avcodec: add FastAudio decoder

1 files changed, 200 insertions, 0 deletions

diff --git a/libavcodec/fastaudio.c b/libavcodec/fastaudio.c
new file mode 100644
index 0000000000..de006acd9b
--- /dev/null
+++ b/libavcodec/fastaudio.c
@@ -0,0 +1,200 @@
+/*
+ * MOFLEX Fast Audio decoder
+ * Copyright (c) 2020 Paul B Mahol
+ *
+ * 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/intreadwrite.h"
+
+#include "avcodec.h"
+#include "bytestream.h"
+#include "internal.h"
+#include "mathops.h"
+
+typedef struct ChannelItems {
+    float f[8];
+    float last;
+} ChannelItems;
+
+typedef struct FastAudioContext {
+    float table[8][64];
+
+    ChannelItems *ch;
+} FastAudioContext;
+
+static av_cold int fastaudio_init(AVCodecContext *avctx)
+{
+    FastAudioContext *s = avctx->priv_data;
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+
+    for (int i = 0; i < 8; i++)
+        s->table[0][i] = (i - 159.5f) / 160.f;
+    for (int i = 0; i < 11; i++)
+        s->table[0][i + 8] = (i - 37.5f) / 40.f;
+    for (int i = 0; i < 27; i++)
+        s->table[0][i + 8 + 11] = (i - 13.f) / 20.f;
+    for (int i = 0; i < 11; i++)
+        s->table[0][i + 8 + 11 + 27] = (i + 27.5f) / 40.f;
+    for (int i = 0; i < 7; i++)
+        s->table[0][i + 8 + 11 + 27 + 11] = (i + 152.5f) / 160.f;
+
+    memcpy(s->table[1], s->table[0], sizeof(s->table[0]));
+
+    for (int i = 0; i < 7; i++)
+        s->table[2][i] = (i - 33.5f) / 40.f;
+    for (int i = 0; i < 25; i++)
+        s->table[2][i + 7] = (i - 13.f) / 20.f;
+
+    for (int i = 0; i < 32; i++)
+        s->table[3][i] = -s->table[2][31 - i];
+
+    for (int i = 0; i < 16; i++)
+        s->table[4][i] = i * 0.22f / 3.f - 0.6f;
+
+    for (int i = 0; i < 16; i++)
+        s->table[5][i] = i * 0.20f / 3.f - 0.3f;
+
+    for (int i = 0; i < 8; i++)
+        s->table[6][i] = i * 0.36f / 3.f - 0.4f;
+
+    for (int i = 0; i < 8; i++)
+        s->table[7][i] = i * 0.34f / 3.f - 0.2f;
+
+    s->ch = av_calloc(avctx->channels, sizeof(*s->ch));
+    if (!s->ch)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int read_bits(int bits, int *ppos, unsigned *src)
+{
+    int r, pos;
+
+    pos = *ppos;
+    pos += bits;
+    r = src[(pos - 1) / 32] >> (32 - pos % 32);
+    *ppos = pos;
+
+    return r & ((1 << (bits % 32)) - 1);
+}
+
+static const uint8_t bits[8] = { 6, 6, 5, 5, 4, 0, 3, 3, };
+
+static void set_sample(int i, int j, int v, float *result, int *pads, float value)
+{
+    result[i * 64 + pads[i] + j * 3] = value * (2 * v - 7);
+}
+
+static int fastaudio_decode(AVCodecContext *avctx, void *data,
+                            int *got_frame, AVPacket *pkt)
+{
+    FastAudioContext *s = avctx->priv_data;
+    GetByteContext gb;
+    AVFrame *frame = data;
+    int subframes;
+    int ret;
+
+    subframes = pkt->size / (40 * avctx->channels);
+    frame->nb_samples = subframes * 256;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+
+    bytestream2_init(&gb, pkt->data, pkt->size);
+
+    for (int subframe = 0; subframe < subframes; subframe++) {
+        for (int channel = 0; channel < avctx->channels; channel++) {
+            ChannelItems *ch = &s->ch[channel];
+            float result[256] = { 0 };
+            unsigned src[10];
+            int inds[4], pads[4];
+            float m[8];
+            int pos = 0;
+
+            for (int i = 0; i < 10; i++)
+                src[i] = bytestream2_get_le32(&gb);
+
+            for (int i = 0; i < 8; i++)
+                m[7 - i] = s->table[i][read_bits(bits[i], &pos, src)];
+
+            for (int i = 0; i < 4; i++)
+                inds[3 - i] = read_bits(6, &pos, src);
+
+            for (int i = 0; i < 4; i++)
+                pads[3 - i] = read_bits(2, &pos, src);
+
+            for (int i = 0, index5 = 0; i < 4; i++) {
+                float value = av_int2float((inds[i] + 1) << 20) * powf(2.f, 116.f);
+
+                for (int j = 0, tmp = 0; j < 21; j++) {
+                    set_sample(i, j, j == 20 ? tmp / 2 : read_bits(3, &pos, src), result, pads, value);
+                    if (j % 10 == 9)
+                        tmp = 4 * tmp + read_bits(2, &pos, src);
+                    if (j == 20)
+                        index5 = FFMIN(2 * index5 + tmp % 2, 63);
+                }
+
+                m[2] = s->table[5][index5];
+            }
+
+            for (int i = 0; i < 256; i++) {
+                float x = result[i];
+
+                for (int j = 0; j < 8; j++) {
+                    x -= m[j] * ch->f[j];
+                    ch->f[j] += m[j] * x;
+                }
+
+                memmove(&ch->f[0], &ch->f[1], sizeof(float) * 7);
+                ch->f[7] = x;
+                ch->last = x + ch->last * 0.86f;
+                result[i] = ch->last * 2.f;
+            }
+
+            memcpy(frame->extended_data[channel] + 1024 * subframe, result, 256 * sizeof(float));
+        }
+    }
+
+    *got_frame = 1;
+
+    return pkt->size;
+}
+
+static av_cold int fastaudio_close(AVCodecContext *avctx)
+{
+    FastAudioContext *s = avctx->priv_data;
+
+    av_freep(&s->ch);
+
+    return 0;
+}
+
+AVCodec ff_fastaudio_decoder = {
+    .name           = "fastaudio",
+    .long_name      = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_FASTAUDIO,
+    .priv_data_size = sizeof(FastAudioContext),
+    .init           = fastaudio_init,
+    .decode         = fastaudio_decode,
+    .close          = fastaudio_close,
+    .capabilities   = AV_CODEC_CAP_DR1,
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
+};