[FFmpeg-devel] [PATCH v11] avformat: add demuxer for Pro Pinball Series' Soundbanks

Zane van Iperen zane at zanevaniperen.com
Tue Apr 28 15:10:33 EEST 2020


Signed-off-by: Zane van Iperen <zane at zanevaniperen.com>
---
 Changelog                |   1 +
 libavformat/Makefile     |   1 +
 libavformat/allformats.c |   1 +
 libavformat/pp_bnk.c     | 289 +++++++++++++++++++++++++++++++++++++++
 libavformat/version.h    |   4 +-
 5 files changed, 294 insertions(+), 2 deletions(-)
 create mode 100644 libavformat/pp_bnk.c

diff --git a/Changelog b/Changelog
index 83b8a4a46e..4cd324ffc2 100644
--- a/Changelog
+++ b/Changelog
@@ -63,6 +63,7 @@ version <next>:
 - maskedthreshold filter
 - Support for muxing pcm and pgs in m2ts
 - Cunning Developments ADPCM decoder
+- 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..5f9fc2d373
--- /dev/null
+++ b/libavformat/pp_bnk.c
@@ -0,0 +1,289 @@
+/*
+ * Pro Pinball Series Soundbank (44c, 22c, 11c, 5c) demuxer.
+ *
+ * Copyright (C) 2020 Zane van Iperen (zane at 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 || 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;
+
+        /*
+         * 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;
+    }
+
+    /* Seek to the start of the first stream. */
+    if ((ret = avio_seek(s->pb, ctx->tracks[0].data_offset, SEEK_SET)) < 0) {
+        goto fail;
+    } else if (ret != ctx->tracks[0].data_offset) {
+        ret = AVERROR(EIO);
+        goto fail;
+    }
+
+    return 0;
+
+fail:
+    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;
+
+    ctx->bytes_read    += ret;
+    pkt->flags         &= ~AV_PKT_FLAG_CORRUPT;
+    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 719cda6b98..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 101
+#define LIBAVFORMAT_VERSION_MINOR  43
+#define LIBAVFORMAT_VERSION_MICRO 100
 
 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
                                                LIBAVFORMAT_VERSION_MINOR, \
-- 
2.24.0




More information about the ffmpeg-devel mailing list