avcodec: add WavArc decoder

8 files changed, 473 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index cdbe43eac1..2d2206e1ba 100644
--- a/Changelog
+++ b/Changelog
@@ -37,6 +37,7 @@ version <next>:
 - XMD ADPCM decoder and demuxer
 - media100 to mjpegb bsf
 - ffmpeg CLI new option: -fix_sub_duration_heartbeat
+- WavArc decoder
 
 
 version 5.1:
diff --git a/doc/general_contents.texi b/doc/general_contents.texi
index 87e180c979..84df3432cf 100644
--- a/doc/general_contents.texi
+++ b/doc/general_contents.texi
@@ -1356,6 +1356,7 @@ following image formats are supported:
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
 @item Voxware MetaSound      @tab     @tab  X
+@item Waveform Archiver      @tab     @tab  X
 @item WavPack                @tab  X  @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 1fb963f820..4971832ff4 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -781,6 +781,7 @@ OBJS-$(CONFIG_VP9_V4L2M2M_DECODER)     += v4l2_m2m_dec.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_VQC_DECODER)             += vqcdec.o
 OBJS-$(CONFIG_WADY_DPCM_DECODER)       += dpcm.o
+OBJS-$(CONFIG_WAVARC_DECODER)          += wavarc.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o wavpackdata.o dsd.o
 OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackdata.o wavpackenc.o
 OBJS-$(CONFIG_WBMP_DECODER)            += wbmpdec.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index ff82423a88..b80b6983e9 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -538,6 +538,7 @@ extern const FFCodec ff_twinvq_decoder;
 extern const FFCodec ff_vmdaudio_decoder;
 extern const FFCodec ff_vorbis_encoder;
 extern const FFCodec ff_vorbis_decoder;
+extern const FFCodec ff_wavarc_decoder;
 extern const FFCodec ff_wavpack_encoder;
 extern const FFCodec ff_wavpack_decoder;
 extern const FFCodec ff_wmalossless_decoder;
diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c
index 8ab228d846..57d0f98211 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3353,6 +3353,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
         .long_name = NULL_IF_CONFIG_SMALL("FTR Voice"),
         .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSY,
     },
+    {
+        .id        = AV_CODEC_ID_WAVARC,
+        .type      = AVMEDIA_TYPE_AUDIO,
+        .name      = "wavarc",
+        .long_name = NULL_IF_CONFIG_SMALL("Waveform Archiver"),
+        .props     = AV_CODEC_PROP_INTRA_ONLY | AV_CODEC_PROP_LOSSLESS,
+    },
 
     /* subtitle codecs */
     {
diff --git a/libavcodec/codec_id.h b/libavcodec/codec_id.h
index 0c574c9619..ad1131b464 100644
--- a/libavcodec/codec_id.h
+++ b/libavcodec/codec_id.h
@@ -536,6 +536,7 @@ enum AVCodecID {
     AV_CODEC_ID_MISC4,
     AV_CODEC_ID_APAC,
     AV_CODEC_ID_FTR,
+    AV_CODEC_ID_WAVARC,
 
     /* subtitle codecs */
     AV_CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 499c6bb175..310c80eeef 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -29,7 +29,7 @@
 
 #include "version_major.h"
 
-#define LIBAVCODEC_VERSION_MINOR  61
+#define LIBAVCODEC_VERSION_MINOR  62
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavcodec/wavarc.c b/libavcodec/wavarc.c
new file mode 100644
index 0000000000..898c3c2055
--- /dev/null
+++ b/libavcodec/wavarc.c
@@ -0,0 +1,460 @@
+/*
+ * WavArc audio decoder
+ * Copyright (c) 2023 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/internal.h"
+#include "libavutil/intreadwrite.h"
+#include "avcodec.h"
+#include "codec_internal.h"
+#include "decode.h"
+#include "get_bits.h"
+#include "bytestream.h"
+#include "mathops.h"
+#include "unary.h"
+
+typedef struct WavArcContext {
+    GetBitContext gb;
+
+    int shift;
+    int nb_samples;
+    int offset;
+
+    int eof;
+    int skip;
+    uint8_t *bitstream;
+    int64_t max_framesize;
+    int bitstream_size;
+    int bitstream_index;
+
+    int pred[2][70];
+    int filter[2][70];
+    int samples[2][640];
+} WavArcContext;
+
+static av_cold int wavarc_init(AVCodecContext *avctx)
+{
+    WavArcContext *s = avctx->priv_data;
+
+    if (avctx->extradata_size < 44)
+        return AVERROR_INVALIDDATA;
+    if (AV_RL32(avctx->extradata + 16) != MKTAG('R','I','F','F'))
+        return AVERROR_INVALIDDATA;
+    if (AV_RL32(avctx->extradata + 24) != MKTAG('W','A','V','E'))
+        return AVERROR_INVALIDDATA;
+    if (AV_RL32(avctx->extradata + 28) != MKTAG('f','m','t',' '))
+        return AVERROR_INVALIDDATA;
+    if (AV_RL16(avctx->extradata + 38) != 1 &&
+        AV_RL16(avctx->extradata + 38) != 2)
+        return AVERROR_INVALIDDATA;
+
+    av_channel_layout_uninit(&avctx->ch_layout);
+    av_channel_layout_default(&avctx->ch_layout, AV_RL16(avctx->extradata + 38));
+    avctx->sample_rate = AV_RL32(avctx->extradata + 40);
+
+    switch (avctx->extradata[36]) {
+    case 0: avctx->sample_fmt = AV_SAMPLE_FMT_U8P;  break;
+    case 1: avctx->sample_fmt = AV_SAMPLE_FMT_S16P; break;
+    }
+
+    s->shift = 0;
+    switch (avctx->codec_tag) {
+    case MKTAG('1','D','I','F'):
+        s->nb_samples = 256;
+        s->offset = 4;
+        break;
+    case MKTAG('2','S','L','P'):
+    case MKTAG('3','N','L','P'):
+    case MKTAG('4','A','L','P'):
+        s->nb_samples = 570;
+        s->offset = 70;
+        break;
+    default:
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->max_framesize = s->nb_samples * 16;
+    s->bitstream = av_calloc(s->max_framesize, sizeof(*s->bitstream));
+    if (!s->bitstream)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static unsigned get_urice(GetBitContext *gb, int k)
+{
+    unsigned x = get_unary(gb, 1, get_bits_left(gb));
+    unsigned y = get_bits_long(gb, k);
+    unsigned z = (x << k) | y;
+
+    return z;
+}
+
+static int get_srice(GetBitContext *gb, int k)
+{
+    unsigned z = get_urice(gb, k);
+
+    return (z & 1) ? ~((int)(z >> 1)) : z >> 1;
+}
+
+static void do_stereo(WavArcContext *s, int ch, int correlated, int len)
+{
+    const int nb_samples = s->nb_samples;
+    const int shift = s->shift;
+
+    if (ch == 0) {
+        if (correlated) {
+            for (int n = 0; n < len; n++) {
+                s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
+                s->samples[1][n] = s->pred[1][n] >> shift;
+            }
+        } else {
+            for (int n = 0; n < len; n++) {
+                s->samples[0][n] = s->samples[0][nb_samples + n] >> shift;
+                s->samples[1][n] = s->pred[0][n] >> shift;
+            }
+        }
+    } else {
+        if (correlated) {
+            for (int n = 0; n < nb_samples; n++)
+                s->samples[1][n + len] += s->samples[0][n + len];
+        }
+        for (int n = 0; n < len; n++) {
+            s->pred[0][n] = s->samples[1][nb_samples + n];
+            s->pred[1][n] = s->pred[0][n] - s->samples[0][nb_samples + n];
+        }
+    }
+}
+
+static int decode_1dif(AVCodecContext *avctx,
+                       WavArcContext *s, GetBitContext *gb)
+{
+    int ch, finished, fill, correlated;
+
+    ch = 0;
+    finished = 0;
+    while (!finished) {
+        int *samples = s->samples[ch];
+        int k, block_type;
+
+        if (get_bits_left(gb) <= 0)
+            return AVERROR_INVALIDDATA;
+
+        block_type = get_urice(gb, 1);
+        if (block_type < 4 && block_type >= 0) {
+            k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
+            k = get_urice(gb, k) + 1;
+        }
+
+        switch (block_type) {
+        case 8:
+            s->eof = 1;
+            return AVERROR_EOF;
+        case 7:
+            s->nb_samples = get_bits(gb, 8);
+            continue;
+        case 6:
+            s->shift = get_urice(gb, 2);
+            continue;
+        case 5:
+            if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
+                fill = (int8_t)get_bits(gb, 8);
+                fill -= 0x80;
+            } else {
+                fill = (int16_t)get_bits(gb, 16);
+                fill -= 0x8000;
+            }
+
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = fill;
+            finished = 1;
+            break;
+        case 4:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = 0;
+            finished = 1;
+            break;
+        case 3:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = get_srice(gb, k) + (samples[n + 3] - samples[n + 2]) * 3 +
+                                          samples[n + 1];
+            finished = 1;
+            break;
+        case 2:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = get_srice(gb, k) + (samples[n + 3] * 2 - samples[n + 2]);
+            finished = 1;
+            break;
+        case 1:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = get_srice(gb, k) + samples[n + 3];
+            finished = 1;
+            break;
+        case 0:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 4] = get_srice(gb, k);
+            finished = 1;
+            break;
+        default:
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
+            if (ch == 0)
+                correlated = get_bits1(gb);
+            finished = ch != 0;
+            do_stereo(s, ch, correlated, 4);
+            ch = 1;
+        }
+    }
+
+    if (avctx->ch_layout.nb_channels == 1) {
+        for (int n = 0; n < 4; n++)
+            s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift;
+    }
+
+    return 0;
+}
+
+static int decode_2slp(AVCodecContext *avctx,
+                       WavArcContext *s, GetBitContext *gb)
+{
+    int ch, finished, fill, correlated, order;
+
+    ch = 0;
+    finished = 0;
+    while (!finished) {
+        int *samples = s->samples[ch];
+        int k, block_type;
+
+        if (get_bits_left(gb) <= 0)
+            return AVERROR_INVALIDDATA;
+
+        block_type = get_urice(gb, 1);
+        if (block_type < 5 && block_type >= 0) {
+            k = 1 + (avctx->sample_fmt == AV_SAMPLE_FMT_S16P);
+            k = get_urice(gb, k) + 1;
+        }
+
+        switch (block_type) {
+        case 9:
+            s->eof = 1;
+            return AVERROR_EOF;
+        case 8:
+            s->nb_samples = get_urice(gb, 8);
+            continue;
+        case 7:
+            s->shift = get_urice(gb, 2);
+            continue;
+        case 6:
+            if (avctx->sample_fmt == AV_SAMPLE_FMT_U8P) {
+                fill = (int8_t)get_bits(gb, 8);
+                fill -= 0x80;
+            } else {
+                fill = (int16_t)get_bits(gb, 16);
+                fill -= 0x8000;
+            }
+
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = fill;
+            finished = 1;
+            break;
+        case 5:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = 0;
+            finished = 1;
+            break;
+        case 4:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = get_srice(gb, k) + (samples[n + 69] - samples[n + 68]) * 3 +
+                                           samples[n + 67];
+            finished = 1;
+            break;
+        case 3:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = get_srice(gb, k) + (samples[n + 69] * 2 - samples[n + 68]);
+            finished = 1;
+            break;
+        case 2:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = get_srice(gb, k);
+            finished = 1;
+            break;
+        case 1:
+            for (int n = 0; n < s->nb_samples; n++)
+                samples[n + 70] = get_srice(gb, k) + samples[n + 69];
+            finished = 1;
+            break;
+        case 0:
+            order = get_urice(gb, 2);
+            for (int o = 0; o < order; o++)
+                s->filter[ch][o] = get_srice(gb, 2);
+            for (int n = 0; n < s->nb_samples; n++) {
+                int sum = 15;
+
+                for (int o = 0; o < order; o++)
+                    sum += s->filter[ch][o] * samples[n + 70 - o - 1];
+
+                samples[n + 70] = get_srice(gb, k) + (sum >> 4);
+            }
+            finished = 1;
+            break;
+        default:
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (finished == 1 && avctx->ch_layout.nb_channels == 2) {
+            if (ch == 0)
+                correlated = get_bits1(gb);
+            finished = ch != 0;
