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Diffstat (limited to 'libavcodec/dcaenc.c')
| -rw-r--r-- | libavcodec/dcaenc.c | 1259 |
1 files changed, 1259 insertions, 0 deletions
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c new file mode 100644 index 0000000..186997c --- /dev/null +++ b/libavcodec/dcaenc.c @@ -0,0 +1,1259 @@ +/* + * 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 -1; + + /* 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) +{ + int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS; + av_freep(&bufer); +} + +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 (subband_bufer_alloc(c)) + return AVERROR(ENOMEM); + + 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, +}; 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