITS;
|
|
} else if (snr_cb >= 0) {
|
|
c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
|
|
ret |= USED_NABITS;
|
|
} else {
|
|
c->abits[band][ch] = 1;
|
|
ret |= USED_1ABITS;
|
|
}
|
|
}
|
|
}
|
|
|
|
for (band = 0; band < 32; band++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++) {
|
|
c->consumed_bits += bit_consumption[c->abits[band][ch]];
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void assign_bits(DCAContext *c)
|
|
{
|
|
/* Find the bounds where the binary search should work */
|
|
int low, high, down;
|
|
int used_abits = 0;
|
|
|
|
init_quantization_noise(c, c->worst_quantization_noise);
|
|
low = high = c->worst_quantization_noise;
|
|
if (c->consumed_bits > c->frame_bits) {
|
|
while (c->consumed_bits > c->frame_bits) {
|
|
av_assert0(used_abits != USED_1ABITS);
|
|
low = high;
|
|
high += snr_fudge;
|
|
used_abits = init_quantization_noise(c, high);
|
|
}
|
|
} else {
|
|
while (c->consumed_bits <= c->frame_bits) {
|
|
high = low;
|
|
if (used_abits == USED_26ABITS)
|
|
goto out; /* The requested bitrate is too high, pad with zeros */
|
|
low -= snr_fudge;
|
|
used_abits = init_quantization_noise(c, low);
|
|
}
|
|
}
|
|
|
|
/* Now do a binary search between low and high to see what fits */
|
|
for (down = snr_fudge >> 1; down; down >>= 1) {
|
|
init_quantization_noise(c, high - down);
|
|
if (c->consumed_bits <= c->frame_bits)
|
|
high -= down;
|
|
}
|
|
init_quantization_noise(c, high);
|
|
out:
|
|
c->worst_quantization_noise = high;
|
|
if (high > c->worst_noise_ever)
|
|
c->worst_noise_ever = high;
|
|
}
|
|
|
|
static void shift_history(DCAContext *c, const int32_t *input)
|
|
{
|
|
int k, ch;
|
|
|
|
for (k = 0; k < 512; k++)
|
|
for (ch = 0; ch < c->channels; ch++)
|
|
c->history[k][ch] = input[k * c->channels + ch];
|
|
}
|
|
|
|
static int32_t quantize_value(int32_t value, softfloat quant)
|
|
{
|
|
int32_t offset = 1 << (quant.e - 1);
|
|
|
|
value = mul32(value, quant.m) + offset;
|
|
value = value >> quant.e;
|
|
return value;
|
|
}
|
|
|
|
static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
|
|
{
|
|
int32_t peak;
|
|
int our_nscale, try_remove;
|
|
softfloat our_quant;
|
|
|
|
av_assert0(peak_cb <= 0);
|
|
av_assert0(peak_cb >= -2047);
|
|
|
|
our_nscale = 127;
|
|
peak = cb_to_level[-peak_cb];
|
|
|
|
for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
|
|
if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
|
|
continue;
|
|
our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
|
|
our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
|
|
if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
|
|
continue;
|
|
our_nscale -= try_remove;
|
|
}
|
|
|
|
if (our_nscale >= 125)
|
|
our_nscale = 124;
|
|
|
|
quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
|
|
quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
|
|
av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
|
|
|
|
return our_nscale;
|
|
}
|
|
|
|
static void calc_scales(DCAContext *c)
|
|
{
|
|
int band, ch;
|
|
|
|
for (band = 0; band < 32; band++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
|
|
c->abits[band][ch],
|
|
&c->quant[band][ch]);
|
|
|
|
if (c->lfe_channel)
|
|
c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
|
|
}
|
|
|
|
static void quantize_all(DCAContext *c)
|
|
{
|
|
int sample, band, ch;
|
|
|
|
for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
|
|
for (band = 0; band < 32; band++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);
|
|
}
|
|
|
|
static void put_frame_header(DCAContext *c)
|
|
{
|
|
/* SYNC */
|
|
put_bits(&c->pb, 16, 0x7ffe);
|
|
put_bits(&c->pb, 16, 0x8001);
|
|
|
|
/* Frame type: normal */
