diff mbox series

[FFmpeg-devel,v7,2/2] avformat: add demuxer for Pro Pinball Series' Soundbanks

Message ID 20200406145233.25651-3-zane@zanevaniperen.com
State Superseded
Headers show
Series Pro Pinball Series Soundbank demuxer + decoder. | expand

Checks

Context Check Description
andriy/ffmpeg-patchwork success Make fate finished

Commit Message

Zane van Iperen April 6, 2020, 2:53 p.m. UTC
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
---
 Changelog                |   1 +
 libavformat/Makefile     |   1 +
 libavformat/allformats.c |   1 +
 libavformat/pp_bnk.c     | 271 +++++++++++++++++++++++++++++++++++++++
 libavformat/version.h    |   2 +-
 5 files changed, 275 insertions(+), 1 deletion(-)
 create mode 100644 libavformat/pp_bnk.c

Comments

Andreas Rheinhardt April 6, 2020, 3:39 p.m. UTC | #1
Zane van Iperen:
> Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
> ---
>  Changelog                |   1 +
>  libavformat/Makefile     |   1 +
>  libavformat/allformats.c |   1 +
>  libavformat/pp_bnk.c     | 271 +++++++++++++++++++++++++++++++++++++++
>  libavformat/version.h    |   2 +-
>  5 files changed, 275 insertions(+), 1 deletion(-)
>  create mode 100644 libavformat/pp_bnk.c
> 
> diff --git a/Changelog b/Changelog
> index f1be5d674e..d0c62000d6 100644
> --- a/Changelog
> +++ b/Changelog
> @@ -57,6 +57,7 @@ version <next>:
>  - overlay_cuda filter
>  - switch from AvxSynth to AviSynth+ on Linux
>  - Cunning Developments ADPCM decoder
> +- Pro Pinball Series Soundbank demuxer
>  
>  
>  version 4.2:
> diff --git a/libavformat/Makefile b/libavformat/Makefile
> index 8fd0d43721..9df99133fa 100644
> --- a/libavformat/Makefile
> +++ b/libavformat/Makefile
> @@ -428,6 +428,7 @@ OBJS-$(CONFIG_PCM_VIDC_DEMUXER)          += pcmdec.o pcm.o
>  OBJS-$(CONFIG_PCM_VIDC_MUXER)            += pcmenc.o rawenc.o
>  OBJS-$(CONFIG_PJS_DEMUXER)               += pjsdec.o subtitles.o
>  OBJS-$(CONFIG_PMP_DEMUXER)               += pmpdec.o
> +OBJS-$(CONFIG_PP_BNK_DEMUXER)            += pp_bnk.o
>  OBJS-$(CONFIG_PVA_DEMUXER)               += pva.o
>  OBJS-$(CONFIG_PVF_DEMUXER)               += pvfdec.o pcm.o
>  OBJS-$(CONFIG_QCP_DEMUXER)               += qcp.o
> diff --git a/libavformat/allformats.c b/libavformat/allformats.c
> index 39d2c352f5..3919c9e4c1 100644
> --- a/libavformat/allformats.c
> +++ b/libavformat/allformats.c
> @@ -341,6 +341,7 @@ extern AVInputFormat  ff_pcm_u8_demuxer;
>  extern AVOutputFormat ff_pcm_u8_muxer;
>  extern AVInputFormat  ff_pjs_demuxer;
>  extern AVInputFormat  ff_pmp_demuxer;
> +extern AVInputFormat  ff_pp_bnk_demuxer;
>  extern AVOutputFormat ff_psp_muxer;
>  extern AVInputFormat  ff_pva_demuxer;
>  extern AVInputFormat  ff_pvf_demuxer;
> diff --git a/libavformat/pp_bnk.c b/libavformat/pp_bnk.c
> new file mode 100644
> index 0000000000..fca0f75666
> --- /dev/null
> +++ b/libavformat/pp_bnk.c
> @@ -0,0 +1,271 @@
> +/*
> + * Pro Pinball Series Soundbank (44c, 22c, 11c, 5c) demuxer.
> + *
> + * Copyright (C) 2020 Zane van Iperen (zane@zanevaniperen.com)
> + *
> + * 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 "avformat.h"
> +#include "internal.h"
> +#include "libavutil/intreadwrite.h"
> +#include "libavutil/avassert.h"
> +#include "libavutil/internal.h"
> +
> +#define PP_BNK_MAX_READ_SIZE    4096
> +#define PP_BNK_FILE_HEADER_SIZE 20
> +#define PP_BNK_TRACK_SIZE       20
> +
> +typedef struct PPBnkHeader {
> +    uint32_t        bank_id;        /*< Bank ID, useless for our purposes. */
> +    uint32_t        sample_rate;    /*< Sample rate of the contained tracks. */
> +    uint32_t        always1;        /*< Unknown, always seems to be 1. */
> +    uint32_t        track_count;    /*< Number of tracks in the file. */
> +    uint32_t        flags;          /*< Flags. */
> +} PPBnkHeader;
> +
> +typedef struct PPBnkTrack {
> +    uint32_t        id;             /*< Track ID. Usually track[i].id == track[i-1].id + 1, but not always */
> +    uint32_t        size;           /*< Size of the data in bytes. */
> +    uint32_t        sample_rate;    /*< Sample rate. */
> +    uint32_t        always1_1;      /*< Unknown, always seems to be 1. */
> +    uint32_t        always1_2;      /*< Unknown, always seems to be 1. */
> +} PPBnkTrack;
> +
> +typedef struct PPBnkCtxTrack {
> +    int64_t         data_offset;
> +    uint32_t        data_size;
> +} PPBnkCtxTrack;
> +
> +typedef struct PPBnkCtx {
> +    int             track_count;
> +    PPBnkCtxTrack   *tracks;
> +    uint32_t        current_track;
> +    uint32_t        bytes_read;
> +} PPBnkCtx;
> +
> +enum {
> +    PP_BNK_FLAG_PERSIST = (1 << 0), /*< This is a large file, keep in memory. */
> +    PP_BNK_FLAG_MUSIC   = (1 << 1), /*< This is music. */
> +    PP_BNK_FLAG_MASK    = (PP_BNK_FLAG_PERSIST | PP_BNK_FLAG_MUSIC)
> +};
> +
> +static void pp_bnk_parse_header(PPBnkHeader *hdr, const uint8_t *buf)
> +{
> +    hdr->bank_id        = AV_RL32(buf +  0);
> +    hdr->sample_rate    = AV_RL32(buf +  4);
> +    hdr->always1        = AV_RL32(buf +  8);
> +    hdr->track_count    = AV_RL32(buf + 12);
> +    hdr->flags          = AV_RL32(buf + 16);
> +}
> +
> +static void pp_bnk_parse_track(PPBnkTrack *trk, const uint8_t *buf)
> +{
> +    trk->id             = AV_RL32(buf +  0);
> +    trk->size           = AV_RL32(buf +  4);
> +    trk->sample_rate    = AV_RL32(buf +  8);
> +    trk->always1_1      = AV_RL32(buf + 12);
> +    trk->always1_2      = AV_RL32(buf + 16);
> +}
> +
> +static int pp_bnk_probe(const AVProbeData *p)
> +{
> +    uint32_t sample_rate = AV_RL32(p->buf +  4);
> +    uint32_t track_count = AV_RL32(p->buf + 12);
> +    uint32_t flags       = AV_RL32(p->buf + 16);
> +
> +    if (track_count == 0 || sample_rate == 0)
> +        return 0;
> +
> +    /* Sometimes we have the first track header, so check that too. */
> +    if (p->buf_size >= 32 && AV_RL32(p->buf + 28) != sample_rate)
> +        return 0;
> +
> +    /* These limits are based on analysing the game files. */
> +    if (track_count > 113)
> +        return 10;
> +
> +    if ((sample_rate !=  5512) && (sample_rate != 11025) &&
> +        (sample_rate != 22050) && (sample_rate != 44100))
> +        return 10;
> +
> +    if ((flags & ~PP_BNK_FLAG_MASK) != 0)
> +        return 10;
> +
> +    return AVPROBE_SCORE_MAX / 4 + 1;
> +}
> +
> +static int pp_bnk_read_header(AVFormatContext *s)
> +{
> +    int ret;
> +    AVStream *st;
> +    AVCodecParameters *par;
> +    PPBnkCtx *ctx = s->priv_data;
> +    uint8_t buf[FFMAX(PP_BNK_FILE_HEADER_SIZE, PP_BNK_TRACK_SIZE)];
> +    PPBnkHeader hdr;
> +
> +    if ((ret = avio_read(s->pb, buf, PP_BNK_FILE_HEADER_SIZE)) < 0)
> +        return ret;
> +    else if (ret != PP_BNK_FILE_HEADER_SIZE)
> +        return AVERROR(EIO);
> +
> +    pp_bnk_parse_header(&hdr, buf);
> +
> +    if (hdr.track_count == 0 || hdr.track_count > INT_MAX)
> +        return AVERROR_INVALIDDATA;
> +
> +    if (hdr.sample_rate == 0 || hdr.sample_rate > INT_MAX)
> +        return AVERROR_INVALIDDATA;
> +
> +    if (hdr.always1 != 1) {
> +        avpriv_request_sample(s, "Non-one header value");
> +        return AVERROR_PATCHWELCOME;
> +    }
> +
> +    ctx->track_count = hdr.track_count;
> +
> +    if (!(ctx->tracks = av_malloc_array(hdr.track_count, sizeof(PPBnkCtxTrack))))
> +        return AVERROR(ENOMEM);
> +
> +    /* Parse and validate each track. */
> +    for (int i = 0; i < hdr.track_count; i++) {
> +        PPBnkTrack e;
> +
> +        if ((ret = avio_read(s->pb, buf, PP_BNK_TRACK_SIZE)) < 0) {
> +            goto done;
> +        } else if (ret != PP_BNK_TRACK_SIZE) {
> +            ret = AVERROR(EIO);
> +            goto done;
> +        }

If your file is truncated and missing the data as well as the header of
one of the latter tracks, you will ignore everything in the file even if
the earlier tracks are completely there. Instead you could reset
track_count to the number of tracks for which data is available and only
error out if that number is zero.

> +
> +        pp_bnk_parse_track(&e, buf);
> +
> +        /* The individual sample rates of all tracks must match that of the file header. */
> +        if (e.sample_rate != hdr.sample_rate) {
> +            ret = AVERROR_INVALIDDATA;
> +            goto done;
> +        }
> +
> +        if (e.always1_1 != 1 || e.always1_2 != 1) {
> +            avpriv_request_sample(s, "Non-one track header values");
> +            ret = AVERROR_PATCHWELCOME;
> +            goto done;
> +        }
> +
> +        ctx->tracks[i].data_offset = avio_tell(s->pb);
> +        ctx->tracks[i].data_size   = e.size;
> +
> +        /* Skip over the data to the next stream header. */
> +        avio_skip(s->pb, e.size);
> +    }
> +
> +    /* Build the streams. */
> +    for (int i = 0; i < hdr.track_count; i++) {
> +        if (!(st = avformat_new_stream(s, NULL))) {
> +            ret = AVERROR(ENOMEM);
> +            goto done;
> +        }
> +
> +        par                         = st->codecpar;
> +        par->codec_type             = AVMEDIA_TYPE_AUDIO;
> +        par->codec_id               = AV_CODEC_ID_ADPCM_IMA_CUNNING;
> +        par->format                 = AV_SAMPLE_FMT_S16;
> +        par->channel_layout         = AV_CH_LAYOUT_MONO;
> +        par->channels               = 1;
> +        par->sample_rate            = hdr.sample_rate;
> +        par->bits_per_coded_sample  = 4;
> +        par->bits_per_raw_sample    = 16;
> +        par->block_align            = 1;
> +        par->bit_rate               = par->sample_rate * par->bits_per_coded_sample;
> +
> +        avpriv_set_pts_info(st, 64, 1, par->sample_rate);
> +        st->start_time              = 0;
> +        st->duration                = ctx->tracks[i].data_size * 2;
> +    }
> +
> +    /* Seek to the start of the first stream. */
> +    if ((ret = avio_seek(s->pb, ctx->tracks[0].data_offset, SEEK_SET)) < 0) {
> +        goto done;
> +    } else if (ret != ctx->tracks[0].data_offset) {
> +        ret = AVERROR(EIO);
> +        goto done;
> +    }
> +
> +    ret = 0;
> +
You could just return 0 here and convert your done into a fail. This
would make the intent clearer.

> +done:
> +    if (ret != 0)
> +        av_freep(&ctx->tracks);
> +
> +    return ret;
> +}
> +
> +static int pp_bnk_read_packet(AVFormatContext *s, AVPacket *pkt)
> +{
> +    int64_t ret;
> +    int size;
> +    PPBnkCtx *ctx = s->priv_data;
> +    PPBnkCtxTrack *trk = ctx->tracks + ctx->current_track;
> +
> +    av_assert0(ctx->bytes_read <= trk->data_size);
> +
> +    if (ctx->bytes_read == trk->data_size) {
> +        if (ctx->current_track == ctx->track_count - 1)
> +            return AVERROR_EOF;
> +
> +        trk++;
> +
> +        if ((ret = avio_seek(s->pb, trk->data_offset, SEEK_SET)) < 0)
> +            return ret;
> +        else if (ret != trk->data_offset)
> +            return AVERROR(EIO);
> +
> +        ctx->bytes_read = 0;
> +        ctx->current_track++;
> +    }
> +
> +    size = FFMIN(trk->data_size - ctx->bytes_read, PP_BNK_MAX_READ_SIZE);
> +
> +    if ((ret = av_get_packet(s->pb, pkt, size)) < 0)
> +        return ret;
> +    else if (ret != size)
> +        return AVERROR_INVALIDDATA;
> +
> +    ctx->bytes_read    += ret;

If you have read something, but not enough, your internal counter will
be out of sync with the file. This doesn't seem to be a problem, so I
don't know if something should be done about it.
> +
> +    pkt->stream_index   = ctx->current_track;
> +    pkt->duration       = ret * 2;
> +    return 0;
> +}
> +
> +static int pp_bnk_read_close(AVFormatContext *s)
> +{
> +    PPBnkCtx *ctx = s->priv_data;
> +
> +    av_freep(&ctx->tracks);
> +
> +    return 0;
> +}
> +
> +AVInputFormat ff_pp_bnk_demuxer = {
> +    .name           = "pp_bnk",
> +    .long_name      = NULL_IF_CONFIG_SMALL("Pro Pinball Series Soundbank"),
> +    .priv_data_size = sizeof(PPBnkCtx),
> +    .read_probe     = pp_bnk_probe,
> +    .read_header    = pp_bnk_read_header,
> +    .read_packet    = pp_bnk_read_packet,
> +    .read_close     = pp_bnk_read_close
> +};
> diff --git a/libavformat/version.h b/libavformat/version.h
> index 18c2f5fec2..493a0b337f 100644
> --- a/libavformat/version.h
> +++ b/libavformat/version.h
> @@ -32,7 +32,7 @@
>  // Major bumping may affect Ticket5467, 5421, 5451(compatibility with Chromium)
>  // Also please add any ticket numbers that you believe might be affected here
>  #define LIBAVFORMAT_VERSION_MAJOR  58
> -#define LIBAVFORMAT_VERSION_MINOR  42
> +#define LIBAVFORMAT_VERSION_MINOR  43
>  #define LIBAVFORMAT_VERSION_MICRO 100
>  
>  #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
>
Zane van Iperen April 7, 2020, 9:59 a.m. UTC | #2
On Mon, 6 Apr 2020 17:39:13 +0200
"Andreas Rheinhardt" <andreas.rheinhardt@gmail.com> wrote:

> > +    /* Parse and validate each track. */
> > +    for (int i = 0; i < hdr.track_count; i++) {
> > +        PPBnkTrack e;
> > +
> > +        if ((ret = avio_read(s->pb, buf, PP_BNK_TRACK_SIZE)) < 0) {
> > +            goto done;
> > +        } else if (ret != PP_BNK_TRACK_SIZE) {
> > +            ret = AVERROR(EIO);
> > +            goto done;
> > +        }  
> 
> If your file is truncated and missing the data as well as the header
> of one of the latter tracks, you will ignore everything in the file
> even if the earlier tracks are completely there. Instead you could
> reset track_count to the number of tracks for which data is available
> and only error out if that number is zero.

Have done this.


> > +
> > +    ret = 0;
> > +  
> You could just return 0 here and convert your done into a fail. This
> would make the intent clearer.


Done.

> > +    size = FFMIN(trk->data_size - ctx->bytes_read,
> > PP_BNK_MAX_READ_SIZE); +
> > +    if ((ret = av_get_packet(s->pb, pkt, size)) < 0)
> > +        return ret;
> > +    else if (ret != size)
> > +        return AVERROR_INVALIDDATA;
> > +
> > +    ctx->bytes_read    += ret;  
> 
> If you have read something, but not enough, your internal counter will
> be out of sync with the file. This doesn't seem to be a problem, so I
> don't know if something should be done about it.

Fixed it anyway, changed to:

    if ((ret = av_get_packet(s->pb, pkt, size)) < 0)
        return ret;

    ctx->bytes_read    += ret;
    pkt->flags         &= ~AV_PKT_FLAG_CORRUPT;
    pkt->stream_index   = ctx->current_track;
    pkt->duration       = ret * 2;

Zane
diff mbox series

Patch

diff --git a/Changelog b/Changelog
index f1be5d674e..d0c62000d6 100644
--- a/Changelog
+++ b/Changelog
@@ -57,6 +57,7 @@  version <next>:
 - overlay_cuda filter
 - switch from AvxSynth to AviSynth+ on Linux
 - Cunning Developments ADPCM decoder
+- Pro Pinball Series Soundbank demuxer
 
 
 version 4.2:
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 8fd0d43721..9df99133fa 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -428,6 +428,7 @@  OBJS-$(CONFIG_PCM_VIDC_DEMUXER)          += pcmdec.o pcm.o
 OBJS-$(CONFIG_PCM_VIDC_MUXER)            += pcmenc.o rawenc.o
 OBJS-$(CONFIG_PJS_DEMUXER)               += pjsdec.o subtitles.o
 OBJS-$(CONFIG_PMP_DEMUXER)               += pmpdec.o
+OBJS-$(CONFIG_PP_BNK_DEMUXER)            += pp_bnk.o
 OBJS-$(CONFIG_PVA_DEMUXER)               += pva.o
 OBJS-$(CONFIG_PVF_DEMUXER)               += pvfdec.o pcm.o
 OBJS-$(CONFIG_QCP_DEMUXER)               += qcp.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index 39d2c352f5..3919c9e4c1 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -341,6 +341,7 @@  extern AVInputFormat  ff_pcm_u8_demuxer;
 extern AVOutputFormat ff_pcm_u8_muxer;
 extern AVInputFormat  ff_pjs_demuxer;
 extern AVInputFormat  ff_pmp_demuxer;
+extern AVInputFormat  ff_pp_bnk_demuxer;
 extern AVOutputFormat ff_psp_muxer;
 extern AVInputFormat  ff_pva_demuxer;
 extern AVInputFormat  ff_pvf_demuxer;
diff --git a/libavformat/pp_bnk.c b/libavformat/pp_bnk.c
new file mode 100644
index 0000000000..fca0f75666
--- /dev/null
+++ b/libavformat/pp_bnk.c
@@ -0,0 +1,271 @@ 
+/*
+ * Pro Pinball Series Soundbank (44c, 22c, 11c, 5c) demuxer.
+ *
+ * Copyright (C) 2020 Zane van Iperen (zane@zanevaniperen.com)
+ *
+ * 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 "avformat.h"
+#include "internal.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/avassert.h"
+#include "libavutil/internal.h"
+
+#define PP_BNK_MAX_READ_SIZE    4096
+#define PP_BNK_FILE_HEADER_SIZE 20
+#define PP_BNK_TRACK_SIZE       20
+
+typedef struct PPBnkHeader {
+    uint32_t        bank_id;        /*< Bank ID, useless for our purposes. */
+    uint32_t        sample_rate;    /*< Sample rate of the contained tracks. */
+    uint32_t        always1;        /*< Unknown, always seems to be 1. */
+    uint32_t        track_count;    /*< Number of tracks in the file. */
+    uint32_t        flags;          /*< Flags. */
+} PPBnkHeader;
+
+typedef struct PPBnkTrack {
+    uint32_t        id;             /*< Track ID. Usually track[i].id == track[i-1].id + 1, but not always */
+    uint32_t        size;           /*< Size of the data in bytes. */
+    uint32_t        sample_rate;    /*< Sample rate. */
+    uint32_t        always1_1;      /*< Unknown, always seems to be 1. */
+    uint32_t        always1_2;      /*< Unknown, always seems to be 1. */
+} PPBnkTrack;
+
+typedef struct PPBnkCtxTrack {
+    int64_t         data_offset;
+    uint32_t        data_size;
+} PPBnkCtxTrack;
+
+typedef struct PPBnkCtx {
+    int             track_count;
+    PPBnkCtxTrack   *tracks;
+    uint32_t        current_track;
+    uint32_t        bytes_read;
+} PPBnkCtx;
+
+enum {
+    PP_BNK_FLAG_PERSIST = (1 << 0), /*< This is a large file, keep in memory. */
+    PP_BNK_FLAG_MUSIC   = (1 << 1), /*< This is music. */
+    PP_BNK_FLAG_MASK    = (PP_BNK_FLAG_PERSIST | PP_BNK_FLAG_MUSIC)
+};
+
+static void pp_bnk_parse_header(PPBnkHeader *hdr, const uint8_t *buf)
+{
+    hdr->bank_id        = AV_RL32(buf +  0);
+    hdr->sample_rate    = AV_RL32(buf +  4);
+    hdr->always1        = AV_RL32(buf +  8);
+    hdr->track_count    = AV_RL32(buf + 12);
+    hdr->flags          = AV_RL32(buf + 16);
+}
+
+static void pp_bnk_parse_track(PPBnkTrack *trk, const uint8_t *buf)
+{
+    trk->id             = AV_RL32(buf +  0);
+    trk->size           = AV_RL32(buf +  4);
+    trk->sample_rate    = AV_RL32(buf +  8);
+    trk->always1_1      = AV_RL32(buf + 12);
+    trk->always1_2      = AV_RL32(buf + 16);
+}
+
+static int pp_bnk_probe(const AVProbeData *p)
+{
+    uint32_t sample_rate = AV_RL32(p->buf +  4);
+    uint32_t track_count = AV_RL32(p->buf + 12);
+    uint32_t flags       = AV_RL32(p->buf + 16);
+
+    if (track_count == 0 || sample_rate == 0)
+        return 0;
+
+    /* Sometimes we have the first track header, so check that too. */
+    if (p->buf_size >= 32 && AV_RL32(p->buf + 28) != sample_rate)
+        return 0;
+
+    /* These limits are based on analysing the game files. */
+    if (track_count > 113)
+        return 10;
+
+    if ((sample_rate !=  5512) && (sample_rate != 11025) &&
+        (sample_rate != 22050) && (sample_rate != 44100))
+        return 10;
+
+    if ((flags & ~PP_BNK_FLAG_MASK) != 0)
+        return 10;
+
+    return AVPROBE_SCORE_MAX / 4 + 1;
+}
+
+static int pp_bnk_read_header(AVFormatContext *s)
+{
+    int ret;
+    AVStream *st;
+    AVCodecParameters *par;
+    PPBnkCtx *ctx = s->priv_data;
+    uint8_t buf[FFMAX(PP_BNK_FILE_HEADER_SIZE, PP_BNK_TRACK_SIZE)];
+    PPBnkHeader hdr;
+
+    if ((ret = avio_read(s->pb, buf, PP_BNK_FILE_HEADER_SIZE)) < 0)
+        return ret;
+    else if (ret != PP_BNK_FILE_HEADER_SIZE)
+        return AVERROR(EIO);
+
+    pp_bnk_parse_header(&hdr, buf);
+
+    if (hdr.track_count == 0 || hdr.track_count > INT_MAX)
+        return AVERROR_INVALIDDATA;
+
+    if (hdr.sample_rate == 0 || hdr.sample_rate > INT_MAX)
+        return AVERROR_INVALIDDATA;
+
+    if (hdr.always1 != 1) {
+        avpriv_request_sample(s, "Non-one header value");
+        return AVERROR_PATCHWELCOME;
+    }
+
+    ctx->track_count = hdr.track_count;
+
+    if (!(ctx->tracks = av_malloc_array(hdr.track_count, sizeof(PPBnkCtxTrack))))
+        return AVERROR(ENOMEM);
+
+    /* Parse and validate each track. */
+    for (int i = 0; i < hdr.track_count; i++) {
+        PPBnkTrack e;
+
+        if ((ret = avio_read(s->pb, buf, PP_BNK_TRACK_SIZE)) < 0) {
+            goto done;
+        } else if (ret != PP_BNK_TRACK_SIZE) {
+            ret = AVERROR(EIO);
+            goto done;
+        }
+
+        pp_bnk_parse_track(&e, buf);
+
+        /* The individual sample rates of all tracks must match that of the file header. */
+        if (e.sample_rate != hdr.sample_rate) {
+            ret = AVERROR_INVALIDDATA;
+            goto done;
+        }
+
+        if (e.always1_1 != 1 || e.always1_2 != 1) {
+            avpriv_request_sample(s, "Non-one track header values");
+            ret = AVERROR_PATCHWELCOME;
+            goto done;
+        }
+
+        ctx->tracks[i].data_offset = avio_tell(s->pb);
+        ctx->tracks[i].data_size   = e.size;
+
+        /* Skip over the data to the next stream header. */
