diff mbox series

[FFmpeg-devel,v6,2/2] avformat: add demuxer for Pro Pinball Series' Soundbanks

Message ID 20200329171534.25441-3-zane@zanevaniperen.com
State Superseded
Headers show
Series Pro Pinball Series Soundbank demuxer + decoder.
Related show

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Context Check Description
andriy/ffmpeg-patchwork pending
andriy/ffmpeg-patchwork success Applied patch
andriy/ffmpeg-patchwork success Configure finished
andriy/ffmpeg-patchwork success Make finished
andriy/ffmpeg-patchwork success Make fate finished

Commit Message

Zane van Iperen March 29, 2020, 5:18 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     | 263 +++++++++++++++++++++++++++++++++++++++
 libavformat/version.h    |   2 +-
 5 files changed, 267 insertions(+), 1 deletion(-)
 create mode 100644 libavformat/pp_bnk.c

Comments

Anton Khirnov April 6, 2020, 1 p.m. UTC | #1
Quoting Zane van Iperen (2020-03-29 19:18:20)
> Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
> +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 ((ret = av_reallocp_array(&ctx->tracks, hdr.track_count, sizeof(PPBnkCtxTrack))))

Why realloc? Seems this is only allocated once.
> +        return ret;
> +
> +    /* 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) {
> +            av_freep(&ctx->tracks);

You are duplicating this free at too many places. Would be better to
have a cleanup block at the end and jump to that.

> +            return ret;
> +        } else if (ret != PP_BNK_TRACK_SIZE) {
> +            av_freep(&ctx->tracks);
> +            return AVERROR(EIO);
> +        }
> +
> +        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) {
> +            av_freep(&ctx->tracks);
> +            return AVERROR_INVALIDDATA;
> +        }
> +
> +        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++) {
> +

nit: unnecessary empty line

> +        if (!(st = avformat_new_stream(s, NULL))) {
> +            av_freep(&ctx->tracks);
> +            return AVERROR(ENOMEM);
> +        }
> +
> +        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) {
> +        av_freep(&ctx->tracks);
> +        return ret;
> +    } else if (ret != ctx->tracks[0].data_offset) {
> +        av_freep(&ctx->tracks);
> +        return AVERROR(EIO);
> +    }
> +
> +    return 0;
> +}
> +
> +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) {
> +

nit: unnecessary empty line

Otherwise looks ok, but would be nice to have some tests.
Zane van Iperen April 6, 2020, 1:26 p.m. UTC | #2
On Mon, 06 Apr 2020 15:00:01 +0200
"Anton Khirnov" <anton@khirnov.net> wrote:

> Quoting Zane van Iperen (2020-03-29 19:18:20)
> > Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
> > +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 ((ret = av_reallocp_array(&ctx->tracks, hdr.track_count,
> > sizeof(PPBnkCtxTrack))))  
> 
> Why realloc? Seems this is only allocated once.

Good question. Fixed.


> > +        return ret;
> > +
> > +    /* 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) {
> > +            av_freep(&ctx->tracks);  
> 
> You are duplicating this free at too many places. Would be better to
> have a cleanup block at the end and jump to that.
> 

I did that originally, however it's at the awkward spot where doing
that causes the code to be larger than the way it is currently.

I'll change it.

> > +            return ret;
> > +        } else if (ret != PP_BNK_TRACK_SIZE) {
> > +            av_freep(&ctx->tracks);
> > +            return AVERROR(EIO);
> > +        }
> > +
> > +        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) {
> > +            av_freep(&ctx->tracks);
> > +            return AVERROR_INVALIDDATA;
> > +        }
> > +
> > +        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++) {
> > +  
> 
> nit: unnecessary empty line
> 

Nit picked.

> > +        if (!(st = avformat_new_stream(s, NULL))) {
> > +            av_freep(&ctx->tracks);
> > +            return AVERROR(ENOMEM);
> > +        }
> > +
> > +        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) {
> > +        av_freep(&ctx->tracks);
> > +        return ret;
> > +    } else if (ret != ctx->tracks[0].data_offset) {
> > +        av_freep(&ctx->tracks);
> > +        return AVERROR(EIO);
> > +    }
> > +
> > +    return 0;
> > +}
> > +
> > +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) {
> > +  
> 
> nit: unnecessary empty line

Fixed.

> 
> Otherwise looks ok, but would be nice to have some tests.
> 

I have tests ready, the plan was to send them if merged (and after
samples are uploaded) to avoid https://patchwork.ffmpeg.org/ failures.

Should I include them in this irregardless?

Zane

> --
> Anton Khirnov
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
> 
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
Paul B Mahol April 6, 2020, 1:29 p.m. UTC | #3
On 4/6/20, Zane van Iperen <zane@zanevaniperen.com> wrote:
> On Mon, 06 Apr 2020 15:00:01 +0200
> "Anton Khirnov" <anton@khirnov.net> wrote:
>
>> Quoting Zane van Iperen (2020-03-29 19:18:20)
>> > Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
>> > +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 ((ret = av_reallocp_array(&ctx->tracks, hdr.track_count,
>> > sizeof(PPBnkCtxTrack))))
>>
>> Why realloc? Seems this is only allocated once.
>
> Good question. Fixed.
>
>
>> > +        return ret;
>> > +
>> > +    /* 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) {
>> > +            av_freep(&ctx->tracks);
>>
>> You are duplicating this free at too many places. Would be better to
>> have a cleanup block at the end and jump to that.
>>
>
> I did that originally, however it's at the awkward spot where doing
> that causes the code to be larger than the way it is currently.
>
> I'll change it.
>
>> > +            return ret;
>> > +        } else if (ret != PP_BNK_TRACK_SIZE) {
>> > +            av_freep(&ctx->tracks);
>> > +            return AVERROR(EIO);
>> > +        }
>> > +
>> > +        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) {
>> > +            av_freep(&ctx->tracks);
>> > +            return AVERROR_INVALIDDATA;
>> > +        }
>> > +
>> > +        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++) {
>> > +
>>
>> nit: unnecessary empty line
>>
>
> Nit picked.
>
>> > +        if (!(st = avformat_new_stream(s, NULL))) {
>> > +            av_freep(&ctx->tracks);
>> > +            return AVERROR(ENOMEM);
>> > +        }
>> > +
>> > +        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) {
>> > +        av_freep(&ctx->tracks);
>> > +        return ret;
>> > +    } else if (ret != ctx->tracks[0].data_offset) {
>> > +        av_freep(&ctx->tracks);
>> > +        return AVERROR(EIO);
>> > +    }
>> > +
>> > +    return 0;
>> > +}
>> > +
>> > +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) {
>> > +
>>
>> nit: unnecessary empty line
>
> Fixed.
>
>>
>> Otherwise looks ok, but would be nice to have some tests.
>>
>
> I have tests ready, the plan was to send them if merged (and after
> samples are uploaded) to avoid https://patchwork.ffmpeg.org/ failures.
>
> Should I include them in this irregardless?

No, Never in same patch.

>
> Zane
>
>> --
>> Anton Khirnov
>> _______________________________________________
>> ffmpeg-devel mailing list
>> ffmpeg-devel@ffmpeg.org
>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>
>> To unsubscribe, visit link above, or email
>> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
>
> _______________________________________________
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
> To unsubscribe, visit link above, or email
> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
diff mbox series

Patch

diff --git a/Changelog b/Changelog
index 4bcd109f64..8e19a71e1a 100644
--- a/Changelog
+++ b/Changelog
@@ -56,6 +56,7 @@  version <next>:
 - CRI HCA demuxer
 - overlay_cuda filter
 - 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..592d25ab52
--- /dev/null
+++ b/libavformat/pp_bnk.c
@@ -0,0 +1,263 @@ 
+/*
+ * 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. 2 == music, 0 == sfx. */
+} 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 ((ret = av_reallocp_array(&ctx->tracks, hdr.track_count, sizeof(PPBnkCtxTrack))))
+        return ret;
+
+    /* 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) {
+            av_freep(&ctx->tracks);
+            return ret;
+        } else if (ret != PP_BNK_TRACK_SIZE) {
+            av_freep(&ctx->tracks);
+            return AVERROR(EIO);
+        }
+
+        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) {
+            av_freep(&ctx->tracks);
+            return AVERROR_INVALIDDATA;
+        }
+
+        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))) {
+            av_freep(&ctx->tracks);
+            return AVERROR(ENOMEM);
+        }
+
+        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) {
+        av_freep(&ctx->tracks);
+        return ret;
+    } else if (ret != ctx->tracks[0].data_offset) {
+        av_freep(&ctx->tracks);
+        return AVERROR(EIO);
+    }
+
+    return 0;
+}
+
+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, \