@@ -604,7 +604,7 @@ following image formats are supported:

     @tab Codec used in Delphine Software International games.

 @item COOK                   @tab     @tab  X

     @tab All versions except 5.1 are supported.

-@item DCA (DTS Coherent Acoustics)  @tab     @tab  X

+@item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X

 @item DPCM id RoQ            @tab  X  @tab  X

     @tab Used in Quake III, Jedi Knight 2, other computer games.

 @item DPCM Interplay         @tab     @tab  X

@@ -105,6 +105,7 @@ OBJS-$(CONFIG_COOK_DECODER)            += cook.o

 OBJS-$(CONFIG_CSCD_DECODER)            += cscd.o

 OBJS-$(CONFIG_CYUV_DECODER)            += cyuv.o

 OBJS-$(CONFIG_DCA_DECODER)             += dca.o synth_filter.o dcadsp.o

+OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o

 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o

 OBJS-$(CONFIG_DNXHD_DECODER)           += dnxhddec.o dnxhddata.o

 OBJS-$(CONFIG_DNXHD_ENCODER)           += dnxhdenc.o dnxhddata.o       \

@@ -246,7 +246,7 @@ void avcodec_register_all(void)

     REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);

     REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);

     REGISTER_DECODER (COOK, cook);

-    REGISTER_DECODER (DCA, dca);

+    REGISTER_ENCDEC  (DCA, dca);

     REGISTER_DECODER (DSICINAUDIO, dsicinaudio);

     REGISTER_DECODER (EAC3, eac3);

     REGISTER_ENCDEC  (FLAC, flac);

new file mode 100644

@@ -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},

+};

new file mode 100644

@@ -0,0 +1,544 @@

+/*

+ * DCA encoder tables

+ * Copyright (C) 2008 Alexander E. Patrakov

+ *

+ * 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

+ */

+

+#ifndef AVCODEC_DCAENC_H

+#define AVCODEC_DCAENC_H

+

+/* This is a scaled version of the response of the reference decoder to

+   this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )

+   */

+

+static const int32_t UnQMF[512] = {

+    7,

+    4,

+    -961,

+    -2844,

+    -8024,

+    -18978,

+    -32081,

+    -15635,

+    -16582,

+    -18359,

+    -17180,

+    -14868,

+    -11664,

+    -8051,

+    -4477,

+    -1327,

+    -1670,

+    -6019,

+    -11590,

+    -18030,

+    -24762,

+    -30965,

+    -35947,

+    -36145,

+    -37223,

+    -86311,

+    -57024,

+    -27215,

+    -11274,

+    -4684,

+    42,

+    108,

+    188,

+    250,

+    -1007,

+    -596,

+    -2289,

+    -12218,

+    -27191,

+    -124367,

+    -184256,

+    -250538,

+    -323499,

+    -397784,

+    -468855,

+    -532072,

+    -583000,

+    -618041,

+    -777916,

+    -783868,

+    -765968,

+    -724740,

+    -662468,

+    -583058,

+    -490548,

+    -401623,

+    -296090,

+    -73154,

+    -36711,

+    -7766,

+    -2363,

+    -4905,

+    2388,

+    2681,

+    5651,

+    4086,

+    71110,

+    139742,

+    188067,

+    151237,

+    101355,

+    309917,

+    343690,

+    358839,

+    357555,

+    334606,

+    289625,

+    224152,

+    142063,

+    48725,

+    74996,

+    238425,

+    411666,

+    584160,

+    744276,

+    880730,

+    983272,

+    1041933,

+    1054396,

+    789531,

+    851022,

+    864032,

+    675431,

+    418134,

+    35762,

+    66911,

+    103502,

+    136403,

+    -55147,

+    -245269,

+    -499595,

+    -808470,

+    -1136858,

+    -2010912,

+    -2581654,

+    -3151901,

+    -3696328,

+    -4196599,

+    -4633761,

+    -4993229,

+    -5262495,

+    -5436311,

+    -477650,

+    -901314,

+    -1308090,

+    -1677468,

+    -1985525,

+    -2212848,

+    -2341196,

+    -2373915,

+    -2269552,

+    -2620489,

+    -2173858,

+    -1629954,

+    -946595,

+    -193499,

+    1119459,

+    1138657,

+    1335311,

+    1126544,

+    2765033,

+    3139603,

+    3414913,

+    3599213,

+    3676363,

+    3448981,

+    3328726,

+    3111551,

+    2810887,

+    2428657,

+    1973684,

+    1457278,

+    893848,

+    300995,

+    -292521,

+    -867621,

+    -1404936,

+    -1871278,

+    -2229831,

+    -2440932,

+    -2462684,

+    -2255006,

+    -1768898,

+    -1079574,

+    82115,

+    1660302,

+    3660715,

+    6123610,

+    8329598,

+    11888744,

+    15722147,

+    19737089,

+    25647773,

+    31039399,

+    36868007,

+    43124253,

+    49737161,

+    56495958,

+    63668945,

+    71039511,

+    78540240,

+    86089058,

+    93600041,

+    100981151,

+    108136061,

+    114970055,

+    121718321,

+    127566038,

+    132774642,

+    137247294,

+    140894737,

+    143635018,

+    145395599,

+    146114032,

+    145742999,

+    144211606,

+    141594341,

+    137808404,

+    132914122,

+    126912246,

+    120243281,

+    112155281,

+    103338368,

+    93904953,

+    83439152,

+    72921548,

+    62192990,

+    51434918,

+    40894003,

+    30786726,

+    21384955,

+    12939112,

+    5718193,

+    -5790,

+    -3959261,

+    -5870978,

+    -5475538,

+    -2517061,

+    3247310,

+    12042937,

+    24076729,

+    39531397,

+    58562863,

+    81297002,

+    107826748,

+    138209187,

+    172464115,

+    210569037,

+    252468018,

+    298045453,

+    347168648,

+    399634888,

+    455137189,

+    513586535,

+    574537650,

+    637645129,

+    702597163,

+    768856566,

+    836022040,

+    903618096,

+    971159680,