| /* |
| * 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 |
| */ |
| |
| #define FFT_FLOAT 0 |
| #define FFT_FIXED_32 1 |
| |
| #include "libavutil/avassert.h" |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/common.h" |
| #include "libavutil/ffmath.h" |
| #include "libavutil/opt.h" |
| #include "avcodec.h" |
| #include "dca.h" |
| #include "dcaadpcm.h" |
| #include "dcamath.h" |
| #include "dca_core.h" |
| #include "dcadata.h" |
| #include "dcaenc.h" |
| #include "fft.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 |
| |
| #define COS_T(x) (c->cos_table[(x) & 2047]) |
| |
| typedef struct CompressionOptions { |
| int adpcm_mode; |
| } CompressionOptions; |
| |
| typedef struct DCAEncContext { |
| AVClass *class; |
| PutBitContext pb; |
| DCAADPCMEncContext adpcm_ctx; |
| FFTContext mdct; |
| CompressionOptions options; |
| 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 prediction_mode[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| int32_t adpcm_history[MAX_CHANNELS][DCAENC_SUBBANDS][DCA_ADPCM_COEFFS * 2]; |
| int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */ |
| int32_t *subband[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| int32_t quantized[MAX_CHANNELS][DCAENC_SUBBANDS][SUBBAND_SAMPLES]; |
| int32_t peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| int32_t diff_peak_cb[MAX_CHANNELS][DCAENC_SUBBANDS]; ///< expected peak of residual signal |
| int32_t downsampled_lfe[DCA_LFE_SAMPLES]; |
| int32_t masking_curve_cb[SUBSUBFRAMES][256]; |
| int32_t bit_allocation_sel[MAX_CHANNELS]; |
| int abits[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| int scale_factor[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| softfloat quant[MAX_CHANNELS][DCAENC_SUBBANDS]; |
| int32_t quant_index_sel[MAX_CHANNELS][DCA_CODE_BOOKS]; |
| 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; |
| int consumed_adpcm_bits; ///< Number of bits to transmit ADPCM related info |
| |
| int32_t cos_table[2048]; |
| int32_t band_interpolation_tab[2][512]; |
| int32_t band_spectrum_tab[2][8]; |
| int32_t auf[9][AUBANDS][256]; |
| int32_t cb_to_add[256]; |
| int32_t cb_to_level[2048]; |
| int32_t lfe_fir_64i[512]; |
| } DCAEncContext; |
| |
| /* 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 subband_bufer_alloc(DCAEncContext *c) |
| { |
| int ch, band; |
| int32_t *bufer = av_calloc(MAX_CHANNELS * DCAENC_SUBBANDS * |
| (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS), |
| sizeof(int32_t)); |
| if (!bufer) |
| return AVERROR(ENOMEM); |
| |
| /* we need a place for DCA_ADPCM_COEFF samples from previous frame |
| * to calc prediction coefficients for each subband */ |
| for (ch = 0; ch < MAX_CHANNELS; ch++) { |
| for (band = 0; band < DCAENC_SUBBANDS; band++) { |
| c->subband[ch][band] = bufer + |
| ch * DCAENC_SUBBANDS * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + |
| band * (SUBBAND_SAMPLES + DCA_ADPCM_COEFFS) + DCA_ADPCM_COEFFS; |
| } |
| } |
| return 0; |
| } |
| |
| static void subband_bufer_free(DCAEncContext *c) |
| { |
| if (c->subband[0][0]) { |
| int32_t *bufer = c->subband[0][0] - DCA_ADPCM_COEFFS; |
| av_free(bufer); |
| c->subband[0][0] = NULL; |
| } |
| } |
| |
| static int encode_init(AVCodecContext *avctx) |
| { |
| DCAEncContext *c = avctx->priv_data; |
| uint64_t layout = avctx->channel_layout; |
| int i, j, k, min_frame_bits; |
| int ret; |
| |
| if ((ret = subband_bufer_alloc(c)) < 0) |
| return ret; |
| |
| c->fullband_channels = c->channels = avctx->channels; |
| c->lfe_channel = (avctx->channels == 3 || avctx->channels == 6); |
| c->band_interpolation = c->band_interpolation_tab[1]; |
| c->band_spectrum = c->band_spectrum_tab[1]; |
| c->worst_quantization_noise = -2047; |
| c->worst_noise_ever = -2047; |
| c->consumed_adpcm_bits = 0; |
| |
| if (ff_dcaadpcm_init(&c->adpcm_ctx)) |
| return AVERROR(ENOMEM); |
| |
| 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 = channel_reorder_lfe[c->channel_config]; |
| } else { |
| c->channel_order_tab = channel_reorder_nolfe[c->channel_config]; |
| } |
| |
| for (i = 0; i < MAX_CHANNELS; i++) { |
| for (j = 0; j < DCA_CODE_BOOKS; j++) { |
| c->quant_index_sel[i][j] = ff_dca_quant_index_group_size[j]; |
| } |
| /* 6 - no Huffman */ |
| c->bit_allocation_sel[i] = 6; |
| |
| for (j = 0; j < DCAENC_SUBBANDS; j++) { |
| /* -1 - no ADPCM */ |
| c->prediction_mode[i][j] = -1; |
| memset(c->adpcm_history[i][j], 0, sizeof(int32_t)*DCA_ADPCM_COEFFS); |
| } |
| } |
| |
| 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.", 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 ((ret = ff_mdct_init(&c->mdct, 9, 0, 1.0)) < 0) |
| return ret; |
| |
| /* Init all tables */ |
| c->cos_table[0] = 0x7fffffff; |
| c->cos_table[512] = 0; |
| c->cos_table[1024] = -c->cos_table[0]; |
| for (i = 1; i < 512; i++) { |
| c->cos_table[i] = (int32_t)(0x7fffffff * cos(M_PI * i / 1024)); |
| c->cos_table[1024-i] = -c->cos_table[i]; |
| c->cos_table[1024+i] = -c->cos_table[i]; |
| c->cos_table[2048-i] = +c->cos_table[i]; |
| } |
| |
| for (i = 0; i < 2048; i++) |
| c->cb_to_level[i] = (int32_t)(0x7fffffff * ff_exp10(-0.005 * i)); |
| |
| for (k = 0; k < 32; k++) { |
| for (j = 0; j < 8; j++) { |
| c->lfe_fir_64i[64 * j + k] = (int32_t)(0xffffff800000ULL * ff_dca_lfe_fir_64[8 * k + j]); |
| c->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++) { |
| c->band_interpolation_tab[0][i] = (int32_t)(0x1000000000ULL * ff_dca_fir_32bands_perfect[i]); |
| c->band_interpolation_tab[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; |
| |
| c->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); |
| c->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); |
| } |
| c->band_spectrum_tab[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); |
| } |
| c->band_spectrum_tab[1][j] = (int32_t)(200 * log10(accum)); |
| } |
| |
| return 0; |
| } |
| |
| static av_cold int encode_close(AVCodecContext *avctx) |
| { |
| DCAEncContext *c = avctx->priv_data; |
| ff_mdct_end(&c->mdct); |
| subband_bufer_free(c); |
| ff_dcaadpcm_free(&c->adpcm_ctx); |
| |
| return 0; |
| } |
| |
| 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]; |
| |
| memcpy(hist, &c->history[ch][0], 512 * sizeof(int32_t)); |
| |
| for (subs = 0; subs < SUBBAND_SAMPLES; subs++) { |
| int32_t accum[64]; |
| int32_t resp; |
| int band; |
| |
| /* Calculate the convolutions at once */ |
| memset(accum, 0, 64 * sizeof(int32_t)); |
| |
| 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[ch][band][subs] = ((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 = lfe_index[c->channel_config]; |
| int i, j, lfes; |
| int32_t hist[512]; |
| int32_t accum; |
| int hist_start = 0; |
| |
