diff options
Diffstat (limited to 'libavcodec/dcaenc.c')
| -rw-r--r-- | libavcodec/dcaenc.c | 975 |
1 files changed, 975 insertions, 0 deletions
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c new file mode 100644 index 0000000000..905cdc9ade --- /dev/null +++ b/libavcodec/dcaenc.c @@ -0,0 +1,975 @@ +/* + * 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); + + 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, +}; |