+        avio_skip(s->pb, e.size);
+    }
+
+    /* Build the streams. */
+    for (int i = 0; i < hdr.track_count; i++) {
+        if (!(st = avformat_new_stream(s, NULL))) {
+            ret = AVERROR(ENOMEM);
+            goto done;
+        }
+
+        par                         = st->codecpar;
+        par->codec_type             = AVMEDIA_TYPE_AUDIO;
+        par->codec_id               = AV_CODEC_ID_ADPCM_IMA_CUNNING;
+        par->format                 = AV_SAMPLE_FMT_S16;
+        par->channel_layout         = AV_CH_LAYOUT_MONO;
+        par->channels               = 1;
+        par->sample_rate            = hdr.sample_rate;
+        par->bits_per_coded_sample  = 4;
+        par->bits_per_raw_sample    = 16;
+        par->block_align            = 1;
+        par->bit_rate               = par->sample_rate * par->bits_per_coded_sample;
+
+        avpriv_set_pts_info(st, 64, 1, par->sample_rate);
+        st->start_time              = 0;
+        st->duration                = ctx->tracks[i].data_size * 2;
+    }
+
+    /* Seek to the start of the first stream. */
+    if ((ret = avio_seek(s->pb, ctx->tracks[0].data_offset, SEEK_SET)) < 0) {
+        goto done;
+    } else if (ret != ctx->tracks[0].data_offset) {
+        ret = AVERROR(EIO);
+        goto done;
+    }
+
+    ret = 0;
+
+done:
+    if (ret != 0)
+        av_freep(&ctx->tracks);
+
+    return ret;
+}
+
+static int pp_bnk_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    int64_t ret;
+    int size;
+    PPBnkCtx *ctx = s->priv_data;
+    PPBnkCtxTrack *trk = ctx->tracks + ctx->current_track;
+
+    av_assert0(ctx->bytes_read <= trk->data_size);
+
+    if (ctx->bytes_read == trk->data_size) {
+        if (ctx->current_track == ctx->track_count - 1)
+            return AVERROR_EOF;
+
+        trk++;
+
+        if ((ret = avio_seek(s->pb, trk->data_offset, SEEK_SET)) < 0)
+            return ret;
+        else if (ret != trk->data_offset)
+            return AVERROR(EIO);
+
+        ctx->bytes_read = 0;
+        ctx->current_track++;
+    }
+
+    size = FFMIN(trk->data_size - ctx->bytes_read, PP_BNK_MAX_READ_SIZE);
+
+    if ((ret = av_get_packet(s->pb, pkt, size)) < 0)
+        return ret;
+    else if (ret != size)
+        return AVERROR_INVALIDDATA;
+
+    ctx->bytes_read    += ret;
+
+    pkt->stream_index   = ctx->current_track;
+    pkt->duration       = ret * 2;
+    return 0;
+}
+
+static int pp_bnk_read_close(AVFormatContext *s)
+{
+    PPBnkCtx *ctx = s->priv_data;
+
+    av_freep(&ctx->tracks);
+
+    return 0;
+}
+
+AVInputFormat ff_pp_bnk_demuxer = {
+    .name           = "pp_bnk",
+    .long_name      = NULL_IF_CONFIG_SMALL("Pro Pinball Series Soundbank"),
+    .priv_data_size = sizeof(PPBnkCtx),
+    .read_probe     = pp_bnk_probe,
+    .read_header    = pp_bnk_read_header,
+    .read_packet    = pp_bnk_read_packet,
+    .read_close     = pp_bnk_read_close
+};
diff --git a/libavformat/version.h b/libavformat/version.h
index 18c2f5fec2..493a0b337f 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -32,7 +32,7 @@ 
 // Major bumping may affect Ticket5467, 5421, 5451(compatibility with Chromium)
 // Also please add any ticket numbers that you believe might be affected here
 #define LIBAVFORMAT_VERSION_MAJOR  58
-#define LIBAVFORMAT_VERSION_MINOR  42
+#define LIBAVFORMAT_VERSION_MINOR  43
 #define LIBAVFORMAT_VERSION_MICRO 100
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \