| /* |
| * 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/internal.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, |
| }; |
