avcodec: add Direct Stream Transfer (DST) decoder

Signed-off-by: Paul B Mahol <onemda@gmail.com>

1 files changed, 374 insertions, 0 deletions

diff --git a/libavcodec/dstdec.c b/libavcodec/dstdec.c
new file mode 100644
index 0000000000..13be24a057
--- /dev/null
+++ b/libavcodec/dstdec.c
@@ -0,0 +1,374 @@
+/*
+ * Direct Stream Transfer (DST) decoder
+ * Copyright (c) 2014 Peter Ross <pross@xvid.org>
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Direct Stream Transfer (DST) decoder
+ * ISO/IEC 14496-3 Part 3 Subpart 10: Technical description of lossless coding of oversampled audio
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/intreadwrite.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "avcodec.h"
+#include "golomb.h"
+#include "mathops.h"
+#include "dsd.h"
+
+#define DST_MAX_CHANNELS 6
+#define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS)
+
+#define DSD_FS44(sample_rate) (sample_rate * 8 / 44100)
+
+#define DST_SAMPLES_PER_FRAME(sample_rate) (588 * DSD_FS44(sample_rate))
+
+static const int8_t fsets_code_pred_coeff[3][3] = {
+    {  -8 },
+    { -16,  8 },
+    {  -9, -5, 6 },
+};
+
+static const int8_t probs_code_pred_coeff[3][3] = {
+    {  -8 },
+    { -16,  8 },
+    { -24, 24, -8 },
+};
+
+typedef struct ArithCoder {
+    unsigned int a;
+    unsigned int c;
+} ArithCoder;
+
+typedef struct Table {
+    unsigned int elements;
+    unsigned int length[DST_MAX_ELEMENTS];
+    int coeff[DST_MAX_ELEMENTS][128];
+} Table;
+
+typedef struct DSTContext {
+    AVClass *class;
+
+    GetBitContext gb;
+    ArithCoder ac;
+    Table fsets, probs;
+    DECLARE_ALIGNED(64, uint8_t, status)[DST_MAX_CHANNELS][16];
+    DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256];
+    DSDContext dsdctx[DST_MAX_CHANNELS];
+} DSTContext;
+
+static av_cold int decode_init(AVCodecContext *avctx)
+{
+    DSTContext *s = avctx->priv_data;
+    int i;
+
+    if (avctx->channels > DST_MAX_CHANNELS) {
+        avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
+
+    for (i = 0; i < avctx->channels; i++)
+        memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf));
+
+    ff_init_dsd_data();
+
+    return 0;
+}
+
+static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels)
+{
+    int ch;
+    t->elements = 1;
+    map[0] = 0;
+    if (!get_bits1(gb)) {
+        for (ch = 1; ch < channels; ch++) {
+            int bits = av_log2(t->elements) + 1;
+            map[ch] = get_bits(gb, bits);
+            if (map[ch] == t->elements) {
+                t->elements++;
+                if (t->elements >= DST_MAX_ELEMENTS)
+                    return AVERROR_INVALIDDATA;
+            } else if (map[ch] > t->elements) {
+                return AVERROR_INVALIDDATA;
+            }
+        }
+    } else {
+        memset(map, 0, sizeof(*map) * DST_MAX_CHANNELS);
+    }
+    return 0;
+}
+
+static av_always_inline int get_sr_golomb_dst(GetBitContext *gb, unsigned int k)
+{
+    int v = get_ur_golomb(gb, k, get_bits_left(gb), 0);
+    if (v && get_bits1(gb))
+        v = -v;
+    return v;
+}
+
+static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements,
+                               int coeff_bits, int is_signed, int offset)
+{
+    int i;
+
+    for (i = 0; i < elements; i++) {
+        dst[i] = (is_signed ? get_sbits(gb, coeff_bits) : get_bits(gb, coeff_bits)) + offset;
+    }
+}
+
+static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3],
+                      int length_bits, int coeff_bits, int is_signed, int offset)
+{
+    unsigned int i, j, k;
+    for (i = 0; i < t->elements; i++) {
+        t->length[i] = get_bits(gb, length_bits) + 1;
+        if (!get_bits1(gb)) {
+            read_uncoded_coeff(gb, t->coeff[i], t->length[i], coeff_bits, is_signed, offset);
+        } else {
+            int method = get_bits(gb, 2), lsb_size;
+            if (method == 3)
+                return AVERROR_INVALIDDATA;
+
+            read_uncoded_coeff(gb, t->coeff[i], method + 1, coeff_bits, is_signed, offset);
+
+            lsb_size  = get_bits(gb, 3);
+            for (j = method + 1; j < t->length[i]; j++) {
+                int c, x = 0;
+                for (k = 0; k < method + 1; k++)
+                    x += code_pred_coeff[method][k] * t->coeff[i][j - k - 1];
+                c = get_sr_golomb_dst(gb, lsb_size);
+                if (x >= 0)
+                    c -= (x + 4) / 8;
+                else
+                    c += (-x + 3) / 8;
+                t->coeff[i][j] = c;
+            }
+        }
+    }
+    return 0;
+}
+
+static void ac_init(ArithCoder *ac, GetBitContext *gb)
+{
+    ac->a = 4095;
+    ac->c = get_bits(gb, 12);
+}
+
+static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)
+{
+    unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1);
+    unsigned int q = k * p;
+    unsigned int a_q = ac->a - q;
+
+    *e = ac->c < a_q;
+    if (*e) {
+        ac->a  = a_q;
+    } else {
+        ac->a  = q;
+        ac->c -= a_q;
+    }
+
+    if (ac->a < 2048) {
+        int n = 11 - av_log2(ac->a);
+        ac->a <<= n;
+        ac->c = (ac->c << n) | get_bits(gb, n);
+    }
+}
+
+static uint8_t prob_dst_x_bit(int c)
+{
+    return (ff_reverse[c & 127] >> 1) + 1;
+}
+
+static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)
+{
+    int i, j, k, l;
+
+    for (i = 0; i < fsets->elements; i++) {
+        int length = fsets->length[i];
+
+        for (j = 0; j < 16; j++) {
+            int total = av_clip(length - j * 8, 0, 8);
+
+            for (k = 0; k < 256; k++) {
+                int v = 0;
+
+                for (l = 0; l < total; l++)
+                    v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];
+                table[i][j][k] = v;
+            }
+        }
+    }
+}
+
+static int decode_frame(AVCodecContext *avctx, void *data,
+                        int *got_frame_ptr, AVPacket *avpkt)
+{
+    unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate);
+    unsigned map_ch_to_felem[DST_MAX_CHANNELS];
+    unsigned map_ch_to_pelem[DST_MAX_CHANNELS];
+    unsigned i, ch, same_map, dst_x_bit;
+    unsigned half_prob[DST_MAX_CHANNELS];
+    const int channels = avctx->channels;
+    DSTContext *s = avctx->priv_data;
+    GetBitContext *gb = &s->gb;
+    ArithCoder *ac = &s->ac;
+    AVFrame *frame = data;
+    uint8_t *dsd;
+    float *pcm;
+    int ret;
+
+    if (avpkt->size <= 1)
+        return AVERROR_INVALIDDATA;
+
+    frame->nb_samples = samples_per_frame / 8;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+    dsd = frame->data[0];
+    pcm = (float *)frame->data[0];
+
+    if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)
+        return ret;
+
+    if (!get_bits1(gb)) {
+        skip_bits1(gb);
+        if (get_bits(gb, 6))
+            return AVERROR_INVALIDDATA;
+        memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * avctx->channels));
+        goto dsd;
+    }
+
+    /* Segmentation (10.4, 10.5, 10.6) */
+
+    if (!get_bits1(gb)) {
+        avpriv_request_sample(avctx, "Not Same Segmentation");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (!get_bits1(gb)) {
+        avpriv_request_sample(avctx, "Not Same Segmentation For All Channels");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    if (!get_bits1(gb)) {
+        avpriv_request_sample(avctx, "Not End Of Channel Segmentation");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    /* Mapping (10.7, 10.8, 10.9) */
+
+    same_map = get_bits1(gb);
+
+    if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, avctx->channels)) < 0)
+        return ret;
+
+    if (same_map) {
+        s->probs.elements = s->fsets.elements;
+        memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem));
+    } else {
+        avpriv_request_sample(avctx, "Not Same Mapping");
+        if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, avctx->channels)) < 0)
+            return ret;
+    }
+
+    /* Half Probability (10.10) */
+
+    for (ch = 0; ch < avctx->channels; ch++)
+        half_prob[ch] = get_bits1(gb);
+
+    /* Filter Coef Sets (10.12) */
+
+    read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
+
+    /* Probability Tables (10.13) */
+
+    read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
+
+    /* Arithmetic Coded Data (10.11) */
+
+    if (get_bits1(gb))
+        return AVERROR_INVALIDDATA;
+    ac_init(ac, gb);
+
+    build_filter(s->filter, &s->fsets);
+
+    memset(s->status, 0xAA, sizeof(s->status));
+    memset(dsd, 0, frame->nb_samples * 4 * avctx->channels);
+
+    ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit);
+
+    for (i = 0; i < samples_per_frame; i++) {
+        for (ch = 0; ch < channels; ch++) {
+            const unsigned felem = map_ch_to_felem[ch];
+            const int16_t (*filter)[256] = s->filter[felem];
+            uint8_t *status = s->status[ch];
+            int prob, residual, v;
+
+#define F(x) filter[(x)][status[(x)]]
+            const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) +
+                                    F( 4) + F( 5) + F( 6) + F( 7) +
+                                    F( 8) + F( 9) + F(10) + F(11) +
+                                    F(12) + F(13) + F(14) + F(15);
+#undef F
+
+            if (!half_prob[ch] || i >= s->fsets.length[felem]) {
+                unsigned pelem = map_ch_to_pelem[ch];
+                unsigned index = FFABS(predict) >> 3;
+                prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)];
+            } else {
+                prob = 128;
+            }
+
+            ac_get(ac, gb, prob, &residual);
+            v = ((predict >> 15) ^ residual) & 1;
+            dsd[((i >> 3) * channels + ch) << 2] |= v << (7 - (i & 0x7 ));
+
+            AV_WN64A(status + 8, (AV_RN64A(status + 8) << 1) | ((AV_RN64A(status) >> 63) & 1));
+            AV_WN64A(status, (AV_RN64A(status) << 1) | v);
+        }
+    }
+
+dsd:
+    for (i = 0; i < avctx->channels; i++) {
+        ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
+                             frame->data[0] + i * 4,
+                             avctx->channels * 4, pcm + i, avctx->channels);
+    }
+
+    *got_frame_ptr = 1;
+
+    return avpkt->size;
+}
+
+AVCodec ff_dst_decoder = {
+    .name           = "dst",
+    .long_name      = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_DST,
+    .priv_data_size = sizeof(DSTContext),
+    .init           = decode_init,
+    .decode         = decode_frame,
+    .capabilities   = AV_CODEC_CAP_DR1,
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+                                                      AV_SAMPLE_FMT_NONE },
+};