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30 KiB
978 lines
30 KiB
/* |
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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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|
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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 "mathops.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 DCAENC_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 DCAEncContext { |
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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][DCAENC_SUBBANDS][MAX_CHANNELS]; |
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int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS]; |
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int32_t peak_cb[DCAENC_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[DCAENC_SUBBANDS][MAX_CHANNELS]; |
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int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS]; |
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softfloat quant[DCAENC_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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} DCAEncContext; |
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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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DCAEncContext *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; ff_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 = ff_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 * ff_dca_lfe_fir_64[i]); |
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lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * ff_dca_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 * ff_dca_fir_32bands_perfect[i]); |
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band_interpolation[1][i] = (int32_t)(0x1000000000ULL * ff_dca_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 = ff_dca_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 = ff_dca_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(DCAEncContext *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(DCAEncContext *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)(DCAEncContext *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(DCAEncContext *c, int band, int channel, |
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walk_band_t walk, int32_t *arg) |
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{ |
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int f; |
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|
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if (band == 0) { |
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for (f = 0; f < 4; f++) |
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walk(c, 0, 0, f, 0, -2047, channel, arg); |
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} else { |
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for (f = 0; f < 8; f++) |
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walk(c, band, band - 1, 8 * band - 4 + f, |
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c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg); |
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} |
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} |
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static void walk_band_high(DCAEncContext *c, int band, int channel, |
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walk_band_t walk, int32_t *arg) |
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{ |
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int f; |
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|
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if (band == 31) { |
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for (f = 0; f < 4; f++) |
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walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg); |
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} else { |
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for (f = 0; f < 8; f++) |
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walk(c, band, band + 1, 8 * band + 4 + f, |
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c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg); |
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} |
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} |
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static void update_band_masking(DCAEncContext *c, int band1, int band2, |
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int f, int32_t spectrum1, int32_t spectrum2, |
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int channel, int32_t * arg) |
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{ |
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int32_t value = c->eff_masking_curve_cb[f] - spectrum1; |
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if (value < c->band_masking_cb[band1]) |
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c->band_masking_cb[band1] = value; |
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} |
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static void calc_masking(DCAEncContext *c, const int32_t *input) |
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{ |
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int i, k, band, ch, ssf; |
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int32_t data[512]; |
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|
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for (i = 0; i < 256; i++) |
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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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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, DCAENC_SUBBANDS - 2); |
|
|
|
/* High frequency VQ start subband */ |
|
for (ch = 0; ch < c->fullband_channels; ch++) |
|
put_bits(&c->pb, 5, DCAENC_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(DCAEncContext *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; |
|
put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]); |
|
} |
|
} |
|
} |
|
|
|
static void put_subframe(DCAEncContext *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 < DCAENC_SUBBANDS; band++) |
|
put_bits(&c->pb, 1, 0); |
|
|
|
/* Prediction VQ address: not transmitted */ |
|
/* Bit allocation index */ |
|
for (ch = 0; ch < c->fullband_channels; ch++) |
|
for (band = 0; band < DCAENC_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 < DCAENC_SUBBANDS; band++) |
|
put_bits(&c->pb, 1, 0); /* codebook A4 */ |
|
} |
|
|
|
/* Scale factors */ |
|
for (ch = 0; ch < c->fullband_channels; ch++) |
|
for (band = 0; band < DCAENC_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 < DCAENC_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) |
|
{ |
|
DCAEncContext *c = avctx->priv_data; |
|
const int32_t *samples; |
|
int ret, i; |
|
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size , 0)) < 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); |
|
|
|
|
|
for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++) |
|
put_bits(&c->pb, 1, 0); |
|
|
|
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(DCAEncContext), |
|
.init = encode_init, |
|
.encode2 = encode_frame, |
|
.capabilities = AV_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, |
|
};
|
|
|