/* * 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 */ #include "libavutil/avassert.h" #include "libavutil/channel_layout.h" #include "libavutil/common.h" #include "avcodec.h" #include "dca.h" #include "dcadata.h" #include "dcaenc.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 DCA_SUBBANDS 32 #define SUBFRAMES 1 #define SUBSUBFRAMES 2 #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8) #define AUBANDS 25 typedef struct DCAContext { PutBitContext pb; 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; int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */ int32_t subband[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS]; int32_t quantized[SUBBAND_SAMPLES][DCA_SUBBANDS][MAX_CHANNELS]; int32_t peak_cb[DCA_SUBBANDS][MAX_CHANNELS]; int32_t downsampled_lfe[DCA_LFE_SAMPLES]; int32_t masking_curve_cb[SUBSUBFRAMES][256]; int abits[DCA_SUBBANDS][MAX_CHANNELS]; int scale_factor[DCA_SUBBANDS][MAX_CHANNELS]; softfloat quant[DCA_SUBBANDS][MAX_CHANNELS]; 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; } DCAContext; static int32_t cos_table[2048]; static int32_t band_interpolation[2][512]; static int32_t band_spectrum[2][8]; static int32_t auf[9][AUBANDS][256]; static int32_t cb_to_add[256]; static int32_t cb_to_level[2048]; static int32_t lfe_fir_64i[512]; /* 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 encode_init(AVCodecContext *avctx) { DCAContext *c = avctx->priv_data; uint64_t layout = avctx->channel_layout; int i, min_frame_bits; c->fullband_channels = c->channels = avctx->channels; c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6); c->band_interpolation = band_interpolation[1]; c->band_spectrum = band_spectrum[1]; c->worst_quantization_noise = -2047; c->worst_noise_ever = -2047; 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--; 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 %i not supported.", avctx->bit_rate); return AVERROR(EINVAL); } for (i = 0; dca_bit_rates[i] < avctx->bit_rate; i++) ; c->bitrate_index = i; avctx->bit_rate = dca_bit_rates[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 (!cos_table[0]) { int j, k; for (i = 0; i < 2048; i++) { cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024)); cb_to_level[i] = (int32_t)(0x7fffffff * pow(10, -0.005 * i)); } /* FIXME: probably incorrect */ for (i = 0; i < 256; i++) { lfe_fir_64i[i] = (int32_t)(0x01ffffff * lfe_fir_64[i]); lfe_fir_64i[511 - i] = (int32_t)(0x01ffffff * lfe_fir_64[i]); } for (i = 0; i < 512; i++) { band_interpolation[0][i] = (int32_t)(0x1000000000ULL * fir_32bands_perfect[i]); band_interpolation[1][i] = (int32_t)(0x1000000000ULL * 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; auf[i][j][k] = (int32_t)(10 * (hom(freq) + gammafilter(j, freq))); } } } for (i = 0; i < 256; i++) { double add = 1 + pow(10, -0.01 * i); 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 = fir_32bands_perfect[i] * ((i & 64) ? (-1) : 1); accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512); } band_spectrum[0][j] = (int32_t)(200 * log10(accum)); } for (j = 0; j < 8; j++) { double accum = 0; for (i = 0; i < 512; i++) { double reconst = fir_32bands_nonperfect[i] * ((i & 64) ? (-1) : 1); accum += reconst * cos(2 * M_PI * (i + 0.5 - 256) * (j + 0.5) / 512); } band_spectrum[1][j] = (int32_t)(200 * log10(accum)); } } return 0; } static inline int32_t cos_t(int x) { return cos_table[x & 2047]; } static inline int32_t sin_t(int x) { return cos_t(x - 512); } static inline int32_t half32(int32_t a) { return (a + 1) >> 1; } static inline int32_t mul32(int32_t a, int32_t b) { int64_t r = (int64_t)a * b + 0x80000000ULL; return r >> 32; } static void subband_transform(DCAContext *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; for (i = 0; i < 512; i++) hist[i] = c->history[i][ch]; for (subs = 0; subs < SUBBAND_SAMPLES; subs++) { int32_t accum[64]; int32_t resp; int band; /* Calculate the convolutions at once */ for (i = 0; i < 64; i++) accum[i] = 0; 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[subs][band][ch] = ((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 + ch]; hist_start = (hist_start + 32) & 511; } } } static void lfe_downsample(DCAContext *c, const int32_t *input) { /* FIXME: make 128x LFE downsampling possible */ int i, j, lfes; int32_t hist[512]; int32_t accum; int hist_start = 0; for (i = 0; i < 512; i++) hist[i] = c->history[i][c->channels - 1]; 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], lfe_fir_64i[j]); for (i = 0; i < hist_start; i++, j++) accum += mul32(hist[i], 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 + c->channels - 1]; hist_start = (hist_start + 64) & 511; } } typedef struct { int32_t re; int32_t im; } cplx32; static void fft(const int32_t in[2 * 256], cplx32 out[256]) { cplx32 buf[256], rin[256], rout[256]; int i, j, k, l; /* do two transforms in parallel */ for (i = 0; i < 256; i++) { /* Apply the Hann window */ rin[i].re = mul32(in[2 * i], 0x3fffffff - (cos_t(8 * i + 2) >> 1)); rin[i].im = mul32(in[2 * i + 1], 0x3fffffff - (cos_t(8 * i + 6) >> 1)); } /* pre-rotation */ for (i = 0; i < 256; i++) { buf[i].re = mul32(cos_t(4 * i + 2), rin[i].re) - mul32(sin_t(4 * i + 2), rin[i].im); buf[i].im = mul32(cos_t(4 * i + 2), rin[i].im) + mul32(sin_t(4 * i + 2), rin[i].re); } for (j = 256, l = 1; j != 1; j >>= 1, l <<= 1) { for (k = 0; k < 256; k += j) { for (i = k; i < k + j / 2; i++) { cplx32 sum, diff; int t = 8 * l * i; sum.re = buf[i].re + buf[i + j / 2].re; sum.im = buf[i].im + buf[i + j / 2].im; diff.re = buf[i].re - buf[i + j / 2].re; diff.im = buf[i].im - buf[i + j / 2].im; buf[i].re = half32(sum.re); buf[i].im = half32(sum.im); buf[i + j / 2].re = mul32(diff.re, cos_t(t)) - mul32(diff.im, sin_t(t)); buf[i + j / 2].im = mul32(diff.im, cos_t(t)) + mul32(diff.re, sin_t(t)); } } } /* post-rotation */ for (i = 0; i < 256; i++) { int b = ff_reverse[i]; rout[i].re = mul32(buf[b].re, cos_t(4 * i)) - mul32(buf[b].im, sin_t(4 * i)); rout[i].im = mul32(buf[b].im, cos_t(4 * i)) + mul32(buf[b].re, sin_t(4 * i)); } for (i = 0; i < 256; i++) { /* separate the results of the two transforms */ cplx32 o1, o2; o1.re = rout[i].re - rout[255 - i].re; o1.im = rout[i].im + rout[255 - i].im; o2.re = rout[i].im - rout[255 - i].im; o2.im = -rout[i].re - rout[255 - i].re; /* combine them into one long transform */ out[i].re = mul32( o1.re + o2.re, cos_t(2 * i + 1)) + mul32( o1.im - o2.im, sin_t(2 * i + 1)); out[i].im = mul32( o1.im + o2.im, cos_t(2 * i + 1)) + mul32(-o1.re + o2.re, sin_t(2 * i + 1)); } } static int32_t get_cb(int32_t in) { int i, res; res = 0; if (in < 0) in = -in; for (i = 1024; i > 0; i >>= 1) { if (cb_to_level[i + res] >= in) res += i; } return -res; } static int32_t add_cb(int32_t a, int32_t b) { if (a < b) FFSWAP(int32_t, a, b); if (a - b >= 256) return a; return a + cb_to_add[a - b]; } static void adjust_jnd(int samplerate_index, const int32_t in[512], int32_t out_cb[256]) { int32_t power[256]; cplx32 out[256]; int32_t out_cb_unnorm[256]; int32_t denom; const int32_t ca_cb = -1114; const int32_t cs_cb = 928; int i, j; fft(in, out); for (j = 0; j < 256; j++) { power[j] = add_cb(get_cb(out[j].re), get_cb(out[j].im)); 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(denom, power[j] + auf[samplerate_index][i][j]); for (j = 0; j < 256; j++) out_cb_unnorm[j] = add_cb(out_cb_unnorm[j], -denom + auf[samplerate_index][i][j]); } for (j = 0; j < 256; j++) out_cb[j] = add_cb(out_cb[j], -out_cb_unnorm[j] - ca_cb - cs_cb); } typedef void (*walk_band_t)(DCAContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t * arg); static void walk_band_low(DCAContext *c, int band, int channel, walk_band_t walk, int32_t *arg) { int f; if (band == 0) { for (f = 0; f < 4; f++) walk(c, 0, 0, f, 0, -2047, channel, arg); } else { for (f = 0; f < 8; f++) walk(c, band, band - 1, 8 * band - 4 + f, c->band_spectrum[7 - f], c->band_spectrum[f], channel, arg); } } static void walk_band_high(DCAContext *c, int band, int channel, walk_band_t walk, int32_t *arg) { int f; if (band == 31) { for (f = 0; f < 4; f++) walk(c, 31, 31, 256 - 4 + f, 0, -2047, channel, arg); } else { for (f = 0; f < 8; f++) walk(c, band, band + 1, 8 * band + 4 + f, c->band_spectrum[f], c->band_spectrum[7 - f], channel, arg); } } static void update_band_masking(DCAContext *c, int band1, int band2, int f, int32_t spectrum1, int32_t spectrum2, int channel, int32_t * arg) { int32_t value = c->eff_masking_curve_cb[f] - spectrum1; if (value < c->band_masking_cb[band1]) c->band_masking_cb[band1] = value; } static void calc_masking(DCAContext *c, const int32_t *input) { int i, k, band, ch, ssf; int32_t data[512]; for (i = 0; i < 256; i++) for (ssf = 0; ssf < SUBSUBFRAMES; ssf++) c->masking_curve_cb[ssf][i] = -2047; for (ssf = 0; ssf < SUBSUBFRAMES; ssf++) for (ch = 0; ch < c->fullband_channels; ch++) { for (i = 0, k = 128 + 256 * ssf; k < 512; i++, k++) data[i] = c->history[k][ch]; for (k -= 512; i < 512; i++, k++) data[i] = input[k * c->channels + ch]; adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]); } for (i = 0; i < 256; i++) { int32_t m = 2048; for (ssf = 0; ssf < SUBSUBFRAMES; ssf++) if (c->masking_curve_cb[ssf][i] < m) m = c->masking_curve_cb[ssf][i]; c->eff_masking_curve_cb[i] = m; } for (band = 0; band < 32; band++) { c->band_masking_cb[band] = 2048; walk_band_low(c, band, 0, update_band_masking, NULL); walk_band_high(c, band, 0, update_band_masking, NULL); } } static void find_peaks(DCAContext *c) { int band, ch; for (band = 0; band < 32; band++) for (ch = 0; ch < c->fullband_channels; ch++) { int sample; int32_t m = 0; for (sample = 0; sample < SUBBAND_SAMPLES; sample++) { int32_t s = abs(c->subband[sample][band][ch]); if (m < s) m = s; } c->peak_cb[band][ch] = get_cb(m); } if (c->lfe_channel) { int sample; int32_t m = 0; for (sample = 0; sample < DCA_LFE_SAMPLES; sample++) if (m < abs(c->downsampled_lfe[sample])) m = abs(c->downsampled_lfe[sample]); c->lfe_peak_cb = get_cb(m); } } static const int snr_fudge = 128; #define USED_1ABITS 1 #define USED_NABITS 2 #define USED_26ABITS 4 static int init_quantization_noise(DCAContext *c, int noise) { int ch, band, ret = 0; c->consumed_bits = 132 + 493 * c->fullband_channels; if (c->lfe_channel) c->consumed_bits += 72; /* attempt to guess the bit distribution based on the prevoius frame */ for (ch = 0; ch < c->fullband_channels; ch++) { for (band = 0; band < 32; band++) { int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise; if (snr_cb >= 1312) { c->abits[band][ch] = 26; ret |= USED_26ABITS; } else if (snr_cb >= 222) { c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000); ret |= USED_NABITS; } else if (snr_cb >= 0) { c->abits[band][ch] = 2 + mul32(snr_cb, 106000000); ret |= USED_NABITS; } else { c->abits[band][ch] = 1; ret |= USED_1ABITS; } } } for (band = 0; band < 32; band++) for (ch = 0; ch < c->fullband_channels; ch++) { c->consumed_bits += bit_consumption[c->abits[band][ch]]; } return ret; } static void assign_bits(DCAContext *c) { /* Find the bounds where the binary search should work */ int low, high, down; int used_abits = 0; init_quantization_noise(c, c->worst_quantization_noise); low = high = c->worst_quantization_noise; if (c->consumed_bits > c->frame_bits) { while (c->consumed_bits > c->frame_bits) { av_assert0(used_abits != USED_1ABITS); low = high; high += snr_fudge; used_abits = init_quantization_noise(c, high); } } else { while (c->consumed_bits <= c->frame_bits) { high = low; if (used_abits == USED_26ABITS) goto out; /* The requested bitrate is too high, pad with zeros */ low -= snr_fudge; used_abits = init_quantization_noise(c, low); } } /* Now do a binary search between low and high to see what fits */ for (down = snr_fudge >> 1; down; down >>= 1) { init_quantization_noise(c, high - down); if (c->consumed_bits <= c->frame_bits) high -= down; } init_quantization_noise(c, high); out: c->worst_quantization_noise = high; if (high > c->worst_noise_ever) c->worst_noise_ever = high; } static void shift_history(DCAContext *c, const int32_t *input) { int k, ch; for (k = 0; k < 512; k++) for (ch = 0; ch < c->channels; ch++) c->history[k][ch] = input[k * c->channels + ch]; } static int32_t quantize_value(int32_t value, softfloat quant) { int32_t offset = 1 << (quant.e - 1); value = mul32(value, quant.m) + offset; value = value >> quant.e; return value; } static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant) { int32_t peak; int our_nscale, try_remove; softfloat our_quant; av_assert0(peak_cb <= 0); av_assert0(peak_cb >= -2047); our_nscale = 127; peak = cb_to_level[-peak_cb]; for (try_remove = 64; try_remove > 0; try_remove >>= 1) { if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17) continue; our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m); our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17; if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant)) continue; our_nscale -= try_remove; } if (our_nscale >= 125) our_nscale = 124; quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m); quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17; av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant)); return our_nscale; } static void calc_scales(DCAContext *c) { int band, ch; for (band = 0; band < 32; band++) for (ch = 0; ch < c->fullband_channels; ch++) c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch], c->abits[band][ch], &c->quant[band][ch]); if (c->lfe_channel) c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant); } static void quantize_all(DCAContext *c) { int sample, band, ch; for (sample = 0; sample < SUBBAND_SAMPLES; sample++) for (band = 0; band < 32; band++) for (ch = 0; ch < c->fullband_channels; ch++) c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]); } static void put_frame_header(DCAContext *c) { /* SYNC */ put_bits(&c->pb, 16, 0x7ffe); put_bits(&c->pb, 16, 0x8001); /* Frame type: normal */ put_bits(&c->pb, 1, 1); /* Deficit sample count: none */ put_bits(&c->pb, 5, 31); /* CRC is not present */ put_bits(&c->pb, 1, 0); /* Number of PCM sample blocks */ put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1); /* Primary frame byte size */ put_bits(&c->pb, 14, c->frame_size - 1); /* Audio channel arrangement */ put_bits(&c->pb, 6, c->channel_config); /* Core audio sampling frequency */ put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]); /* Transmission bit rate */ put_bits(&c->pb, 5, c->bitrate_index); /* Embedded down mix: disabled */ put_bits(&c->pb, 1, 0); /* Embedded dynamic range flag: not present */ put_bits(&c->pb, 1, 0); /* Embedded time stamp flag: not present */ put_bits(&c->pb, 1, 0); /* Auxiliary data flag: not present */ put_bits(&c->pb, 1, 0); /* HDCD source: no */ put_bits(&c->pb, 1, 0); /* Extension audio ID: N/A */ put_bits(&c->pb, 3, 0); /* Extended audio data: not present */ put_bits(&c->pb, 1, 0); /* Audio sync word insertion flag: after each sub-frame */ put_bits(&c->pb, 1, 0); /* Low frequency effects flag: not present or 64x subsampling */ put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0); /* Predictor history switch flag: on */ put_bits(&c->pb, 1, 1); /* No CRC */ /* Multirate interpolator switch: non-perfect reconstruction */ put_bits(&c->pb, 1, 0); /* Encoder software revision: 7 */ put_bits(&c->pb, 4, 7); /* Copy history: 0 */ put_bits(&c->pb, 2, 0); /* Source PCM resolution: 16 bits, not DTS ES */ put_bits(&c->pb, 3, 0); /* Front sum/difference coding: no */ put_bits(&c->pb, 1, 0); /* Surrounds sum/difference coding: no */ put_bits(&c->pb, 1, 0); /* Dialog normalization: 0 dB */ put_bits(&c->pb, 4, 0); } static void put_primary_audio_header(DCAContext *c) { static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; int ch, i; /* Number of subframes */ put_bits(&c->pb, 4, SUBFRAMES - 1); /* Number of primary audio channels */ put_bits(&c->pb, 3, c->fullband_channels - 1); /* Subband activity count */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 5, DCA_SUBBANDS - 2); /* High frequency VQ start subband */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 5, DCA_SUBBANDS - 1); /* Joint intensity coding index: 0, 0 */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 3, 0); /* Transient mode codebook: A4, A4 (arbitrary) */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 2, 0); /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 3, 6); /* Bit allocation quantizer select: linear 5-bit */ for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, 3, 6); /* Quantization index codebook select: dummy data to avoid transmission of scale factor adjustment */ for (i = 1; i < 11; i++) for (ch = 0; ch < c->fullband_channels; ch++) put_bits(&c->pb, bitlen[i], thr[i]); /* Scale factor adjustment index: not transmitted */ /* Audio header CRC check word: not transmitted */ } static void put_subframe_samples(DCAContext *c, int ss, int band, int ch) { if (c->abits[band][ch] <= 7) { int sum, i, j; for (i = 0; i < 8; i += 4) { sum = 0; for (j = 3; j >= 0; j--) { sum *= quant_levels[c->abits[band][ch]]; sum += c->quantized[ss * 8 + i + j][band][ch]; sum += (quant_levels[c->abits[band][ch]] - 1) / 2; } put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum); } } else { int i; for (i = 0; i < 8; i++) { int bits = bit_consumption[c->abits[band][ch]] / 16; int32_t mask = (1 << bits) - 1; put_bits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch] & mask); } } } static void put_subframe(DCAContext *c, int subframe) { int i, band, ss, ch; /* Subsubframes count */ put_bits(&c->pb, 2, SUBSUBFRAMES -1); /* Partial subsubframe sample count: dummy */ put_bits(&c->pb, 3, 0); /* Prediction mode: no ADPCM, in each channel and subband */ for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCA_SUBBANDS; band++) put_bits(&c->pb, 1, 0); /* Prediction VQ address: not transmitted */ /* Bit allocation index */ for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCA_SUBBANDS; band++) put_bits(&c->pb, 5, c->abits[band][ch]); if (SUBSUBFRAMES > 1) { /* Transition mode: none for each channel and subband */ for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCA_SUBBANDS; band++) put_bits(&c->pb, 1, 0); /* codebook A4 */ } /* Scale factors */ for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCA_SUBBANDS; band++) put_bits(&c->pb, 7, c->scale_factor[band][ch]); /* Joint subband scale factor codebook select: not transmitted */ /* Scale factors for joint subband coding: not transmitted */ /* Stereo down-mix coefficients: not transmitted */ /* Dynamic range coefficient: not transmitted */ /* Stde information CRC check word: not transmitted */ /* VQ encoded high frequency subbands: not transmitted */ /* LFE data: 8 samples and scalefactor */ if (c->lfe_channel) { for (i = 0; i < DCA_LFE_SAMPLES; i++) put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff); put_bits(&c->pb, 8, c->lfe_scale_factor); } /* Audio data (subsubframes) */ for (ss = 0; ss < SUBSUBFRAMES ; ss++) for (ch = 0; ch < c->fullband_channels; ch++) for (band = 0; band < DCA_SUBBANDS; band++) put_subframe_samples(c, ss, band, ch); /* DSYNC */ put_bits(&c->pb, 16, 0xffff); } static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { DCAContext *c = avctx->priv_data; const int32_t *samples; int ret, i; if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size )) < 0) return ret; samples = (const int32_t *)frame->data[0]; subband_transform(c, samples); if (c->lfe_channel) lfe_downsample(c, samples); calc_masking(c, samples); find_peaks(c); assign_bits(c); calc_scales(c); quantize_all(c); shift_history(c, samples); init_put_bits(&c->pb, avpkt->data, avpkt->size); put_frame_header(c); put_primary_audio_header(c); for (i = 0; i < SUBFRAMES; i++) put_subframe(c, i); 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(DCAContext), .init = encode_init, .encode2 = encode_frame, .capabilities = CODEC_CAP_EXPERIMENTAL, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_NONE }, .supported_samplerates = sample_rates, .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_2_2, AV_CH_LAYOUT_5POINT0, AV_CH_LAYOUT_5POINT1, 0 }, .defaults = defaults, };