/* * DCA encoder * Copyright (C) 2008-2012 Alexander E. Patrakov * 2010 Benjamin Larsson * 2011 Xiang Wang * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #define FFT_FLOAT 0 #define FFT_FIXED_32 1 #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "libavutil/ffmath.h" #include "libavutil/opt.h" #include "avcodec.h" #include "dca.h" #include "dcaadpcm.h" #include "dcamath.h" #include "dca_core.h" #include "dcadata.h" #include "dcaenc.h" #include "fft.h" #include "internal.h" #include "mathops.h" #include "put_bits.h" #define MAX_CHANNELS 6 #define DCA_MAX_FRAME_SIZE 16384 #define DCA_HEADER_SIZE 13 #define DCA_LFE_SAMPLES 8 #define DCAENC_SUBBANDS 32 #define SUBFRAMES 1 #define SUBSUBFRAMES 2 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8) #define AUBANDS 25 #define COS_T(x) (c->cos_table[(x) & 2047]) typedef struct CompressionOptions { int adpcm_mode; } CompressionOptions; typedef struct DCAEncContext { AVClass *class; PutBitContext pb; DCAADPCMEncContext adpcm_ctx; FFTContext mdct; CompressionOptions options; int frame_size; int frame_bits; int fullband_channels; int channels; int lfe_channel; int samplerate_index; int bitrate_index; int channel_config; const int32_t *band_interpolation; const int32_t *band_spectrum; int lfe_scale_factor; softfloat lfe_quant; int32_t lfe_peak_cb; const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe int32_t prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS]; int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2]; int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */ int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS]; int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES]; int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal int32_t downsampled_lfe[DCA_LFE_SAMPLES]; int32_t masking_curve_cb[SUBSUBFRAMES][256]; int32_t bit_allocation_sel[MAX_CHANNELS]; int abits[MAX_CHANNELS][DCAENC_SUBBANDS]; int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS]; softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS]; int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS]; int32_t eff_masking_curve_cb[256]; int32_t band_masking_cb[32]; int32_t worst_quantization_noise; int32_t worst_noise_ever; int consumed_bits; int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info int32_t cos_table[2048]; int32_t band_interpolation_tab[2][512]; int32_t band_spectrum_tab[2][8]; int32_t auf[9][AUBANDS][256]; int32_t cb_to_add[256]; int32_t cb_to_level[2048]; int32_t lfe_fir_64i[512]; } DCAEncContext; /* Transfer function of outer and middle ear, Hz -> dB */ static double hom(double f) { double f1 = f / 1000; return -3.64 * pow(f1, -0.8) + 6.8 * exp(-0.6 * (f1 - 3.4) * (f1 - 3.4)) - 6.0 * exp(-0.15 * (f1 - 8.7) * (f1 - 8.7)) - 0.0006 * (f1 * f1) * (f1 * f1); } static double gammafilter(int i, double f) { double h = (f - fc[i]) / erb[i]; h = 1 + h * h; h = 1 / (h * h); return 20 * log10(h); } static int subband_bufer_alloc(DCAEncContext *c) { int ch, band; int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS), sizeof(int32_t)); if (!bufer) return AVERROR(ENOMEM); /* we need a place for DCA_ADPCM_COEFF samples from previous frame * to calc prediction coefficients for each subband */ for (ch = 0; ch < MAX_CHANNELS; ch++) { for (band = 0; band < DCAENC_SUBBANDS; band++) { c->subband[ch][band] = bufer + ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS; } } return 0; } static void subband_bufer_free(DCAEncContext *c) { if (c->subband[0][0]) { int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS; av_free(bufer); c->subband[0][0] = NULL; } } static int encode_init(AVCodecContext *avctx) { DCAEncContext *c = avctx->priv_data; uint64_t layout = avctx->channel_layout; int i, j, k, min_frame_bits; int ret; if ((ret = subband_bufer_alloc(c)) < 0) return ret; c->fullband_channels = c->channels = avctx->channels; c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6); c->band_interpolation = c->band_interpolation_tab[1]; c->band_spectrum = c->band_spectrum_tab[1]; c->worst_quantization_noise = -2047; c->worst_noise_ever = -2047; c->consumed_adpcm_bits = 0; if (ff_dcaadpcm_init(&c->adpcm_ctx)) return AVERROR(ENOMEM); if (!layout) { av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The " "encoder will guess the layout, but it " "might be incorrect.\n"); layout = av_get_default_channel_layout(avctx->channels); } switch (layout) { case AV_CH_LAYOUT_MONO: c->channel_config = 0; break; case AV_CH_LAYOUT_STEREO: c->channel_config = 2; break; case AV_CH_LAYOUT_2_2: c->channel_config = 8; break; case AV_CH_LAYOUT_5POINT0: c->channel_config = 9; break; case AV_CH_LAYOUT_5POINT1: c->channel_config = 9; break; default: av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout!\n"); return AVERROR_PATCHWELCOME; } if (c->lfe_channel) { c->fullband_channels--; c->channel_order_tab = channel_reorder_lfe[c->channel_config]; } else { c->channel_order_tab = channel_reorder_nolfe[c->channel_config]; } for (i = 0; i < MAX_CHANNELS; i++) { for (j = 0; j < DCA_CODE_BOOKS; j++) { c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j]; } /* 6 - no Huffman */ c->bit_allocation_sel[i] = 6; for (j = 0; j < DCAENC_SUBBANDS; j++) { /* -1 - no ADPCM */ c->prediction_mode[i][j] = -1; memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS); } } for (i = 0; i < 9; i++) { if (sample_rates[i] == avctx->sample_rate) break; } if (i == 9) return AVERROR(EINVAL); c->samplerate_index = i; if (avctx->bit_rate < 32000 || avctx->bit_rate > 3840000) { av_log(avctx, AV_LOG_ERROR, "Bit rate %"PRId64" not supported.", avctx->bit_rate); return AVERROR(EINVAL); } for (i = 0; ff_dca_bit_rates[i] < avctx->bit_rate; i++) ; c->bitrate_index = i; c->frame_bits = FFALIGN((avctx->bit_rate * 512 + avctx->sample_rate - 1) / avctx->sample_rate, 32); min_frame_bits = 132 + (493 + 28 * 32) * c->fullband_channels + c->lfe_channel * 72; if (c->frame_bits < min_frame_bits || c->frame_bits > (DCA_MAX_FRAME_SIZE << 3)) return AVERROR(EINVAL); c->frame_size = (c->frame_bits + 7) / 8; avctx->frame_size = 32 * SUBBAND_SAMPLES; if ((ret = ff_mdct_init(&c->mdct, 9, 0, 1.0)) < 0) return ret; /* Init all tables */ c->cos_table[0] = 0x7fffffff; c->cos_table[512] = 0; c->cos_table[1024] = -c->cos_table[0]; for (i = 1; i < 512; i++) { c->cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024)); c->cos_table[1024-i] = -c->cos_table[i]; c->cos_table[1024+i] = -c->cos_table[i]; c->cos_table[2048-i] = +c->cos_table[i]; } for (i = 0; i < 2048; i++) c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i)); for (k = 0; k < 32; k++) { for (j = 0; j < 8; j++) { c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]); c->lfe_fir_64i[64 * (7-j) + (63 - k)] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]); } } for (i = 0; i < 512; i++) { c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]); c->band_interpolation_tab[1][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_nonperfect[i]); } for (i = 0; i < 9; i++) { for (j = 0; j < AUBANDS; j++) { for (k = 0; k < 256; k++) { double freq = sample_rates[i] * (k + 0.5) / 512; c->auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq))); } } } for (i = 0; i < 256; i++) { double add = 1 + ff_exp10(-0.01 * i); c->cb_to_add[i] = (int32_t)(100 * log10(add)); } for (j = 0; j < 8; j++) { double accum = 0; for (i = 0; i < 512; i++) { double reconst = ff_dca_fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1); accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512); } c->band_spectrum_tab[0][j] = (int32_t)(200 * log10(accum)); } for (j = 0; j < 8; j++) { double accum = 0; for (i = 0; i < 512; i++) { double reconst = ff_dca_fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1); accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512); } c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum)); } return 0; } static av_cold int encode_close(AVCodecContext *avctx) { DCAEncContext *c = avctx->priv_data; ff_mdct_end(&c->mdct); subband_bufer_free(c); ff_dcaadpcm_free(&c->adpcm_ctx); return 0; } static void subband_transform(DCAEncContext *c, const int32_t *input) { int ch, subs, i, k, j; for (ch = 0; ch < c->fullband_channels; ch++) { /* History is copied because it is also needed for PSY */ int32_t hist[512]; int hist_start = 0; const int chi = c->channel_order_tab[ch]; memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t)); for (subs = 0; subs < SUBBAND_SAMPLES; subs++) { int32_t accum[64]; int32_t resp; int band; /* Calculate the convolutions at once */ memset(accum, 0, 64 * sizeof(int32_t)); for (k = 0, i = hist_start, j = 0; i < 512; k = (k + 1) & 63, i++, j++) accum[k] += mul32(hist[i], c->band_interpolation[j]); for (i = 0; i < hist_start; k = (k + 1) & 63, i++, j++) accum[k] += mul32(hist[i], c->band_interpolation[j]); for (k = 16; k < 32; k++) accum[k] = accum[k] - accum[31 - k]; for (k = 32; k < 48; k++) accum[k] = accum[k] + accum[95 - k]; for (band = 0; band < 32; band++) { resp = 0; for (i = 16; i < 48; i++) { int s = (2 * band + 1) * (2 * (i + 16) + 1); resp += mul32(accum[i], COS_T(s << 3)) >> 3; } c->subband[ch][band][subs] = ((band + 1) & 2) ? -resp : resp; } /* Copy in 32 new samples from input */ for (i = 0; i < 32; i++) hist[i + hist_start] = input[(subs * 32 + i) * c->channels + chi]; hist_start = (hist_start + 32) & 511; } } } static void lfe_downsample(DCAEncContext *c, const int32_t *input) { /* FIXME: make 128x LFE downsampling possible */ const int lfech = lfe_index[c->channel_config]; int i, j, lfes; int32_t hist[512]; int32_t accum; int hist_start = 0; memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t)); for (lfes = 0; lfes < DCA_LFE_SAMPLES; lfes++) { /* Calculate the convolution */ accum = 0; for (i = hist_start, j = 0; i < 512; i++, j++) accum += mul32(hist[i], c->lfe_fir_64i[j]); for (i = 0; i < hist_start; i++, j++) accum += mul32(hist[i], c->lfe_fir_64i[j]); c->downsampled_lfe[lfes] = accum; /* Copy in 64 new samples from input */ for (i = 0; i < 64; i++) hist[i + hist_start] = input[(lfes * 64 + i) * c->channels + lfech]; hist_start = (hist_start + 64) & 511; } } static int32_t get_cb(DCAEncContext *c, int32_t in) { int i, res = 0; in = FFABS(in); for (i = 1024; i > 0; i >>= 1) { if (c->cb_to_level[i + res] >= in) res += i; } return -res; } static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b) { if (a < b) FFSWAP(int32_t, a, b); if (a - b >= 256) return a; return a + c->cb_to_add[a - b]; } static void calc_power(DCAEncContext *c, const int32_t in[2 * 256], int32_t power[256]) { int i; LOCAL_ALIGNED_32(int32_t, data, [512]); LOCAL_ALIGNED_32(int32_t, coeff, [256]); for (i = 0; i < 512; i++) data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4); c->mdct.mdct_calc(&c->mdct, coeff, data); for (i = 0; i < 256; i++) { const int32_t cb = get_cb(c, coeff[i]); power[i] = add_cb(c, cb, cb); } } static void adjust_jnd(DCAEncContext *c, const int32_t in[512], int32_t out_cb[256]) { int32_t power[256]; int32_t out_cb_unnorm[256]; int32_t denom; const int32_t ca_cb = -1114; const int32_t cs_cb = 928; const int samplerate_index = c->samplerate_index; int i, j; calc_power(c, in, power); for (j = 0; j < 256; j++) out_cb_unnorm[j] = -2047; /* and can only grow */ for (i = 0; i < AUBANDS; i++) { denom = ca_cb; /* and can only grow */ for (j = 0; j < 256; j++) denom = add_cb(c, denom, power[j] + c->auf[samplerate_index][i][j]); for (j = 0; j < 256; j++) out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j], -denom + c->auf[samplerate_index][i][j]); } for (j = 0; j < 256; j++) out_cb[j] = add_cb(c, out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb); } typedef void (*walk_band_t)(DCAEncContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t * arg); static void walk_band_low(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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(DCAEncContext *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++) { const int chi = c->channel_order_tab[ch]; for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++) data[i] = c->history[ch][k]; for (k -= 512; i < 512; i++, k++) data[i] = input[k * c->channels + chi]; adjust_jnd(c, 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 inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len) { int sample; int32_t m = 0; for (sample = 0; sample < len; sample++) { int32_t s = abs(in[sample]); if (m < s) m = s; } return get_cb(c, m); } static void find_peaks(DCAEncContext *c) { int band, ch; for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band], SUBBAND_SAMPLES); } if (c->lfe_channel) c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES); } static void adpcm_analysis(DCAEncContext *c) { int ch, band; int pred_vq_id; int32_t *samples; int32_t estimated_diff[SUBBAND_SAMPLES]; c->consumed_adpcm_bits = 0; for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) { samples = c->subband[ch][band] - DCA_ADPCM_COEFFS; pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples, SUBBAND_SAMPLES, estimated_diff); if (pred_vq_id >= 0) { c->prediction_mode[ch][band] = pred_vq_id; c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16); } else { c->prediction_mode[ch][band] = -1; } } } } static const int snr_fudge = 128; #define USED_1ABITS 1 #define USED_26ABITS 4 static inline int32_t get_step_size(DCAEncContext *c, int ch, int band) { int32_t step_size; if (c->bitrate_index == 3) step_size = ff_dca_lossless_quant[c->abits[ch][band]]; else step_size = ff_dca_lossy_quant[c->abits[ch][band]]; return step_size; } static int calc_one_scale(DCAEncContext *c, 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 = c->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 ((ff_dca_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((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant)); return our_nscale; } static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band) { int32_t step_size; int32_t diff_peak_cb = c->diff_peak_cb[ch][band]; c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb, c->abits[ch][band], &c->quant[ch][band]); step_size = get_step_size(c, ch, band); ff_dcaadpcm_do_real(c->prediction_mode[ch][band], c->quant[ch][band], ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], step_size, c->adpcm_history[ch][band], c->subband[ch][band], c->adpcm_history[ch][band] + 4, c->quantized[ch][band], SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]); } static void quantize_adpcm(DCAEncContext *c) { int band, ch; for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < 32; band++) if (c->prediction_mode[ch][band] >= 0) quantize_adpcm_subband(c, ch, band); } static void quantize_pcm(DCAEncContext *c) { int sample, band, ch; for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) { if (c->prediction_mode[ch][band] == -1) { for (sample = 0; sample < SUBBAND_SAMPLES; sample++) { int32_t val = quantize_value(c->subband[ch][band][sample], c->quant[ch][band]); c->quantized[ch][band][sample] = val; } } } } } static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, uint32_t *result) { uint8_t sel, id = abits - 1; for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++) result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES, sel, id); } static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7], uint32_t clc_bits[DCA_CODE_BOOKS], int32_t res[DCA_CODE_BOOKS]) { uint8_t i, sel; uint32_t best_sel_bits[DCA_CODE_BOOKS]; int32_t best_sel_id[DCA_CODE_BOOKS]; uint32_t t, bits = 0; for (i = 0; i < DCA_CODE_BOOKS; i++) { av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i]))); if (vlc_bits[i][0] == 0) { /* do not transmit adjustment index for empty codebooks */ res[i] = ff_dca_quant_index_group_size[i]; /* and skip it */ continue; } best_sel_bits[i] = vlc_bits[i][0]; best_sel_id[i] = 0; for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) { if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) { best_sel_bits[i] = vlc_bits[i][sel]; best_sel_id[i] = sel; } } /* 2 bits to transmit scale factor adjustment index */ t = best_sel_bits[i] + 2; if (t < clc_bits[i]) { res[i] = best_sel_id[i]; bits += t; } else { res[i] = ff_dca_quant_index_group_size[i]; bits += clc_bits[i]; } } return bits; } static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands, int32_t *res) { uint8_t i; uint32_t t; int32_t best_sel = 6; int32_t best_bits = bands * 5; /* Check do we have subband which cannot be encoded by Huffman tables */ for (i = 0; i < bands; i++) { if (abits[i] > 12 || abits[i] == 0) { *res = best_sel; return best_bits; } } for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) { t = ff_dca_vlc_calc_alloc_bits(abits, bands, i); if (t < best_bits) { best_bits = t; best_sel = i; } } *res = best_sel; return best_bits; } static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero) { int ch, band, ret = USED_26ABITS | USED_1ABITS; uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7]; uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS]; uint32_t bits_counter = 0; c->consumed_bits = 132 + 333 * c->fullband_channels; c->consumed_bits += c->consumed_adpcm_bits; 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[ch][band] - c->band_masking_cb[band] - noise; if (snr_cb >= 1312) { c->abits[ch][band] = 26; ret &= ~USED_1ABITS; } else if (snr_cb >= 222) { c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000); ret &= ~(USED_26ABITS | USED_1ABITS); } else if (snr_cb >= 0) { c->abits[ch][band] = 2 + mul32(snr_cb, 106000000); ret &= ~(USED_26ABITS | USED_1ABITS); } else if (forbid_zero || snr_cb >= -140) { c->abits[ch][band] = 1; ret &= ~USED_26ABITS; } else { c->abits[ch][band] = 0; ret &= ~(USED_26ABITS | USED_1ABITS); } } c->consumed_bits += set_best_abits_code(c->abits[ch], 32, &c->bit_allocation_sel[ch]); } /* Recalc scale_factor each time to get bits consumption in case of Huffman coding. It is suboptimal solution */ /* TODO: May be cache scaled values */ for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) { if (c->prediction_mode[ch][band] == -1) { c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band], c->abits[ch][band], &c->quant[ch][band]); } } } quantize_adpcm(c); quantize_pcm(c); memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t)); memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t)); for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) { if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) { accumulate_huff_bit_consumption(c->abits[ch][band], c->quantized[ch][band], huff_bit_count_accum[ch][c->abits[ch][band] - 1]); clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]]; } else { bits_counter += bit_consumption[c->abits[ch][band]]; } } } for (ch = 0; ch < c->fullband_channels; ch++) { bits_counter += set_best_code(huff_bit_count_accum[ch], clc_bit_count_accum[ch], c->quant_index_sel[ch]); } c->consumed_bits += bits_counter; 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; int forbid_zero = 1; restart: init_quantization_noise(c, c->worst_quantization_noise, forbid_zero); low = high = c->worst_quantization_noise; if (c->consumed_bits > c->frame_bits) { while (c->consumed_bits > c->frame_bits) { if (used_abits == USED_1ABITS && forbid_zero) { forbid_zero = 0; goto restart; } low = high; high += snr_fudge; used_abits = init_quantization_noise(c, high, forbid_zero); } } 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, forbid_zero); } } /* 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, forbid_zero); if (c->consumed_bits <= c->frame_bits) high -= down; } init_quantization_noise(c, high, forbid_zero); 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++) { const int chi = c->channel_order_tab[ch]; c->history[ch][k] = input[k * c->channels + chi]; } } static void fill_in_adpcm_bufer(DCAEncContext *c) { int ch, band; int32_t step_size; /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded * in current frame - we need this data if subband of next frame is * ADPCM */ for (ch = 0; ch < c->channels; ch++) { for (band = 0; band < 32; band++) { int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS; if (c->prediction_mode[ch][band] == -1) { step_size = get_step_size(c, ch, band); ff_dca_core_dequantize(c->adpcm_history[ch][band], c->quantized[ch][band]+12, step_size, ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4); } else { AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4); } /* Copy dequantized values for LPC analysis. * It reduces artifacts in case of extreme quantization, * example: in current frame abits is 1 and has no prediction flag, * but end of this frame is sine like signal. In this case, if LPC analysis uses * original values, likely LPC analysis returns good prediction gain, and sets prediction flag. * But there are no proper value in decoder history, so likely result will be no good. * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands */ samples[0] = c->adpcm_history[ch][band][0] << 7; samples[1] = c->adpcm_history[ch][band][1] << 7; samples[2] = c->adpcm_history[ch][band][2] << 7; samples[3] = c->adpcm_history[ch][band][3] << 7; } } } static void calc_lfe_scales(DCAEncContext *c) { if (c->lfe_channel) c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant); } 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) { 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, c->bit_allocation_sel[ch]); /* Quantization index codebook select */ for (i = 0; i < DCA_CODE_BOOKS; i++) for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]); /* Scale factor adjustment index: transmitted in case of Huffman coding */ for (i = 0; i < DCA_CODE_BOOKS; i++) for (ch = 0; ch < c->fullband_channels; ch++) if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i]) put_bits(&c->pb, 2, 0); /* Audio header CRC check word: not transmitted */ } static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch) { int i, j, sum, bits, sel; if (c->abits[ch][band] <= DCA_CODE_BOOKS) { av_assert0(c->abits[ch][band] > 0); sel = c->quant_index_sel[ch][c->abits[ch][band] - 1]; // Huffman codes if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) { ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8, sel, c->abits[ch][band] - 1); return; } // Block codes if (c->abits[ch][band] <= 7) { for (i = 0; i < 8; i += 4) { sum = 0; for (j = 3; j >= 0; j--) { sum *= ff_dca_quant_levels[c->abits[ch][band]]; sum += c->quantized[ch][band][ss * 8 + i + j]; sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2; } put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum); } return; } } for (i = 0; i < 8; i++) { bits = bit_consumption[c->abits[ch][band]] / 16; put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]); } } 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, !(c->prediction_mode[ch][band] == -1)); /* Prediction VQ address */ for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCAENC_SUBBANDS; band++) if (c->prediction_mode[ch][band] >= 0) put_bits(&c->pb, 12, c->prediction_mode[ch][band]); /* Bit allocation index */ for (ch = 0; ch < c->fullband_channels; ch++) { if (c->bit_allocation_sel[ch] == 6) { for (band = 0; band < DCAENC_SUBBANDS; band++) { put_bits(&c->pb, 5, c->abits[ch][band]); } } else { ff_dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS, c->bit_allocation_sel[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++) if (c->abits[ch][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++) if (c->abits[ch][band]) put_bits(&c->pb, 7, c->scale_factor[ch][band]); /* 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++) if (c->abits[ch][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); if (c->options.adpcm_mode) adpcm_analysis(c); find_peaks(c); assign_bits(c); calc_lfe_scales(c); shift_history(c, samples); init_put_bits(&c->pb, avpkt->data, avpkt->size); fill_in_adpcm_bufer(c); 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 = put_bits_count(&c->pb) >> 3; *got_packet_ptr = 1; return 0; } #define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM static const AVOption options[] = { { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS }, { NULL }, }; static const AVClass dcaenc_class = { .class_name = "DCA (DTS Coherent Acoustics)", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; 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, .close = encode_close, .encode2 = encode_frame, .capabilities = AV_CODEC_CAP_EXPERIMENTAL, .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, .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, .priv_class = &dcaenc_class, };