+            do_stereo(s, ch, correlated, 70);
+            ch = 1;
+        }
+    }
+
+    if (avctx->ch_layout.nb_channels == 1) {
+        for (int n = 0; n < 70; n++)
+            s->samples[0][n] = s->samples[0][s->nb_samples + n] >> s->shift;
+    }
+
+    return 0;
+}
+
+static int wavarc_decode(AVCodecContext *avctx, AVFrame *frame,
+                         int *got_frame_ptr, AVPacket *pkt)
+{
+    WavArcContext *s = avctx->priv_data;
+    GetBitContext *gb = &s->gb;
+    int buf_size, input_buf_size;
+    const uint8_t *buf;
+    int ret, n;
+
+    if ((!pkt->size && !s->bitstream_size) || s->nb_samples == 0 || s->eof) {
+        *got_frame_ptr = 0;
+        return pkt->size;
+    }
+
+    buf_size = FFMIN(pkt->size, s->max_framesize - s->bitstream_size);
+    input_buf_size = buf_size;
+    if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > s->max_framesize) {
+        memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
+        s->bitstream_index = 0;
+    }
+    if (pkt->data)
+        memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], pkt->data, buf_size);
+    buf                = &s->bitstream[s->bitstream_index];
+    buf_size          += s->bitstream_size;
+    s->bitstream_size  = buf_size;
+    if (buf_size < s->max_framesize && pkt->data) {
+        *got_frame_ptr = 0;
+        return input_buf_size;
+    }
+
+    if ((ret = init_get_bits8(gb, buf, buf_size)) < 0)
+        return ret;
+    skip_bits(gb, s->skip);
+
+    switch (avctx->codec_tag) {
+    case MKTAG('1','D','I','F'):
+        ret = decode_1dif(avctx, s, gb);
+        break;
+    case MKTAG('2','S','L','P'):
+    case MKTAG('3','N','L','P'):
+    case MKTAG('4','A','L','P'):
+        ret = decode_2slp(avctx, s, gb);
+        break;
+    default:
+        ret = AVERROR_INVALIDDATA;
+    }
+
+    if (ret < 0)
+        goto fail;
+
+    s->skip = get_bits_count(gb) - 8 * (get_bits_count(gb) / 8);
+    n = get_bits_count(gb) / 8;
+
+    if (n > buf_size) {
+fail:
+        s->bitstream_size = 0;
+        s->bitstream_index = 0;
+        return ret;
+    }
+
+    frame->nb_samples = s->nb_samples;
+    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+        return ret;
+
+    switch (avctx->sample_fmt) {
+    case AV_SAMPLE_FMT_U8P:
+        for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
+            uint8_t *dst = (uint8_t *)frame->extended_data[ch];
+            const int *src = s->samples[ch] + s->offset;
+
+            for (int n = 0; n < frame->nb_samples; n++)
+                dst[n] = src[n] * (1 << s->shift);
+        }
+        break;
+    case AV_SAMPLE_FMT_S16P:
+        for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
+            int16_t *dst = (int16_t *)frame->extended_data[ch];
+            const int *src = s->samples[ch] + s->offset;
+
+            for (int n = 0; n < frame->nb_samples; n++)
+                dst[n] = src[n] * (1 << s->shift);
+        }
+        break;
+    }
+
+    *got_frame_ptr = 1;
+
+    if (s->bitstream_size) {
+        s->bitstream_index += n;
+        s->bitstream_size  -= n;
+        return input_buf_size;
+    }
+
+    return n;
+}
+
+static av_cold int wavarc_close(AVCodecContext *avctx)
+{
+    WavArcContext *s = avctx->priv_data;
+
+    av_freep(&s->bitstream);
+    s->bitstream_size = 0;
+
+    return 0;
+}
+
+const FFCodec ff_wavarc_decoder = {
+    .p.name           = "wavarc",
+    CODEC_LONG_NAME("Waveform Archiver"),
+    .p.type           = AVMEDIA_TYPE_AUDIO,
+    .p.id             = AV_CODEC_ID_WAVARC,
+    .priv_data_size   = sizeof(WavArcContext),
+    .init             = wavarc_init,
+    FF_CODEC_DECODE_CB(wavarc_decode),
+    .close            = wavarc_close,
+    .p.capabilities   = AV_CODEC_CAP_DR1 |
+                        AV_CODEC_CAP_SUBFRAMES |
+                        AV_CODEC_CAP_DELAY,
+    .p.sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
+                                                        AV_SAMPLE_FMT_S16P,
+                                                        AV_SAMPLE_FMT_NONE },
+};