diff options
Diffstat (limited to 'ffmpeg/libavcodec/dcaenc.c')
| -rw-r--r-- | ffmpeg/libavcodec/dcaenc.c | 974 |
1 files changed, 0 insertions, 974 deletions
diff --git a/ffmpeg/libavcodec/dcaenc.c b/ffmpeg/libavcodec/dcaenc.c deleted file mode 100644 index cb73f42..0000000 --- a/ffmpeg/libavcodec/dcaenc.c +++ /dev/null @@ -1,974 +0,0 @@ -/* - * 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 "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 addres: 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, -}; |