diff mbox series

[FFmpeg-devel,3/3] avcodec: add FastAudio decoder

Message ID 20200824232858.18938-3-onemda@gmail.com
State Accepted
Headers show
Series [FFmpeg-devel,1/3] avformat: add moflex demuxer
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andriy/default pending
andriy/make success Make finished
andriy/make_fate success Make fate finished

Commit Message

Paul B Mahol Aug. 24, 2020, 11:28 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavcodec/Makefile     |   1 +
 libavcodec/allcodecs.c  |   1 +
 libavcodec/codec_desc.c |   7 ++
 libavcodec/codec_id.h   |   1 +
 libavcodec/fastaudio.c  | 200 ++++++++++++++++++++++++++++++++++++++++
 libavcodec/utils.c      |   2 +
 6 files changed, 212 insertions(+)
 create mode 100644 libavcodec/fastaudio.c
diff mbox series

Patch

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 191c4e0a7c..6f75f26c84 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -317,6 +317,7 @@  OBJS-$(CONFIG_ESCAPE124_DECODER)       += escape124.o
 OBJS-$(CONFIG_ESCAPE130_DECODER)       += escape130.o
 OBJS-$(CONFIG_EVRC_DECODER)            += evrcdec.o acelp_vectors.o lsp.o
 OBJS-$(CONFIG_EXR_DECODER)             += exr.o exrdsp.o
+OBJS-$(CONFIG_FASTAUDIO_DECODER)       += fastaudio.o
 OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 084a289c9e..3920eb37ce 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -430,6 +430,7 @@  extern AVCodec ff_dst_decoder;
 extern AVCodec ff_eac3_encoder;
 extern AVCodec ff_eac3_decoder;
 extern AVCodec ff_evrc_decoder;
+extern AVCodec ff_fastaudio_decoder;
 extern AVCodec ff_ffwavesynth_decoder;
 extern AVCodec ff_flac_encoder;
 extern AVCodec ff_flac_decoder;
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index b5bc5c3c71..9a3eaf7d98 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3130,6 +3130,13 @@  static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("CRI HCA"),
         .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
     },
+    {
+        .id        = AV_CODEC_ID_FASTAUDIO,
+        .type      = AVMEDIA_TYPE_AUDIO,
+        .name      = "fastaudio",
+        .long_name = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"),
+        .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
+    },
 
     /* subtitle codecs */
     {
diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h
index aac1174f28..aac7f63eb6 100644
--- a/libavcodec/codec_id.h
+++ b/libavcodec/codec_id.h
@@ -504,6 +504,7 @@  enum AVCodecID {
     AV_CODEC_ID_MPEGH_3D_AUDIO,
     AV_CODEC_ID_SIREN,
     AV_CODEC_ID_HCA,
+    AV_CODEC_ID_FASTAUDIO,
 
     /* subtitle codecs */
     AV_CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.
diff --git a/libavcodec/fastaudio.c b/libavcodec/fastaudio.c
new file mode 100644
index 0000000000..354336d17e
--- /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 tblVal)
+{
+    result[i * 64 + pads[i] + j * 3] = tblVal * (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 tblVal = 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, tblVal);
+                    if (j % 10 == 9)
+                        tmp = 4 * tmp + read_bits(2, &pos, src);
+                    if (j == 20)
+                        index5 = 2 * index5 + tmp % 2;
+                }
+
+                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 },
+};
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 187ce230b8..2e5185f364 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -1664,6 +1664,8 @@  static int get_audio_frame_duration(enum AVCodecID id, int sr, int ch, int ba,
         if (ch > 0 && ch < INT_MAX/16) {
             /* calc from frame_bytes and channels */
             switch (id) {
+            case AV_CODEC_ID_FASTAUDIO:
+                return frame_bytes / (40 * ch) * 256;
             case AV_CODEC_ID_ADPCM_IMA_MOFLEX:
                 return (frame_bytes - 4 * ch) / (128 * ch) * 256;
             case AV_CODEC_ID_ADPCM_AFC: