diff mbox series

[FFmpeg-devel,v13] avformat: add demuxer for Pro Pinball Series' Soundbanks

Message ID 20200504142515.190777-1-zane@zanevaniperen.com
State Accepted
Headers show
Series [FFmpeg-devel,v13] avformat: add demuxer for Pro Pinball Series' Soundbanks | expand

Checks

Context Check Description
andriy/default pending
andriy/make success Make finished
andriy/make_fate success Make fate finished

Commit Message

Zane van Iperen May 4, 2020, 2:25 p.m. UTC
Adds support for the soundbank files used by the Pro Pinball series of games.

v13:
  - Increment current_track after reading a packet.

v12: [9]
  - Read packets in a round-robin fashion to
    avoid "Too many packets buffered" errors.

v11: [8]
  - Change probe function to be all-or-nothing

v10: [7]
  - Change while() to for().

v9: [6]
  - Rebase after codec_id.h changes
  - style fixes
  - Fix an uninitialised variable read

v8: [5]
  - change "goto done" to a return + "goto fail"
  - Handle truncated files
  - Fix potential byte counter desync

v7: [4]
  - Fix empty lines
  - Use av_malloc_array() instead of av_reallocp_array()
  - Replace multiple av_freep()'s with a goto
  - Minor comment cleanups
  - Ask for a sample if unexpected header values are found

v6: [3]
  - fix tools/probetest failure

v5:
  - add probe function
  - add flag #define's

v4: [2]
  - fix adpcm index table type

v3: [1]
  - fix potential memory leak if read_header() fails
  - fix a buffer overread
  - attempt seek before updating state
  - remove unneeded check
  - naming fixes

v2:
  - Add sanity checks in header fields
  - Formatting and comment fixes
  - Change the struct names to match the files

[1]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/258672.html
[2]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/258918.html
[3]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/259278.html
[4]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/259864.html
[5]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/259863.html
[6]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/260706.html
[7]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/260854.html
[8]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/261497.html
[9]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-May/262030.html

Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
---
 Changelog                |   1 +
 libavformat/Makefile     |   1 +
 libavformat/allformats.c |   1 +
 libavformat/pp_bnk.c     | 292 +++++++++++++++++++++++++++++++++++++++
 libavformat/version.h    |   4 +-
 5 files changed, 297 insertions(+), 2 deletions(-)
 create mode 100644 libavformat/pp_bnk.c

Comments

Michael Niedermayer May 5, 2020, 5:42 p.m. UTC | #1
On Mon, May 04, 2020 at 02:25:56PM +0000, Zane van Iperen wrote:
> Adds support for the soundbank files used by the Pro Pinball series of games.
> 
> v13:
>   - Increment current_track after reading a packet.
> 
> v12: [9]
>   - Read packets in a round-robin fashion to
>     avoid "Too many packets buffered" errors.
> 
> v11: [8]
>   - Change probe function to be all-or-nothing
> 
> v10: [7]
>   - Change while() to for().
> 
> v9: [6]
>   - Rebase after codec_id.h changes
>   - style fixes
>   - Fix an uninitialised variable read
> 
> v8: [5]
>   - change "goto done" to a return + "goto fail"
>   - Handle truncated files
>   - Fix potential byte counter desync
> 
> v7: [4]
>   - Fix empty lines
>   - Use av_malloc_array() instead of av_reallocp_array()
>   - Replace multiple av_freep()'s with a goto
>   - Minor comment cleanups
>   - Ask for a sample if unexpected header values are found
> 
> v6: [3]
>   - fix tools/probetest failure
> 
> v5:
>   - add probe function
>   - add flag #define's
> 
> v4: [2]
>   - fix adpcm index table type
> 
> v3: [1]
>   - fix potential memory leak if read_header() fails
>   - fix a buffer overread
>   - attempt seek before updating state
>   - remove unneeded check
>   - naming fixes
> 
> v2:
>   - Add sanity checks in header fields
>   - Formatting and comment fixes
>   - Change the struct names to match the files
> 
> [1]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/258672.html
> [2]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/258918.html
> [3]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-March/259278.html
> [4]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/259864.html
> [5]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/259863.html
> [6]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/260706.html
> [7]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/260854.html
> [8]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-April/261497.html
> [9]: https://ffmpeg.org/pipermail/ffmpeg-devel/2020-May/262030.html

will apply without this development history
we normally dont include such things in git log and its quite long

but ill leave one link to the mailing list thread so anyone interrested
can find it

thx

[...]
diff mbox series

Patch

diff --git a/Changelog b/Changelog
index 9b3e34560f..aed70d8da5 100644
--- a/Changelog
+++ b/Changelog
@@ -64,6 +64,7 @@  version <next>:
 - Support for muxing pcm and pgs in m2ts
 - Cunning Developments ADPCM decoder
 - asubboost filter
+- Pro Pinball Series Soundbank demuxer
 
 
 version 4.2:
diff --git a/libavformat/Makefile b/libavformat/Makefile
index d4bed3c113..b744eb69b2 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..8364de1fd9
--- /dev/null
+++ b/libavformat/pp_bnk.c
@@ -0,0 +1,292 @@ 
+/*
+ * 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;
+    uint32_t        bytes_read;
+} PPBnkCtxTrack;
+
+typedef struct PPBnkCtx {
+    int             track_count;
+    PPBnkCtxTrack   *tracks;
+    uint32_t        current_track;
+} 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 || track_count > INT_MAX)
+        return 0;
+
+    if ((sample_rate !=  5512) && (sample_rate != 11025) &&
+        (sample_rate != 22050) && (sample_rate != 44100))
+        return 0;
+
+    /* Check the first track header. */
+    if (AV_RL32(p->buf + 28) != sample_rate)
+        return 0;
+
+    if ((flags & ~PP_BNK_FLAG_MASK) != 0)
+        return 0;
+
+    return AVPROBE_SCORE_MAX / 4 + 1;
+}
+
+static int pp_bnk_read_header(AVFormatContext *s)
+{
+    int64_t 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;
+        PPBnkCtxTrack *trk = ctx->tracks + i;
+
+        ret = avio_read(s->pb, buf, PP_BNK_TRACK_SIZE);
+        if (ret < 0 && ret != AVERROR_EOF)
+            goto fail;
+
+        /* Short byte-count or EOF, we have a truncated file. */
+        if (ret != PP_BNK_TRACK_SIZE) {
+            av_log(s, AV_LOG_WARNING, "File truncated at %d/%u track(s)\n",
+                   i, hdr.track_count);
+            ctx->track_count = i;
+            break;
+        }
+
+        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 fail;
+        }
+
+        if (e.always1_1 != 1 || e.always1_2 != 1) {
+            avpriv_request_sample(s, "Non-one track header values");
+            ret = AVERROR_PATCHWELCOME;
+            goto fail;
+        }
+
+        trk->data_offset = avio_tell(s->pb);
+        trk->data_size   = e.size;
+        trk->bytes_read  = 0;
+
+        /*
+         * Skip over the data to the next stream header.
+         * Sometimes avio_skip() doesn't detect EOF. If it doesn't, either:
+         *   - the avio_read() above will, or
+         *   - pp_bnk_read_packet() will read a truncated last track.
+         */
+        if ((ret = avio_skip(s->pb, e.size)) == AVERROR_EOF) {
+            ctx->track_count = i + 1;
+            av_log(s, AV_LOG_WARNING,
+                   "Track %d has truncated data, assuming track count == %d\n",
+                   i, ctx->track_count);
+            break;
+        } else if (ret < 0) {
+            goto fail;
+        }
+    }
+
+    /* File is only a header. */
+    if (ctx->track_count == 0) {
+        ret = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    /* Build the streams. */
+    for (int i = 0; i < ctx->track_count; i++) {
+        if (!(st = avformat_new_stream(s, NULL))) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
+        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;
+    }
+
+    return 0;
+
+fail:
+    av_freep(&ctx->tracks);
+    return ret;
+}
+
+static int pp_bnk_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    PPBnkCtx *ctx = s->priv_data;
+
+    /*
+     * Read a packet from each track, round-robin style.
+     * This method is nasty, but needed to avoid "Too many packets buffered" errors.
+     */
+    for (int i = 0; i < ctx->track_count; i++, ctx->current_track++)
+    {
+        int64_t ret;
+        int size;
+        PPBnkCtxTrack *trk;
+
+        ctx->current_track %= ctx->track_count;
+
+        trk = ctx->tracks + ctx->current_track;
+
+        if (trk->bytes_read == trk->data_size)
+            continue;
+
+        if ((ret = avio_seek(s->pb, trk->data_offset + trk->bytes_read, SEEK_SET)) < 0)
+            return ret;
+        else if (ret != trk->data_offset + trk->bytes_read)
+            return AVERROR(EIO);
+
+        size = FFMIN(trk->data_size - trk->bytes_read, PP_BNK_MAX_READ_SIZE);
+
+        if ((ret = av_get_packet(s->pb, pkt, size)) == AVERROR_EOF) {
+            /* If we've hit EOF, don't attempt this track again. */
+            trk->data_size = trk->bytes_read;
+            continue;
+        } else if (ret < 0) {
+            return ret;
+        }
+
+        trk->bytes_read    += ret;
+        pkt->flags         &= ~AV_PKT_FLAG_CORRUPT;
+        pkt->stream_index   = ctx->current_track++;
+        pkt->duration       = ret * 2;
+        return 0;
+    }
+
+    /* If we reach here, we're done. */
+    return AVERROR_EOF;
+}
+
+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 b9a014749c..493a0b337f 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -32,8 +32,8 @@ 
 // 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_MICRO 102
+#define LIBAVFORMAT_VERSION_MINOR  43
+#define LIBAVFORMAT_VERSION_MICRO 100
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \