DCA/DTA encoder

1 files changed, 587 insertions, 0 deletions

diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
new file mode 100644
index 0000000000..2b61bec98c
--- /dev/null
+++ b/libavcodec/dcaenc.c
@@ -0,0 +1,587 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 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/common.h"
+#include "libavutil/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+static const int8_t dca_lfe_index[] = {
+    1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+    { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1,  2, -1, -1, -1, -1, -1 },
+    { 1,  2,  0, -1,  3, -1, -1, -1, -1 },
+    { 0,  1, -1,  2,  3, -1, -1, -1, -1 },
+    { 1,  2,  0, -1,  3,  4, -1, -1, -1 },
+    { 2,  3, -1,  0,  1,  4,  5, -1, -1 },
+    { 1,  2,  0, -1,  3,  4,  5, -1, -1 },
+    { 0, -1,  4,  5,  2,  3,  1, -1, -1 },
+    { 3,  4,  1, -1,  0,  2,  5,  6, -1 },
+    { 2,  3, -1,  5,  7,  0,  1,  4,  6 },
+    { 3,  4,  1, -1,  0,  2,  5,  7,  6 },
+};
+
+static const int8_t dca_channel_reorder_nolfe[][9] = {
+    { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 0,  1, -1, -1, -1, -1, -1, -1, -1 },
+    { 1,  2,  0, -1, -1, -1, -1, -1, -1 },
+    { 0,  1,  2, -1, -1, -1, -1, -1, -1 },
+    { 1,  2,  0,  3, -1, -1, -1, -1, -1 },
+    { 0,  1,  2,  3, -1, -1, -1, -1, -1 },
+    { 1,  2,  0,  3,  4, -1, -1, -1, -1 },
+    { 2,  3,  0,  1,  4,  5, -1, -1, -1 },
+    { 1,  2,  0,  3,  4,  5, -1, -1, -1 },
+    { 0,  4,  5,  2,  3,  1, -1, -1, -1 },
+    { 3,  4,  1,  0,  2,  5,  6, -1, -1 },
+    { 2,  3,  5,  7,  0,  1,  4,  6, -1 },
+    { 3,  4,  1,  0,  2,  5,  7,  6, -1 },
+};
+
+typedef struct {
+    PutBitContext pb;
+    int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
+    int start[MAX_CHANNELS];
+    int frame_size;
+    int prim_channels;
+    int lfe_channel;
+    int sample_rate_code;
+    int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
+    int lfe_scale_factor;
+    int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
+
+    int a_mode;                         ///< audio channels arrangement
+    int num_channel;
+    int lfe_state;
+    int lfe_offset;
+    const int8_t *channel_order_tab;    ///< channel reordering table, lfe and non lfe
+
+    int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
+    int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /* [sample][channel][subband] */
+} DCAContext;
+
+static int32_t cos_table[128];
+
+static inline int32_t mul32(int32_t a, int32_t b)
+{
+    int64_t r = (int64_t) a * b;
+    /* round the result before truncating - improves accuracy */
+    return (r + 0x80000000) >> 32;
+}
+
+/* Integer version of the cosine modulated Pseudo QMF */
+
+static void qmf_init(void)
+{
+    int i;
+    int32_t c[17], s[17];
+    s[0] = 0;           /* sin(index * PI / 64) * 0x7fffffff */
+    c[0] = 0x7fffffff;  /* cos(index * PI / 64) * 0x7fffffff */
+
+    for (i = 1; i <= 16; i++) {
+        s[i] = 2 * (mul32(c[i - 1], 105372028)  + mul32(s[i - 1], 2144896908));
+        c[i] = 2 * (mul32(c[i - 1], 2144896908) - mul32(s[i - 1], 105372028));
+    }
+
+    for (i = 0; i < 16; i++) {
+        cos_table[i      ]  =  c[i]      >> 3; /* avoid output overflow */
+        cos_table[i +  16]  =  s[16 - i] >> 3;
+        cos_table[i +  32]  = -s[i]      >> 3;
+        cos_table[i +  48]  = -c[16 - i] >> 3;
+        cos_table[i +  64]  = -c[i]      >> 3;
+        cos_table[i +  80]  = -s[16 - i] >> 3;
+        cos_table[i +  96]  =  s[i]      >> 3;
+        cos_table[i + 112]  =  c[16 - i] >> 3;
+    }
+}
+
+static int32_t band_delta_factor(int band, int sample_num)
+{
+    int index = band * (2 * sample_num + 1);
+    if (band == 0)
+        return 0x07ffffff;
+    else
+        return cos_table[index & 127];
+}
+
+static void add_new_samples(DCAContext *c, const int32_t *in,
+                            int count, int channel)
+{
+    int i;
+
+    /* Place new samples into the history buffer */
+    for (i = 0; i < count; i++) {
+        c->history[channel][c->start[channel] + i] = in[i];
+        av_assert0(c->start[channel] + i < 512);
+    }
+    c->start[channel] += count;
+    if (c->start[channel] == 512)
+        c->start[channel] = 0;
+    av_assert0(c->start[channel] < 512);
+}
+
+static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
+                          int channel)
+{
+    int band, i, j, k;
+    int32_t resp;
+    int32_t accum[DCA_SUBBANDS_32] = {0};
+
+    add_new_samples(c, in, DCA_SUBBANDS_32, channel);
+
+    /* Calculate the dot product of the signal with the (possibly inverted)
+       reference decoder's response to this vector:
+       (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
+       so that -1.0 cancels 1.0 from the previous step */
+
+    for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
+        accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+    for (i = 0; i < c->start[channel]; k++, j++, i++)
+        accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] += mul32(c->history[channel][i], UnQMF[j]);
+
+    resp = 0;
+    /* TODO: implement FFT instead of this naive calculation */
+    for (band = 0; band < DCA_SUBBANDS_32; band++) {
+        for (j = 0; j < 32; j++)
+            resp += mul32(accum[j], band_delta_factor(band, j));
+
+        out[band] = (band & 2) ? (-resp) : resp;
+    }
+}
+
+static int32_t lfe_fir_64i[512];
+static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION])
+{
+    int i, j;
+    int channel = c->prim_channels;
+    int32_t accum = 0;
+
+    add_new_samples(c, in, LFE_INTERPOLATION, channel);
+    for (i = c->start[channel], j = 0; i < 512; i++, j++)
+        accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
+    for (i = 0; i < c->start[channel]; i++, j++)
+        accum += mul32(c->history[channel][i], lfe_fir_64i[j]);
+    return accum;
+}
+
+static void init_lfe_fir(void)
+{
+    static int initialized = 0;
+    int i;
+    if (initialized)
+        return;
+
+    for (i = 0; i < 512; i++)
+        lfe_fir_64i[i] = lfe_fir_64[i] * (1 << 25); //float -> int32_t
+    initialized = 1;
+}
+
+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, PCM_SAMPLES-1);
+
+    /* Primary frame byte size */
+    put_bits(&c->pb, 14, c->frame_size-1);
+
+    /* Audio channel arrangement: L + R (stereo) */
+    put_bits(&c->pb, 6, c->num_channel);
+
+    /* Core audio sampling frequency */
+    put_bits(&c->pb, 4, c->sample_rate_code);
+
+    /* Transmission bit rate: 1411.2 kbps */
+    put_bits(&c->pb, 5, 0x16); /* FIXME: magic number */
+
+    /* 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 interpolation factor=64 */
+    put_bits(&c->pb, 2, c->lfe_state);
+
+    /* 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->prim_channels - 1);
+
+    /* Subband activity count */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        put_bits(&c->pb, 5, DCA_SUBBANDS - 2);
+
+    /* High frequency VQ start subband */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        put_bits(&c->pb, 5, DCA_SUBBANDS - 1);
+
+    /* Joint intensity coding index: 0, 0 */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        put_bits(&c->pb, 3, 0);
+
+    /* Transient mode codebook: A4, A4 (arbitrary) */
+    for (ch = 0; ch < c->prim_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->prim_channels; ch++)
+        put_bits(&c->pb, 3, 6);
+
+    /* Bit allocation quantizer select: linear 5-bit */
+    for (ch = 0; ch < c->prim_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->prim_channels; ch++)
+            put_bits(&c->pb, bitlen[i], thr[i]);
+
+    /* Scale factor adjustment index: not transmitted */
+}
+
+/**
+ * 8-23 bits quantization
+ * @param sample
+ * @param bits
+ */
+static inline uint32_t quantize(int32_t sample, int bits)
+{
+    av_assert0(sample <    1 << (bits - 1));
+    av_assert0(sample >= -(1 << (bits - 1)));
+    sample &= sample & ((1 << bits) - 1);
+    return sample;
+}
+
+static inline int find_scale_factor7(int64_t max_value, int bits)
+{
+    int i = 0, j = 128, q;
+    max_value = ((max_value << 15) / lossy_quant[bits + 3]) >> (bits - 1);
+    while (i < j) {
+        q = (i + j) >> 1;
+        if (max_value < scale_factor_quant7[q])
+            j = q;
+        else
+            i = q + 1;
+    }
+    av_assert1(i < 128);
+    return i;
+}
+
+static inline void put_sample7(DCAContext *c, int64_t sample, int bits,
+                               int scale_factor)
+{
+    sample = (sample << 15) / ((int64_t) lossy_quant[bits + 3] * scale_factor_quant7[scale_factor]);
+    put_bits(&c->pb, bits, quantize((int) sample, bits));
+}
+
+static void put_subframe(DCAContext *c,
+                         int32_t subband_data[8 * SUBSUBFRAMES][MAX_CHANNELS][32],
+                         int subframe)
+{
+    int i, sub, ss, ch, max_value;
+    int32_t *lfe_data = c->lfe_data + 4 * SUBSUBFRAMES * subframe;
+
+    /* 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->prim_channels; ch++)
+        for (sub = 0; sub < DCA_SUBBANDS; sub++)
+            put_bits(&c->pb, 1, 0);
+
+    /* Prediction VQ addres: not transmitted */
+    /* Bit allocation index */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        for (sub = 0; sub < DCA_SUBBANDS; sub++)
+            put_bits(&c->pb, 5, QUANTIZER_BITS+3);
+
+    if (SUBSUBFRAMES > 1) {
+        /* Transition mode: none for each channel and subband */
+        for (ch = 0; ch < c->prim_channels; ch++)
+            for (sub = 0; sub < DCA_SUBBANDS; sub++)
+                put_bits(&c->pb, 1, 0); /* codebook A4 */
+    }
+
+    /* Determine scale_factor */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        for (sub = 0; sub < DCA_SUBBANDS; sub++) {
+            max_value = 0;
+            for (i = 0; i < 8 * SUBSUBFRAMES; i++)
+                max_value = FFMAX(max_value, FFABS(subband_data[i][ch][sub]));
+            c->scale_factor[ch][sub] = find_scale_factor7(max_value, QUANTIZER_BITS);
+        }
+
+    if (c->lfe_channel) {
+        max_value = 0;
+        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
+            max_value = FFMAX(max_value, FFABS(lfe_data[i]));
+        c->lfe_scale_factor = find_scale_factor7(max_value, LFE_BITS);
+    }
+
+    /* Scale factors: the same for each channel and subband,
+       encoded according to Table D.1.2 */
+    for (ch = 0; ch < c->prim_channels; ch++)
+        for (sub = 0; sub < DCA_SUBBANDS; sub++)
+            put_bits(&c->pb, 7, c->scale_factor[ch][sub]);
+
+    /* 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 */
+    if (c->lfe_channel) {
+        for (i = 0; i < 4 * SUBSUBFRAMES; i++)
+            put_sample7(c, lfe_data[i], LFE_BITS, c->lfe_scale_factor);
+        put_bits(&c->pb, 8, c->lfe_scale_factor);
+    }
+
+    /* Audio data (subsubframes) */
+
+    for (ss = 0; ss < SUBSUBFRAMES ; ss++)
+        for (ch = 0; ch < c->prim_channels; ch++)
+            for (sub = 0; sub < DCA_SUBBANDS; sub++)
+                for (i = 0; i < 8; i++)
+                    put_sample7(c, subband_data[ss * 8 + i][ch][sub], QUANTIZER_BITS, c->scale_factor[ch][sub]);
+
+    /* DSYNC */
+    put_bits(&c->pb, 16, 0xffff);
+}
+
+static void put_frame(DCAContext *c,
+                      int32_t subband_data[PCM_SAMPLES][MAX_CHANNELS][32],
+                      uint8_t *frame)
+{
+    int i;
+    init_put_bits(&c->pb, frame + DCA_HEADER_SIZE, DCA_MAX_FRAME_SIZE-DCA_HEADER_SIZE);
+
+    put_primary_audio_header(c);
+    for (i = 0; i < SUBFRAMES; i++)
+        put_subframe(c, &subband_data[SUBSUBFRAMES * 8 * i], i);
+
+    flush_put_bits(&c->pb);
+    c->frame_size = (put_bits_count(&c->pb) >> 3) + DCA_HEADER_SIZE;
+
+    init_put_bits(&c->pb, frame, DCA_HEADER_SIZE);
+    put_frame_header(c);
+    flush_put_bits(&c->pb);
+}
+
+static int encode_frame(AVCodecContext *avctx, uint8_t *frame,
+                        int buf_size, void *data)
+{
+    int i, k, channel;
+    DCAContext *c = avctx->priv_data;
+    int16_t *samples = data;
+    int real_channel = 0;
+
+    for (i = 0; i < PCM_SAMPLES; i ++) { /* i is the decimated sample number */
+        for (channel = 0; channel < c->prim_channels + 1; channel++) {
+            /* Get 32 PCM samples */
+            for (k = 0; k < 32; k++) { /* k is the sample number in a 32-sample block */
+                c->pcm[k] = samples[avctx->channels * (32 * i + k) + channel] << 16;
+            }
+            /* Put subband samples into the proper place */
+            real_channel = c->channel_order_tab[channel];
+            if (real_channel >= 0) {
+                qmf_decompose(c, c->pcm, &c->subband[i][real_channel][0], real_channel);
+            }
+        }
+    }
+
+    if (c->lfe_channel) {
+        for (i = 0; i < PCM_SAMPLES / 2; i++) {
+            for (k = 0; k < LFE_INTERPOLATION; k++) /* k is the sample number in a 32-sample block */
+                c->pcm[k] = samples[avctx->channels * (LFE_INTERPOLATION*i+k) + c->lfe_offset] << 16;
+            c->lfe_data[i] = lfe_downsample(c, c->pcm);
+        }
+    }
+
+    put_frame(c, c->subband, frame);
+
+    return c->frame_size;
+}
+
+static int encode_init(AVCodecContext *avctx)
+{
+    DCAContext *c = avctx->priv_data;
+    int i;
+
+    c->prim_channels = avctx->channels;
+    c->lfe_channel   = (avctx->channels == 3 || avctx->channels == 6);
+
+    switch (avctx->channel_layout) {
+    case AV_CH_LAYOUT_STEREO:       c->a_mode = 2; c->num_channel = 2; break;
+    case AV_CH_LAYOUT_5POINT0:      c->a_mode = 9; c->num_channel = 9; break;
+    case AV_CH_LAYOUT_5POINT1:      c->a_mode = 9; c->num_channel = 9; break;
+    case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9; break;
+    case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9; break;
+    default:
+    av_log(avctx, AV_LOG_ERROR,
+           "Only stereo, 5.0, 5.1 channel layouts supported at the moment!\n");
+    return AVERROR_PATCHWELCOME;
+    }
+
+    if (c->lfe_channel) {
+        init_lfe_fir();
+        c->prim_channels--;
+        c->channel_order_tab = dca_channel_reorder_lfe[c->a_mode];
+        c->lfe_state         = LFE_PRESENT;
+        c->lfe_offset        = dca_lfe_index[c->a_mode];
+    } else {
+        c->channel_order_tab = dca_channel_reorder_nolfe[c->a_mode];
+        c->lfe_state         = LFE_MISSING;
+    }
+
+    for (i = 0; i < 16; i++) {
+        if (dca_sample_rates[i] && (dca_sample_rates[i] == avctx->sample_rate))
+            break;
+    }
+    if (i == 16) {
+        av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported, only ", avctx->sample_rate);
+        for (i = 0; i < 16; i++)
+            av_log(avctx, AV_LOG_ERROR, "%d, ", dca_sample_rates[i]);
+        av_log(avctx, AV_LOG_ERROR, "supported.\n");
+        return -1;
+    }
+    c->sample_rate_code = i;
+
+    avctx->frame_size = 32 * PCM_SAMPLES;
+
+    if (!cos_table[127])
+        qmf_init();
+    return 0;
+}
+
+AVCodec ff_dca_encoder = {
+    .name           = "dca",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = CODEC_ID_DTS,
+    .priv_data_size = sizeof(DCAContext),
+    .init           = encode_init,
+    .encode         = encode_frame,
+    .capabilities   = CODEC_CAP_EXPERIMENTAL,
+    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
+};