[FFmpeg-devel] [PATCH] Waveform Archiver decoder and demuxer

* [FFmpeg-devel] [PATCH] Waveform Archiver decoder and demuxer
@ 2023-01-29 12:00 Paul B Mahol
  2023-01-31 17:11 ` Paul B Mahol
                   ` (2 more replies)
  0 siblings, 3 replies; 8+ messages in thread
From: Paul B Mahol @ 2023-01-29 12:00 UTC (permalink / raw)
  To: FFmpeg development discussions and patches

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Patches attached

[-- Attachment #2: 0001-avcodec-add-WavArc-decoder.patch --]
[-- Type: text/x-patch, Size: 17172 bytes --]

From 5516601630ae83098e2a3d27823f21e86bc0e60d Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sat, 21 Jan 2023 19:25:41 +0100
Subject: [PATCH 1/2] avcodec: add WavArc decoder

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/wavarc.c     | 460 ++++++++++++++++++++++++++++++++++++++++
 5 files changed, 470 insertions(+)
 create mode 100644 libavcodec/wavarc.c

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index f0ffd0b961..e936c19f33 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -777,6 +777,7 @@ 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 f30047e17a..0d6998786c 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 50f9794e10..6ecdf1002b 100644
--- a/libavcodec/codec_desc.c
+++ b/libavcodec/codec_desc.c
@@ -3346,6 +3346,13 @@ 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 cdf7eb79c3..ba3b432387 100644
--- a/libavcodec/codec_id.h
+++ b/libavcodec/codec_id.h
@@ -535,6 +535,7 @@ 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/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 },
+};
-- 
2.39.1


[-- Attachment #3: 0002-avformat-add-WavArc-demuxer.patch --]
[-- Type: text/x-patch, Size: 5889 bytes --]

From b54f2ba1e206bff2ef5d9e8882cbdce4a5fbccc4 Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sat, 21 Jan 2023 19:17:38 +0100
Subject: [PATCH 2/2] avformat: add WavArc demuxer

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 libavformat/Makefile     |   1 +
 libavformat/allformats.c |   1 +
 libavformat/wavarc.c     | 127 +++++++++++++++++++++++++++++++++++++++
 3 files changed, 129 insertions(+)
 create mode 100644 libavformat/wavarc.c

diff --git a/libavformat/Makefile b/libavformat/Makefile
index 2d11bcd7a3..9052b023a7 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -598,6 +598,7 @@ OBJS-$(CONFIG_VQF_DEMUXER)               += vqf.o
 OBJS-$(CONFIG_W64_DEMUXER)               += wavdec.o w64.o pcm.o
 OBJS-$(CONFIG_W64_MUXER)                 += wavenc.o w64.o
 OBJS-$(CONFIG_WADY_DEMUXER)              += wady.o pcm.o
+OBJS-$(CONFIG_WAVARC_DEMUXER)            += wavarc.o rawdec.o
 OBJS-$(CONFIG_WAV_DEMUXER)               += wavdec.o pcm.o
 OBJS-$(CONFIG_WAV_MUXER)                 += wavenc.o
 OBJS-$(CONFIG_WC3_DEMUXER)               += wc3movie.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index bf8afe2078..7c01c7f098 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -480,6 +480,7 @@ extern const AVInputFormat  ff_vqf_demuxer;
 extern const AVInputFormat  ff_w64_demuxer;
 extern const AVOutputFormat ff_w64_muxer;
 extern const AVInputFormat  ff_wady_demuxer;
+extern const AVInputFormat  ff_wavarc_demuxer;
 extern const AVInputFormat  ff_wav_demuxer;
 extern const AVOutputFormat ff_wav_muxer;
 extern const AVInputFormat  ff_wc3_demuxer;
diff --git a/libavformat/wavarc.c b/libavformat/wavarc.c
new file mode 100644
index 0000000000..a475c85f8e
--- /dev/null
+++ b/libavformat/wavarc.c
@@ -0,0 +1,127 @@
+/*
+ * WavArc demuxer
+ * 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/channel_layout.h"
+#include "libavutil/intreadwrite.h"
+#include "avformat.h"
+#include "demux.h"
+#include "internal.h"
+#include "rawdec.h"
+
+static int wavarc_probe(const AVProbeData *p)
+{
+    int len = p->buf[0];
+    uint32_t id;
+
+    if (len == 0 || len + 6 >= p->buf_size)
+        return 0;
+
+    if (p->buf[len + 1] != 0)
+        return 0;
+
+    id = AV_RL32(p->buf + len + 2);
+    if (id != MKTAG('1','D','I','F') &&
+        id != MKTAG('2','S','L','P') &&
+        id != MKTAG('3','N','L','P') &&
+        id != MKTAG('4','A','L','P') &&
+        id != MKTAG('5','E','L','P'))
+        return 0;
+
+    return AVPROBE_SCORE_MAX / 3 * 2;
+}
+
+static int wavarc_read_header(AVFormatContext *s)
+{
+    AVIOContext *pb = s->pb;
+    AVCodecParameters *par;
+    int filename_len, fmt_len, ret;
+    uint8_t data[36];
+    AVStream *st;
+    uint32_t id;
+
+    filename_len = avio_r8(pb);
+    if (filename_len == 0)
+        return AVERROR_INVALIDDATA;
+    avio_skip(pb, filename_len);
+    if (avio_r8(pb))
+        return AVERROR_INVALIDDATA;
+    id = avio_rl32(pb);
+    if (avio_read(pb, data, sizeof(data)) != sizeof(data))
+        return AVERROR(EIO);
+    fmt_len = AV_RL32(data + 32);
+    if (fmt_len < 12)
+        return AVERROR_INVALIDDATA;
+
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+    par = st->codecpar;
+
+    ret = ff_alloc_extradata(par, fmt_len + sizeof(data));
+    if (ret < 0)
+        return ret;
+    memcpy(par->extradata, data, sizeof(data));
+    avio_read(pb, par->extradata + sizeof(data), fmt_len);
+
+    par->codec_type = AVMEDIA_TYPE_AUDIO;
+    par->codec_id   = AV_CODEC_ID_WAVARC;
+    par->codec_tag  = id;
+
+    do {
+        id = avio_rl32(pb);
+        if (id != MKTAG('d','a','t','a'))
+            avio_skip(pb, avio_rl32(pb));
+    } while (id != MKTAG('d','a','t','a'));
+    avio_skip(pb, 4);
+
+    if (AV_RL32(par->extradata + 16) != MKTAG('R','I','F','F'))
+        return AVERROR_INVALIDDATA;
+    if (AV_RL32(par->extradata + 24) != MKTAG('W','A','V','E'))
+        return AVERROR_INVALIDDATA;
+    if (AV_RL32(par->extradata + 28) != MKTAG('f','m','t',' '))
+        return AVERROR_INVALIDDATA;
+
+    av_channel_layout_uninit(&par->ch_layout);
+    av_channel_layout_default(&par->ch_layout, AV_RL16(par->extradata + 38));
+    par->sample_rate = AV_RL32(par->extradata + 40);
+    avpriv_set_pts_info(st, 64, 1, par->sample_rate);
+    st->start_time = 0;
+
+    switch (par->extradata[36]) {
+    case 0: par->format = AV_SAMPLE_FMT_U8P;  break;
+    case 1: par->format = AV_SAMPLE_FMT_S16P; break;
+    }
+
+    return 0;
+}
+
+const AVInputFormat ff_wavarc_demuxer = {
+    .name           = "wavarc",
+    .long_name      = NULL_IF_CONFIG_SMALL("Waveform Archiver"),
+    .read_probe     = wavarc_probe,
+    .read_packet    = ff_raw_read_partial_packet,
+    .flags          = AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK | AVFMT_NOTIMESTAMPS,
+    .read_header    = wavarc_read_header,
+    .extensions     = "wa",
+    .raw_codec_id   = AV_CODEC_ID_WAVARC,
+    .priv_data_size = sizeof(FFRawDemuxerContext),
+    .priv_class     = &ff_raw_demuxer_class,
+};
-- 
2.39.1


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