| memcpy(hist, &c->history[c->channels - 1][0], 512 * sizeof(int32_t)); |
| |
| 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], c->lfe_fir_64i[j]); |
| for (i = 0; i < hist_start; i++, j++) |
| accum += mul32(hist[i], c->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; |
| } |
| } |
| |
| static int32_t get_cb(DCAEncContext *c, int32_t in) |
| { |
| int i, res = 0; |
| in = FFABS(in); |
| |
| for (i = 1024; i > 0; i >>= 1) { |
| if (c->cb_to_level[i + res] >= in) |
| res += i; |
| } |
| return -res; |
| } |
| |
| static int32_t add_cb(DCAEncContext *c, int32_t a, int32_t b) |
| { |
| if (a < b) |
| FFSWAP(int32_t, a, b); |
| |
| if (a - b >= 256) |
| return a; |
| return a + c->cb_to_add[a - b]; |
| } |
| |
| static void calc_power(DCAEncContext *c, |
| const int32_t in[2 * 256], int32_t power[256]) |
| { |
| int i; |
| LOCAL_ALIGNED_32(int32_t, data, [512]); |
| LOCAL_ALIGNED_32(int32_t, coeff, [256]); |
| |
| for (i = 0; i < 512; i++) |
| data[i] = norm__(mul32(in[i], 0x3fffffff - (COS_T(4 * i + 2) >> 1)), 4); |
| |
| c->mdct.mdct_calc(&c->mdct, coeff, data); |
| for (i = 0; i < 256; i++) { |
| const int32_t cb = get_cb(c, coeff[i]); |
| power[i] = add_cb(c, cb, cb); |
| } |
| } |
| |
| static void adjust_jnd(DCAEncContext *c, |
| const int32_t in[512], int32_t out_cb[256]) |
| { |
| int32_t power[256]; |
| int32_t out_cb_unnorm[256]; |
| int32_t denom; |
| const int32_t ca_cb = -1114; |
| const int32_t cs_cb = 928; |
| const int samplerate_index = c->samplerate_index; |
| int i, j; |
| |
| calc_power(c, in, power); |
| |
| for (j = 0; j < 256; j++) |
| 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(c, denom, power[j] + c->auf[samplerate_index][i][j]); |
| for (j = 0; j < 256; j++) |
| out_cb_unnorm[j] = add_cb(c, out_cb_unnorm[j], |
| -denom + c->auf[samplerate_index][i][j]); |
| } |
| |
| for (j = 0; j < 256; j++) |
| out_cb[j] = add_cb(c, 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[ch][k]; |
| for (k -= 512; i < 512; i++, k++) |
| data[i] = input[k * c->channels + chi]; |
| adjust_jnd(c, 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 inline int32_t find_peak(DCAEncContext *c, const int32_t *in, int len) |
| { |
| int sample; |
| int32_t m = 0; |
| for (sample = 0; sample < len; sample++) { |
| int32_t s = abs(in[sample]); |
| if (m < s) |
| m = s; |
| } |
| return get_cb(c, m); |
| } |
| |
| static void find_peaks(DCAEncContext *c) |
| { |
| int band, ch; |
| |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| for (band = 0; band < 32; band++) |
| c->peak_cb[ch][band] = find_peak(c, c->subband[ch][band], |
| SUBBAND_SAMPLES); |
| } |
| |
| if (c->lfe_channel) |
| c->lfe_peak_cb = find_peak(c, c->downsampled_lfe, DCA_LFE_SAMPLES); |
| } |
| |
| static void adpcm_analysis(DCAEncContext *c) |
| { |
| int ch, band; |
| int pred_vq_id; |
| int32_t *samples; |
| int32_t estimated_diff[SUBBAND_SAMPLES]; |
| |
| c->consumed_adpcm_bits = 0; |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| for (band = 0; band < 32; band++) { |
| samples = c->subband[ch][band] - DCA_ADPCM_COEFFS; |
| pred_vq_id = ff_dcaadpcm_subband_analysis(&c->adpcm_ctx, samples, |
| SUBBAND_SAMPLES, estimated_diff); |
| if (pred_vq_id >= 0) { |
| c->prediction_mode[ch][band] = pred_vq_id; |
| c->consumed_adpcm_bits += 12; //12 bits to transmit prediction vq index |
| c->diff_peak_cb[ch][band] = find_peak(c, estimated_diff, 16); |
| } else { |
| c->prediction_mode[ch][band] = -1; |
| } |
| } |
| } |
| } |
| |
| static const int snr_fudge = 128; |
| #define USED_1ABITS 1 |
| #define USED_26ABITS 4 |
| |
| static inline int32_t get_step_size(DCAEncContext *c, int ch, int band) |
| { |
| int32_t step_size; |
| |
| if (c->bitrate_index == 3) |
| step_size = ff_dca_lossless_quant[c->abits[ch][band]]; |
| else |
| step_size = ff_dca_lossy_quant[c->abits[ch][band]]; |
| |
| return step_size; |
| } |
| |
| static int calc_one_scale(DCAEncContext *c, 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 = c->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 ((ff_dca_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((ff_dca_quant_levels[abits] - 1) / 2 >= quantize_value(peak, *quant)); |
| |
| return our_nscale; |
| } |
| |
| static inline void quantize_adpcm_subband(DCAEncContext *c, int ch, int band) |
| { |
| int32_t step_size; |
| int32_t diff_peak_cb = c->diff_peak_cb[ch][band]; |
| c->scale_factor[ch][band] = calc_one_scale(c, diff_peak_cb, |
| c->abits[ch][band], |
| &c->quant[ch][band]); |
| |
| step_size = get_step_size(c, ch, band); |
| ff_dcaadpcm_do_real(c->prediction_mode[ch][band], |
| c->quant[ch][band], |
| ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], |
| step_size, c->adpcm_history[ch][band], c->subband[ch][band], |
| c->adpcm_history[ch][band] + 4, c->quantized[ch][band], |
| SUBBAND_SAMPLES, c->cb_to_level[-diff_peak_cb]); |
| } |
| |
| static void quantize_adpcm(DCAEncContext *c) |
| { |
| int band, ch; |
| |
| for (ch = 0; ch < c->fullband_channels; ch++) |
| for (band = 0; band < 32; band++) |
| if (c->prediction_mode[ch][band] >= 0) |
| quantize_adpcm_subband(c, ch, band); |
| } |
| |
| static void quantize_pcm(DCAEncContext *c) |
| { |
| int sample, band, ch; |
| |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| for (band = 0; band < 32; band++) { |
| if (c->prediction_mode[ch][band] == -1) { |
| for (sample = 0; sample < SUBBAND_SAMPLES; sample++) { |
| int32_t val = quantize_value(c->subband[ch][band][sample], |
| c->quant[ch][band]); |
| c->quantized[ch][band][sample] = val; |
| } |
| } |
| } |
| } |
| } |
| |
| static void accumulate_huff_bit_consumption(int abits, int32_t *quantized, |
| uint32_t *result) |
| { |
| uint8_t sel, id = abits - 1; |
| for (sel = 0; sel < ff_dca_quant_index_group_size[id]; sel++) |
| result[sel] += ff_dca_vlc_calc_quant_bits(quantized, SUBBAND_SAMPLES, |
| sel, id); |
| } |
| |
| static uint32_t set_best_code(uint32_t vlc_bits[DCA_CODE_BOOKS][7], |
| uint32_t clc_bits[DCA_CODE_BOOKS], |
| int32_t res[DCA_CODE_BOOKS]) |
| { |
| uint8_t i, sel; |
| uint32_t best_sel_bits[DCA_CODE_BOOKS]; |
| int32_t best_sel_id[DCA_CODE_BOOKS]; |
| uint32_t t, bits = 0; |
| |
| for (i = 0; i < DCA_CODE_BOOKS; i++) { |
| |
| av_assert0(!((!!vlc_bits[i][0]) ^ (!!clc_bits[i]))); |
| if (vlc_bits[i][0] == 0) { |
| /* do not transmit adjustment index for empty codebooks */ |
| res[i] = ff_dca_quant_index_group_size[i]; |
| /* and skip it */ |
| continue; |
| } |
| |
| best_sel_bits[i] = vlc_bits[i][0]; |
| best_sel_id[i] = 0; |
| for (sel = 0; sel < ff_dca_quant_index_group_size[i]; sel++) { |
| if (best_sel_bits[i] > vlc_bits[i][sel] && vlc_bits[i][sel]) { |
| best_sel_bits[i] = vlc_bits[i][sel]; |
| best_sel_id[i] = sel; |
| } |
| } |
| |
| /* 2 bits to transmit scale factor adjustment index */ |
| t = best_sel_bits[i] + 2; |
| if (t < clc_bits[i]) { |
| res[i] = best_sel_id[i]; |
| bits += t; |
| } else { |
| res[i] = ff_dca_quant_index_group_size[i]; |
| bits += clc_bits[i]; |
| } |
| } |
| return bits; |
| } |
| |
| static uint32_t set_best_abits_code(int abits[DCAENC_SUBBANDS], int bands, |
| int32_t *res) |
| { |
| uint8_t i; |
| uint32_t t; |
| int32_t best_sel = 6; |
| int32_t best_bits = bands * 5; |
| |
| /* Check do we have subband which cannot be encoded by Huffman tables */ |
| for (i = 0; i < bands; i++) { |
| if (abits[i] > 12 || abits[i] == 0) { |
| *res = best_sel; |
| return best_bits; |
| } |
| } |
| |
| for (i = 0; i < DCA_BITALLOC_12_COUNT; i++) { |
| t = ff_dca_vlc_calc_alloc_bits(abits, bands, i); |
| if (t < best_bits) { |
| best_bits = t; |
| best_sel = i; |
| } |
| } |
| |
| *res = best_sel; |
| return best_bits; |
| } |
| |
| static int init_quantization_noise(DCAEncContext *c, int noise, int forbid_zero) |
| { |
| int ch, band, ret = USED_26ABITS | USED_1ABITS; |
| uint32_t huff_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS][7]; |
| uint32_t clc_bit_count_accum[MAX_CHANNELS][DCA_CODE_BOOKS]; |
| uint32_t bits_counter = 0; |
| |
| c->consumed_bits = 132 + 333 * c->fullband_channels; |
| c->consumed_bits += c->consumed_adpcm_bits; |
| 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[ch][band] - c->band_masking_cb[band] - noise; |
| |
| if (snr_cb >= 1312) { |
| c->abits[ch][band] = 26; |
| ret &= ~USED_1ABITS; |
| } else if (snr_cb >= 222) { |
| c->abits[ch][band] = 8 + mul32(snr_cb - 222, 69000000); |
| ret &= ~(USED_26ABITS | USED_1ABITS); |
| } else if (snr_cb >= 0) { |
| c->abits[ch][band] = 2 + mul32(snr_cb, 106000000); |
| ret &= ~(USED_26ABITS | USED_1ABITS); |
| } else if (forbid_zero || snr_cb >= -140) { |
| c->abits[ch][band] = 1; |
| ret &= ~USED_26ABITS; |
| } else { |
| c->abits[ch][band] = 0; |
| ret &= ~(USED_26ABITS | USED_1ABITS); |
| } |
| } |
| c->consumed_bits += set_best_abits_code(c->abits[ch], 32, |
| &c->bit_allocation_sel[ch]); |
| } |
| |
| /* Recalc scale_factor each time to get bits consumption in case of Huffman coding. |
| It is suboptimal solution */ |
| /* TODO: May be cache scaled values */ |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| for (band = 0; band < 32; band++) { |
| if (c->prediction_mode[ch][band] == -1) { |
| c->scale_factor[ch][band] = calc_one_scale(c, c->peak_cb[ch][band], |
| c->abits[ch][band], |
| &c->quant[ch][band]); |
| } |
| } |
| } |
| quantize_adpcm(c); |
| quantize_pcm(c); |
| |
| memset(huff_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * 7 * sizeof(uint32_t)); |
| memset(clc_bit_count_accum, 0, MAX_CHANNELS * DCA_CODE_BOOKS * sizeof(uint32_t)); |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| for (band = 0; band < 32; band++) { |
| if (c->abits[ch][band] && c->abits[ch][band] <= DCA_CODE_BOOKS) { |
| accumulate_huff_bit_consumption(c->abits[ch][band], |
| c->quantized[ch][band], |
| huff_bit_count_accum[ch][c->abits[ch][band] - 1]); |
| clc_bit_count_accum[ch][c->abits[ch][band] - 1] += bit_consumption[c->abits[ch][band]]; |
| } else { |
| bits_counter += bit_consumption[c->abits[ch][band]]; |
| } |
| } |
| } |
| |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| bits_counter += set_best_code(huff_bit_count_accum[ch], |
| clc_bit_count_accum[ch], |
| c->quant_index_sel[ch]); |
| } |
| |
| c->consumed_bits += bits_counter; |
| |
| 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; |
| int forbid_zero = 1; |
| restart: |
| init_quantization_noise(c, c->worst_quantization_noise, forbid_zero); |
| low = high = c->worst_quantization_noise; |
| if (c->consumed_bits > c->frame_bits) { |
| while (c->consumed_bits > c->frame_bits) { |
| if (used_abits == USED_1ABITS && forbid_zero) { |
| forbid_zero = 0; |
| goto restart; |
| } |
| low = high; |
| high += snr_fudge; |
| used_abits = init_quantization_noise(c, high, forbid_zero); |
| } |
| } 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, forbid_zero); |
| } |
| } |
| |
| /* 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, forbid_zero); |
| if (c->consumed_bits <= c->frame_bits) |
| high -= down; |
| } |
| init_quantization_noise(c, high, forbid_zero); |
| 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[ch][k] = input[k * c->channels + chi]; |
| } |
| } |
| |
| static void fill_in_adpcm_bufer(DCAEncContext *c) |
| { |
| int ch, band; |
| int32_t step_size; |
| /* We fill in ADPCM work buffer for subbands which hasn't been ADPCM coded |
| * in current frame - we need this data if subband of next frame is |
| * ADPCM |
| */ |
| for (ch = 0; ch < c->channels; ch++) { |
| for (band = 0; band < 32; band++) { |
| int32_t *samples = c->subband[ch][band] - DCA_ADPCM_COEFFS; |
| if (c->prediction_mode[ch][band] == -1) { |
| step_size = get_step_size(c, ch, band); |
| |
| ff_dca_core_dequantize(c->adpcm_history[ch][band], |
| c->quantized[ch][band]+12, step_size, |
| ff_dca_scale_factor_quant7[c->scale_factor[ch][band]], 0, 4); |
| } else { |
| AV_COPY128U(c->adpcm_history[ch][band], c->adpcm_history[ch][band]+4); |
| } |
| /* Copy dequantized values for LPC analysis. |
| * It reduces artifacts in case of extreme quantization, |
| * example: in current frame abits is 1 and has no prediction flag, |
| * but end of this frame is sine like signal. In this case, if LPC analysis uses |
| * original values, likely LPC analysis returns good prediction gain, and sets prediction flag. |
| * But there are no proper value in decoder history, so likely result will be no good. |
| * Bitstream has "Predictor history flag switch", but this flag disables history for all subbands |
| */ |
| samples[0] = c->adpcm_history[ch][band][0] << 7; |
| samples[1] = c->adpcm_history[ch][band][1] << 7; |
| samples[2] = c->adpcm_history[ch][band][2] << 7; |
| samples[3] = c->adpcm_history[ch][band][3] << 7; |
| } |
| } |
| } |
| |
| static void calc_lfe_scales(DCAEncContext *c) |
| { |
| if (c->lfe_channel) |
| c->lfe_scale_factor = calc_one_scale(c, c->lfe_peak_cb, 11, &c->lfe_quant); |
| } |
| |
| 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) |
| { |
| 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, c->bit_allocation_sel[ch]); |
| |
| /* Quantization index codebook select */ |
| for (i = 0; i < DCA_CODE_BOOKS; i++) |
| for (ch = 0; ch < c->fullband_channels; ch++) |
| put_bits(&c->pb, ff_dca_quant_index_sel_nbits[i], c->quant_index_sel[ch][i]); |
| |
| /* Scale factor adjustment index: transmitted in case of Huffman coding */ |
| for (i = 0; i < DCA_CODE_BOOKS; i++) |
| for (ch = 0; ch < c->fullband_channels; ch++) |
| if (c->quant_index_sel[ch][i] < ff_dca_quant_index_group_size[i]) |
| put_bits(&c->pb, 2, 0); |
| |
| /* Audio header CRC check word: not transmitted */ |
| } |
| |
| static void put_subframe_samples(DCAEncContext *c, int ss, int band, int ch) |
| { |
| int i, j, sum, bits, sel; |
| if (c->abits[ch][band] <= DCA_CODE_BOOKS) { |
| av_assert0(c->abits[ch][band] > 0); |
| sel = c->quant_index_sel[ch][c->abits[ch][band] - 1]; |
| // Huffman codes |
| if (sel < ff_dca_quant_index_group_size[c->abits[ch][band] - 1]) { |
| ff_dca_vlc_enc_quant(&c->pb, &c->quantized[ch][band][ss * 8], 8, |
| sel, c->abits[ch][band] - 1); |
| return; |
| } |
| |
| // Block codes |
| if (c->abits[ch][band] <= 7) { |
| for (i = 0; i < 8; i += 4) { |
| sum = 0; |
| for (j = 3; j >= 0; j--) { |
| sum *= ff_dca_quant_levels[c->abits[ch][band]]; |
| sum += c->quantized[ch][band][ss * 8 + i + j]; |
| sum += (ff_dca_quant_levels[c->abits[ch][band]] - 1) / 2; |
| } |
| put_bits(&c->pb, bit_consumption[c->abits[ch][band]] / 4, sum); |
| } |
| return; |
| } |
| } |
| |
| for (i = 0; i < 8; i++) { |
| bits = bit_consumption[c->abits[ch][band]] / 16; |
| put_sbits(&c->pb, bits, c->quantized[ch][band][ss * 8 + i]); |
| } |
| } |
| |
| 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, !(c->prediction_mode[ch][band] == -1)); |
| |
| /* Prediction VQ address */ |
| for (ch = 0; ch < c->fullband_channels; ch++) |
| for (band = 0; band < DCAENC_SUBBANDS; band++) |
| if (c->prediction_mode[ch][band] >= 0) |
| put_bits(&c->pb, 12, c->prediction_mode[ch][band]); |
| |
| /* Bit allocation index */ |
| for (ch = 0; ch < c->fullband_channels; ch++) { |
| if (c->bit_allocation_sel[ch] == 6) { |
| for (band = 0; band < DCAENC_SUBBANDS; band++) { |
| put_bits(&c->pb, 5, c->abits[ch][band]); |
| } |
| } else { |
| ff_dca_vlc_enc_alloc(&c->pb, c->abits[ch], DCAENC_SUBBANDS, |
| c->bit_allocation_sel[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++) |
| if (c->abits[ch][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++) |
| if (c->abits[ch][band]) |
| put_bits(&c->pb, 7, c->scale_factor[ch][band]); |
| |
| /* 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++) |
| if (c->abits[ch][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); |
| if (c->options.adpcm_mode) |
| adpcm_analysis(c); |
| find_peaks(c); |
| assign_bits(c); |
| calc_lfe_scales(c); |
| shift_history(c, samples); |
| |
| init_put_bits(&c->pb, avpkt->data, avpkt->size); |
| fill_in_adpcm_bufer(c); |
| 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; |
| } |
| |
| #define DCAENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM |
| |
| static const AVOption options[] = { |
| { "dca_adpcm", "Use ADPCM encoding", offsetof(DCAEncContext, options.adpcm_mode), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, DCAENC_FLAGS }, |
| { NULL }, |
| }; |
| |
| static const AVClass dcaenc_class = { |
| .class_name = "DCA (DTS Coherent Acoustics)", |
| .item_name = av_default_item_name, |
| .option = options, |
| .version = LIBAVUTIL_VERSION_INT, |
| }; |
| |
| 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, |
| .close = encode_close, |
| .encode2 = encode_frame, |
| .capabilities = AV_CODEC_CAP_EXPERIMENTAL, |
| .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, |
| .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, |
| .priv_class = &dcaenc_class, |
| }; |
