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ffmpeg/libavformat/pp_bnk.c

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/*
* 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 "demux.h"
#include "internal.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.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;
int is_music;
} 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)
return ret;
/* 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)
return AVERROR_INVALIDDATA;
if (e.always1_1 != 1 || e.always1_2 != 1) {
avpriv_request_sample(s, "Non-one track header values");
return AVERROR_PATCHWELCOME;
}
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) {
return ret;
}
}
/* File is only a header. */
if (ctx->track_count == 0)
return AVERROR_INVALIDDATA;
ctx->is_music = (hdr.flags & PP_BNK_FLAG_MUSIC) &&
(ctx->track_count == 2) &&
(ctx->tracks[0].data_size == ctx->tracks[1].data_size);
/* Build the streams. */
for (int i = 0; i < (ctx->is_music ? 1 : ctx->track_count); i++) {
if (!(st = avformat_new_stream(s, NULL)))
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_S16P;
av_channel_layout_default(&par->ch_layout, ctx->is_music + 1);
par->sample_rate = hdr.sample_rate;
par->bits_per_coded_sample = 4;
par->block_align = 1;
par->bit_rate = par->sample_rate * (int64_t)par->bits_per_coded_sample *
par->ch_layout.nb_channels;
avpriv_set_pts_info(st, 64, 1, par->sample_rate);
st->start_time = 0;
st->duration = ctx->tracks[i].data_size * 2;
}
return 0;
}
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 (!ctx->is_music) {
ret = av_get_packet(s->pb, pkt, size);
if (ret == AVERROR_EOF) {
/* If we've hit EOF, don't attempt this track again. */
trk->data_size = trk->bytes_read;
continue;
}
} else {
if (!pkt->data && (ret = av_new_packet(pkt, size * 2)) < 0)
return ret;
ret = avio_read(s->pb, pkt->data + size * ctx->current_track, size);
if (ret >= 0 && ret != size) {
/* Only return stereo packets if both tracks could be read. */
ret = AVERROR_EOF;
}
}
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;
if (ctx->is_music) {
if (pkt->stream_index == 0)
continue;
pkt->stream_index = 0;
}
ctx->current_track++;
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;
}
static int pp_bnk_seek(AVFormatContext *s, int stream_index,
int64_t pts, int flags)
{
PPBnkCtx *ctx = s->priv_data;
if (pts != 0)
return AVERROR(EINVAL);
if (ctx->is_music) {
av_assert0(stream_index == 0);
ctx->tracks[0].bytes_read = 0;
ctx->tracks[1].bytes_read = 0;
} else {
ctx->tracks[stream_index].bytes_read = 0;
}
return 0;
}
const FFInputFormat ff_pp_bnk_demuxer = {
.p.name = "pp_bnk",
.p.long_name = NULL_IF_CONFIG_SMALL("Pro Pinball Series Soundbank"),
.priv_data_size = sizeof(PPBnkCtx),
.flags_internal = FF_FMT_INIT_CLEANUP,
.read_probe = pp_bnk_probe,
.read_header = pp_bnk_read_header,
.read_packet = pp_bnk_read_packet,
.read_close = pp_bnk_read_close,
.read_seek = pp_bnk_seek,
};
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