|
- /*
- * 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 "libavutil/ffmath.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 DCAENC_SUBBANDS 32
- #define SUBFRAMES 1
- #define SUBSUBFRAMES 2
- #define SUBBAND_SAMPLES (SUBFRAMES * SUBSUBFRAMES * 8)
- #define AUBANDS 25
-
- typedef struct DCAEncContext {
- 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;
- const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe
-
- int32_t history[512][MAX_CHANNELS]; /* This is a circular buffer */
- int32_t subband[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
- int32_t quantized[SUBBAND_SAMPLES][DCAENC_SUBBANDS][MAX_CHANNELS];
- int32_t peak_cb[DCAENC_SUBBANDS][MAX_CHANNELS];
- int32_t downsampled_lfe[DCA_LFE_SAMPLES];
- int32_t masking_curve_cb[SUBSUBFRAMES][256];
- int abits[DCAENC_SUBBANDS][MAX_CHANNELS];
- int scale_factor[DCAENC_SUBBANDS][MAX_CHANNELS];
- softfloat quant[DCAENC_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;
- } DCAEncContext;
-
- 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)
- {
- DCAEncContext *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--;
- c->channel_order_tab = ff_dca_channel_reorder_lfe[c->channel_config];
- } else {
- c->channel_order_tab = ff_dca_channel_reorder_nolfe[c->channel_config];
- }
-
- 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.", (int64_t)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 (!cos_table[0]) {
- int j, k;
-
- cos_table[0] = 0x7fffffff;
- cos_table[512] = 0;
- cos_table[1024] = -cos_table[0];
- for (i = 1; i < 512; i++) {
- cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024));
- cos_table[1024-i] = -cos_table[i];
- cos_table[1024+i] = -cos_table[i];
- cos_table[2048-i] = cos_table[i];
- }
- for (i = 0; i < 2048; i++) {
- cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i));
- }
-
- for (k = 0; k < 32; k++) {
- for (j = 0; j < 8; j++) {
- lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]);
- 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++) {
- band_interpolation[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]);
- band_interpolation[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;
-
- 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);
- 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);
- }
- 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 = ff_dca_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(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];
-
- 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 + 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 = ff_dca_lfe_index[c->channel_config];
- 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 + lfech];
-
- 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)(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[k][ch];
- for (k -= 512; i < 512; i++, k++)
- data[i] = input[k * c->channels + chi];
- adjust_jnd(c->samplerate_index, data, c->masking_curve_cb[ssf]);
- }
- for (i = 0; i < 256; i++) {
- int32_t m = 2048;
-
- for (ssf = 0; ssf < SUBSUBFRAMES; ssf++)
- if (c->masking_curve_cb[ssf][i] < m)
- m = c->masking_curve_cb[ssf][i];
- c->eff_masking_curve_cb[i] = m;
- }
-
- for (band = 0; band < 32; band++) {
- c->band_masking_cb[band] = 2048;
- walk_band_low(c, band, 0, update_band_masking, NULL);
- walk_band_high(c, band, 0, update_band_masking, NULL);
- }
- }
-
- static void find_peaks(DCAEncContext *c)
- {
- int band, ch;
-
- for (band = 0; band < 32; band++)
- for (ch = 0; ch < c->fullband_channels; ch++) {
- int sample;
- int32_t m = 0;
-
- for (sample = 0; sample < SUBBAND_SAMPLES; sample++) {
- int32_t s = abs(c->subband[sample][band][ch]);
- if (m < s)
- m = s;
- }
- c->peak_cb[band][ch] = get_cb(m);
- }
-
- if (c->lfe_channel) {
- int sample;
- int32_t m = 0;
-
- for (sample = 0; sample < DCA_LFE_SAMPLES; sample++)
- if (m < abs(c->downsampled_lfe[sample]))
- m = abs(c->downsampled_lfe[sample]);
- c->lfe_peak_cb = get_cb(m);
- }
- }
-
- static const int snr_fudge = 128;
- #define USED_1ABITS 1
- #define USED_NABITS 2
- #define USED_26ABITS 4
-
- static int init_quantization_noise(DCAEncContext *c, int noise)
- {
- int ch, band, ret = 0;
-
- c->consumed_bits = 132 + 493 * c->fullband_channels;
- if (c->lfe_channel)
- c->consumed_bits += 72;
-
- /* attempt to guess the bit distribution based on the prevoius frame */
- for (ch = 0; ch < c->fullband_channels; ch++) {
- for (band = 0; band < 32; band++) {
- int snr_cb = c->peak_cb[band][ch] - c->band_masking_cb[band] - noise;
-
- if (snr_cb >= 1312) {
- c->abits[band][ch] = 26;
- ret |= USED_26ABITS;
- } else if (snr_cb >= 222) {
- c->abits[band][ch] = 8 + mul32(snr_cb - 222, 69000000);
- ret |= USED_NABITS;
- } else if (snr_cb >= 0) {
- c->abits[band][ch] = 2 + mul32(snr_cb, 106000000);
- ret |= USED_NABITS;
- } else {
- c->abits[band][ch] = 1;
- ret |= USED_1ABITS;
- }
- }
- }
-
- for (band = 0; band < 32; band++)
- for (ch = 0; ch < c->fullband_channels; ch++) {
- c->consumed_bits += bit_consumption[c->abits[band][ch]];
- }
-
- return ret;
- }
-
- static void assign_bits(DCAEncContext *c)
- {
- /* Find the bounds where the binary search should work */
- int low, high, down;
- int used_abits = 0;
-
- init_quantization_noise(c, c->worst_quantization_noise);
- low = high = c->worst_quantization_noise;
- if (c->consumed_bits > c->frame_bits) {
- while (c->consumed_bits > c->frame_bits) {
- av_assert0(used_abits != USED_1ABITS);
- low = high;
- high += snr_fudge;
- used_abits = init_quantization_noise(c, high);
- }
- } else {
- while (c->consumed_bits <= c->frame_bits) {
- high = low;
- if (used_abits == USED_26ABITS)
- goto out; /* The requested bitrate is too high, pad with zeros */
- low -= snr_fudge;
- used_abits = init_quantization_noise(c, low);
- }
- }
-
- /* Now do a binary search between low and high to see what fits */
- for (down = snr_fudge >> 1; down; down >>= 1) {
- init_quantization_noise(c, high - down);
- if (c->consumed_bits <= c->frame_bits)
- high -= down;
- }
- init_quantization_noise(c, high);
- out:
- c->worst_quantization_noise = high;
- if (high > c->worst_noise_ever)
- c->worst_noise_ever = high;
- }
-
- static void shift_history(DCAEncContext *c, const int32_t *input)
- {
- int k, ch;
-
- for (k = 0; k < 512; k++)
- for (ch = 0; ch < c->channels; ch++) {
- const int chi = c->channel_order_tab[ch];
-
- c->history[k][ch] = input[k * c->channels + chi];
- }
- }
-
- static int32_t quantize_value(int32_t value, softfloat quant)
- {
- int32_t offset = 1 << (quant.e - 1);
-
- value = mul32(value, quant.m) + offset;
- value = value >> quant.e;
- return value;
- }
-
- static int calc_one_scale(int32_t peak_cb, int abits, softfloat *quant)
- {
- int32_t peak;
- int our_nscale, try_remove;
- softfloat our_quant;
-
- av_assert0(peak_cb <= 0);
- av_assert0(peak_cb >= -2047);
-
- our_nscale = 127;
- peak = cb_to_level[-peak_cb];
-
- for (try_remove = 64; try_remove > 0; try_remove >>= 1) {
- if (scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e <= 17)
- continue;
- our_quant.m = mul32(scalefactor_inv[our_nscale - try_remove].m, stepsize_inv[abits].m);
- our_quant.e = scalefactor_inv[our_nscale - try_remove].e + stepsize_inv[abits].e - 17;
- if ((quant_levels[abits] - 1) / 2 < quantize_value(peak, our_quant))
- continue;
- our_nscale -= try_remove;
- }
-
- if (our_nscale >= 125)
- our_nscale = 124;
-
- quant->m = mul32(scalefactor_inv[our_nscale].m, stepsize_inv[abits].m);
- quant->e = scalefactor_inv[our_nscale].e + stepsize_inv[abits].e - 17;
- av_assert0((quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant));
-
- return our_nscale;
- }
-
- static void calc_scales(DCAEncContext *c)
- {
- int band, ch;
-
- for (band = 0; band < 32; band++)
- for (ch = 0; ch < c->fullband_channels; ch++)
- c->scale_factor[band][ch] = calc_one_scale(c->peak_cb[band][ch],
- c->abits[band][ch],
- &c->quant[band][ch]);
-
- if (c->lfe_channel)
- c->lfe_scale_factor = calc_one_scale(c->lfe_peak_cb, 11, &c->lfe_quant);
- }
-
- static void quantize_all(DCAEncContext *c)
- {
- int sample, band, ch;
-
- for (sample = 0; sample < SUBBAND_SAMPLES; sample++)
- for (band = 0; band < 32; band++)
- for (ch = 0; ch < c->fullband_channels; ch++)
- c->quantized[sample][band][ch] = quantize_value(c->subband[sample][band][ch], c->quant[band][ch]);
- }
-
- static void put_frame_header(DCAEncContext *c)
- {
- /* SYNC */
- put_bits(&c->pb, 16, 0x7ffe);
- put_bits(&c->pb, 16, 0x8001);
-
- /* Frame type: normal */
- put_bits(&c->pb, 1, 1);
-
- /* Deficit sample count: none */
- put_bits(&c->pb, 5, 31);
-
- /* CRC is not present */
- put_bits(&c->pb, 1, 0);
-
- /* Number of PCM sample blocks */
- put_bits(&c->pb, 7, SUBBAND_SAMPLES - 1);
-
- /* Primary frame byte size */
- put_bits(&c->pb, 14, c->frame_size - 1);
-
- /* Audio channel arrangement */
- put_bits(&c->pb, 6, c->channel_config);
-
- /* Core audio sampling frequency */
- put_bits(&c->pb, 4, bitstream_sfreq[c->samplerate_index]);
-
- /* Transmission bit rate */
- put_bits(&c->pb, 5, c->bitrate_index);
-
- /* Embedded down mix: disabled */
- put_bits(&c->pb, 1, 0);
-
- /* Embedded dynamic range flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Embedded time stamp flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Auxiliary data flag: not present */
- put_bits(&c->pb, 1, 0);
-
- /* HDCD source: no */
- put_bits(&c->pb, 1, 0);
-
- /* Extension audio ID: N/A */
- put_bits(&c->pb, 3, 0);
-
- /* Extended audio data: not present */
- put_bits(&c->pb, 1, 0);
-
- /* Audio sync word insertion flag: after each sub-frame */
- put_bits(&c->pb, 1, 0);
-
- /* Low frequency effects flag: not present or 64x subsampling */
- put_bits(&c->pb, 2, c->lfe_channel ? 2 : 0);
-
- /* Predictor history switch flag: on */
- put_bits(&c->pb, 1, 1);
-
- /* No CRC */
- /* Multirate interpolator switch: non-perfect reconstruction */
- put_bits(&c->pb, 1, 0);
-
- /* Encoder software revision: 7 */
- put_bits(&c->pb, 4, 7);
-
- /* Copy history: 0 */
- put_bits(&c->pb, 2, 0);
-
- /* Source PCM resolution: 16 bits, not DTS ES */
- put_bits(&c->pb, 3, 0);
-
- /* Front sum/difference coding: no */
- put_bits(&c->pb, 1, 0);
-
- /* Surrounds sum/difference coding: no */
- put_bits(&c->pb, 1, 0);
-
- /* Dialog normalization: 0 dB */
- put_bits(&c->pb, 4, 0);
- }
-
- static void put_primary_audio_header(DCAEncContext *c)
- {
- static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
- static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
-
- int ch, i;
- /* Number of subframes */
- put_bits(&c->pb, 4, SUBFRAMES - 1);
-
- /* Number of primary audio channels */
- put_bits(&c->pb, 3, c->fullband_channels - 1);
-
- /* Subband activity count */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 5, DCAENC_SUBBANDS - 2);
-
- /* High frequency VQ start subband */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 5, DCAENC_SUBBANDS - 1);
-
- /* Joint intensity coding index: 0, 0 */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 3, 0);
-
- /* Transient mode codebook: A4, A4 (arbitrary) */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 2, 0);
-
- /* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each channel) */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 3, 6);
-
- /* Bit allocation quantizer select: linear 5-bit */
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, 3, 6);
-
- /* Quantization index codebook select: dummy data
- to avoid transmission of scale factor adjustment */
- for (i = 1; i < 11; i++)
- for (ch = 0; ch < c->fullband_channels; ch++)
- put_bits(&c->pb, bitlen[i], thr[i]);
-
- /* Scale factor adjustment index: not transmitted */
- /* Audio header CRC check word: not transmitted */
- }
-
- static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch)
- {
- if (c->abits[band][ch] <= 7) {
- int sum, i, j;
- for (i = 0; i < 8; i += 4) {
- sum = 0;
- for (j = 3; j >= 0; j--) {
- sum *= quant_levels[c->abits[band][ch]];
- sum += c->quantized[ss * 8 + i + j][band][ch];
- sum += (quant_levels[c->abits[band][ch]] - 1) / 2;
- }
- put_bits(&c->pb, bit_consumption[c->abits[band][ch]] / 4, sum);
- }
- } else {
- int i;
- for (i = 0; i < 8; i++) {
- int bits = bit_consumption[c->abits[band][ch]] / 16;
- put_sbits(&c->pb, bits, c->quantized[ss * 8 + i][band][ch]);
- }
- }
- }
-
- static void put_subframe(DCAEncContext *c, int subframe)
- {
- int i, band, ss, ch;
-
- /* Subsubframes count */
- put_bits(&c->pb, 2, SUBSUBFRAMES -1);
-
- /* Partial subsubframe sample count: dummy */
- put_bits(&c->pb, 3, 0);
-
- /* Prediction mode: no ADPCM, in each channel and subband */
- for (ch = 0; ch < c->fullband_channels; ch++)
- for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_bits(&c->pb, 1, 0);
-
- /* Prediction VQ address: not transmitted */
- /* Bit allocation index */
- for (ch = 0; ch < c->fullband_channels; ch++)
- for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_bits(&c->pb, 5, c->abits[band][ch]);
-
- if (SUBSUBFRAMES > 1) {
- /* Transition mode: none for each channel and subband */
- for (ch = 0; ch < c->fullband_channels; ch++)
- for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_bits(&c->pb, 1, 0); /* codebook A4 */
- }
-
- /* Scale factors */
- for (ch = 0; ch < c->fullband_channels; ch++)
- for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_bits(&c->pb, 7, c->scale_factor[band][ch]);
-
- /* Joint subband scale factor codebook select: not transmitted */
- /* Scale factors for joint subband coding: not transmitted */
- /* Stereo down-mix coefficients: not transmitted */
- /* Dynamic range coefficient: not transmitted */
- /* Stde information CRC check word: not transmitted */
- /* VQ encoded high frequency subbands: not transmitted */
-
- /* LFE data: 8 samples and scalefactor */
- if (c->lfe_channel) {
- for (i = 0; i < DCA_LFE_SAMPLES; i++)
- put_bits(&c->pb, 8, quantize_value(c->downsampled_lfe[i], c->lfe_quant) & 0xff);
- put_bits(&c->pb, 8, c->lfe_scale_factor);
- }
-
- /* Audio data (subsubframes) */
- for (ss = 0; ss < SUBSUBFRAMES ; ss++)
- for (ch = 0; ch < c->fullband_channels; ch++)
- for (band = 0; band < DCAENC_SUBBANDS; band++)
- put_subframe_samples(c, ss, band, ch);
-
- /* DSYNC */
- put_bits(&c->pb, 16, 0xffff);
- }
-
- static int encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
- const AVFrame *frame, int *got_packet_ptr)
- {
- DCAEncContext *c = avctx->priv_data;
- const int32_t *samples;
- int ret, i;
-
- if ((ret = ff_alloc_packet2(avctx, avpkt, c->frame_size, 0)) < 0)
- return ret;
-
- samples = (const int32_t *)frame->data[0];
-
- subband_transform(c, samples);
- if (c->lfe_channel)
- lfe_downsample(c, samples);
-
- calc_masking(c, samples);
- find_peaks(c);
- assign_bits(c);
- calc_scales(c);
- quantize_all(c);
- shift_history(c, samples);
-
- init_put_bits(&c->pb, avpkt->data, avpkt->size);
- put_frame_header(c);
- put_primary_audio_header(c);
- for (i = 0; i < SUBFRAMES; i++)
- put_subframe(c, i);
-
-
- for (i = put_bits_count(&c->pb); i < 8*c->frame_size; i++)
- put_bits(&c->pb, 1, 0);
-
- flush_put_bits(&c->pb);
-
- avpkt->pts = frame->pts;
- avpkt->duration = ff_samples_to_time_base(avctx, frame->nb_samples);
- avpkt->size = put_bits_count(&c->pb) >> 3;
- *got_packet_ptr = 1;
- return 0;
- }
-
- static const AVCodecDefault defaults[] = {
- { "b", "1411200" },
- { NULL },
- };
-
- AVCodec ff_dca_encoder = {
- .name = "dca",
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .type = AVMEDIA_TYPE_AUDIO,
- .id = AV_CODEC_ID_DTS,
- .priv_data_size = sizeof(DCAEncContext),
- .init = encode_init,
- .encode2 = encode_frame,
- .capabilities = AV_CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32,
- AV_SAMPLE_FMT_NONE },
- .supported_samplerates = sample_rates,
- .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO,
- AV_CH_LAYOUT_STEREO,
- AV_CH_LAYOUT_2_2,
- AV_CH_LAYOUT_5POINT0,
- AV_CH_LAYOUT_5POINT1,
- 0 },
- .defaults = defaults,
- };
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