|
|
put_bits(&c->pb, 1, 1);
|
|
|
|
/* Deficit sample count: none */
|
|
put_bits(&c->pb, 5, 31);
|
|
|
|
/* CRC is not present */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Number of PCM sample blocks */
|
|
put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
|
|
|
|
/* Primary frame byte size */
|
|
put_bits(&c->pb, 14, c->frame_size - 1);
|
|
|
|
/* Audio channel arrangement */
|
|
put_bits(&c->pb, 6, c->channel_config);
|
|
|
|
/* Core audio sampling frequency */
|
|
put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
|
|
|
|
/* Transmission bit rate */
|
|
put_bits(&c->pb, 5, c->bitrate_index);
|
|
|
|
/* Embedded down mix: disabled */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Embedded dynamic range flag: not present */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Embedded time stamp flag: not present */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Auxiliary data flag: not present */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* HDCD source: no */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Extension audio ID: N/A */
|
|
put_bits(&c->pb, 3, 0);
|
|
|
|
/* Extended audio data: not present */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Audio sync word insertion flag: after each sub-frame */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Low frequency effects flag: not present or 64x subsampling */
|
|
put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
|
|
|
|
/* Predictor history switch flag: on */
|
|
put_bits(&c->pb, 1, 1);
|
|
|
|
/* No CRC */
|
|
/* Multirate interpolator switch: non-perfect reconstruction */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Encoder software revision: 7 */
|
|
put_bits(&c->pb, 4, 7);
|
|
|
|
/* Copy history: 0 */
|
|
put_bits(&c->pb, 2, 0);
|
|
|
|
/* Source PCM resolution: 16 bits, not DTS ES */
|
|
put_bits(&c->pb, 3, 0);
|
|
|
|
/* Front sum/difference coding: no */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Surrounds sum/difference coding: no */
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Dialog normalization: 0 dB */
|
|
put_bits(&c->pb, 4, 0);
|
|
}
|
|
|
|
static void put_primary_audio_header(DCAContext *c)
|
|
{
|
|
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
|
|
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
|
|
|
|
int ch, i;
|
|
/* Number of subframes */
|
|
put_bits(&c->pb, 4, SUBFRAMES - 1);
|
|
|
|
/* Number of primary audio channels */
|
|
put_bits(&c->pb, 3, c->fullband_channels - 1);
|
|
|
|
/* Subband activity count */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
|
|
|
|
/* High frequency VQ start subband */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
|
|
|
|
/* Joint intensity coding index: 0, 0 */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 3, 0);
|
|
|
|
/* Transient mode codebook: A4, A4 (arbitrary) */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 2, 0);
|
|
|
|
/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 3, 6);
|
|
|
|
/* Bit allocation quantizer select: linear 5-bit */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, 3, 6);
|
|
|
|
/* Quantization index codebook select: dummy data
|
|
to avoid transmission of scale factor adjustment */
|
|
for (i = 1; i < 11; i++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
put_bits(&c->pb, bitlen[i], thr[i]);
|
|
|
|
/* Scale factor adjustment index: not transmitted */
|
|
/* Audio header CRC check word: not transmitted */
|
|
}
|
|
|
|
static void put_subframe_samples(DCAContext *c, int ss, int band, int ch)
|
|
{
|
|
if (c->abits[band][ch] <= 7) {
|
|
int sum, i, j;
|
|
for (i = 0; i < 8; i += 4) {
|
|
sum = 0;
|
|
for (j = 3; j >= 0; j--) {
|
|
sum *= quant_levels[c->abits[band][ch]];
|
|
sum += c->quantized[ss * 8 + i + j][band][ch];
|
|
sum += (quant_levels[c->abits[band][ch]] - 1) / 2;
|
|
}
|
|
put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
|
|
}
|
|
} else {
|
|
int i;
|
|
for (i = 0; i < 8; i++) {
|
|
int bits = bit_consumption[c->abits[band][ch]] / 16;
|
|
int32_t mask = (1 << bits) - 1;
|
|
put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void put_subframe(DCAContext *c, int subframe)
|
|
{
|
|
int i, band, ss, ch;
|
|
|
|
/* Subsubframes count */
|
|
put_bits(&c->pb, 2, SUBSUBFRAMES -1);
|
|
|
|
/* Partial subsubframe sample count: dummy */
|
|
put_bits(&c->pb, 3, 0);
|
|
|
|
/* Prediction mode: no ADPCM, in each channel and subband */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
for (band = 0; band < DCA_SUBBANDS; band++)
|
|
put_bits(&c->pb, 1, 0);
|
|
|
|
/* Prediction VQ addres: not transmitted */
|
|
/* Bit allocation index */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
for (band = 0; band < DCA_SUBBANDS; band++)
|
|
put_bits(&c->pb, 5, c->abits[band][ch]);
|
|
|
|
if (SUBSUBFRAMES > 1) {
|
|
/* Transition mode: none for each channel and subband */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
for (band = 0; band < DCA_SUBBANDS; band++)
|
|
put_bits(&c->pb, 1, 0); /* codebook A4 */
|
|
}
|
|
|
|
/* Scale factors */
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
for (band = 0; band < DCA_SUBBANDS; band++)
|
|
put_bits(&c->pb, 7, c->scale_factor[band][ch]);
|
|
|
|
/* 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: 8 samples and scalefactor */
|
|
if (c->lfe_channel) {
|
|
for (i = 0; i < DCA_LFE_SAMPLES; i++)
|
|
put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
|
|
put_bits(&c->pb, 8, c->lfe_scale_factor);
|
|
}
|
|
|
|
/* Audio data (subsubframes) */
|
|
for (ss = 0; ss < SUBSUBFRAMES ; ss++)
|
|
for (ch = 0; ch < c->fullband_channels; ch++)
|
|
for (band = 0; band < DCA_SUBBANDS; band++)
|
|
put_subframe_samples(c, ss, band, ch);
|
|
|
|
/* DSYNC */
|
|
put_bits(&c->pb, 16, 0xffff);
|
|
}
|
|
|
|
static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
DCAContext *c = avctx->priv_data;
|
|
const int32_t *samples;
|
|
int ret, i;
|
|
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 0)
|
|
return ret;
|
|
|
|
samples = (const int32_t *)frame->data[0];
|
|
|
|
subband_transform(c, samples);
|
|
if (c->lfe_channel)
|
|
lfe_downsample(c, samples);
|
|
|
|
calc_masking(c, samples);
|
|
find_peaks(c);
|
|
assign_bits(c);
|
|
calc_scales(c);
|
|
quantize_all(c);
|
|
shift_history(c, samples);
|
|
|
|
init_put_bits(&c->pb, avpkt->data, avpkt->size);
|
|
put_frame_header(c);
|
|
put_primary_audio_header(c);
|
|
for (i = 0; i < SUBFRAMES; i++)
|
|
put_subframe(c, i);
|
|
|
|
flush_put_bits(&c->pb);
|
|
|
|
avpkt->pts = frame->pts;
|
|
avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
|
|
avpkt->size = c->frame_size + 1;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
static const AVCodecDefault defaults[] = {
|
|
{ "b", "1411200" },
|
|
{ NULL },
|
|
};
|
|
|
|
AVCodec ff_dca_encoder = {
|
|
.name = "dca",
|
|
.long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_DTS,
|
|
.priv_data_size = sizeof(DCAContext),
|
|
.init = encode_init,
|
|
.encode2 = encode_frame,
|
|
.capabilities = CODEC_CAP_EXPERIMENTAL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.supported_samplerates = sample_rates,
|
|
.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
|
|
AV_CH_LAYOUT_STEREO,
|
|
AV_CH_LAYOUT_2_2,
|
|
AV_CH_LAYOUT_5POINT0,
|
|
AV_CH_LAYOUT_5POINT1,
|
|
0 },
|
|
.defaults = defaults,
|
|
};
|