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329 lines
10 KiB
C
329 lines
10 KiB
C
/*
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* Pro Pinball Series Soundbank (44c, 22c, 11c, 5c) demuxer.
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*
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* Copyright (C) 2020 Zane van Iperen (zane@zanevaniperen.com)
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include "internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/internal.h"
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#define PP_BNK_MAX_READ_SIZE 4096
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#define PP_BNK_FILE_HEADER_SIZE 20
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#define PP_BNK_TRACK_SIZE 20
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typedef struct PPBnkHeader {
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uint32_t bank_id; /*< Bank ID, useless for our purposes. */
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uint32_t sample_rate; /*< Sample rate of the contained tracks. */
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uint32_t always1; /*< Unknown, always seems to be 1. */
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uint32_t track_count; /*< Number of tracks in the file. */
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uint32_t flags; /*< Flags. */
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} PPBnkHeader;
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typedef struct PPBnkTrack {
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uint32_t id; /*< Track ID. Usually track[i].id == track[i-1].id + 1, but not always */
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uint32_t size; /*< Size of the data in bytes. */
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uint32_t sample_rate; /*< Sample rate. */
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uint32_t always1_1; /*< Unknown, always seems to be 1. */
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uint32_t always1_2; /*< Unknown, always seems to be 1. */
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} PPBnkTrack;
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typedef struct PPBnkCtxTrack {
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int64_t data_offset;
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uint32_t data_size;
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uint32_t bytes_read;
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} PPBnkCtxTrack;
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typedef struct PPBnkCtx {
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int track_count;
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PPBnkCtxTrack *tracks;
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uint32_t current_track;
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int is_music;
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} PPBnkCtx;
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enum {
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PP_BNK_FLAG_PERSIST = (1 << 0), /*< This is a large file, keep in memory. */
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PP_BNK_FLAG_MUSIC = (1 << 1), /*< This is music. */
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PP_BNK_FLAG_MASK = (PP_BNK_FLAG_PERSIST | PP_BNK_FLAG_MUSIC)
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};
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static void pp_bnk_parse_header(PPBnkHeader *hdr, const uint8_t *buf)
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{
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hdr->bank_id = AV_RL32(buf + 0);
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hdr->sample_rate = AV_RL32(buf + 4);
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hdr->always1 = AV_RL32(buf + 8);
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hdr->track_count = AV_RL32(buf + 12);
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hdr->flags = AV_RL32(buf + 16);
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}
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static void pp_bnk_parse_track(PPBnkTrack *trk, const uint8_t *buf)
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{
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trk->id = AV_RL32(buf + 0);
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trk->size = AV_RL32(buf + 4);
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trk->sample_rate = AV_RL32(buf + 8);
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trk->always1_1 = AV_RL32(buf + 12);
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trk->always1_2 = AV_RL32(buf + 16);
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}
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static int pp_bnk_probe(const AVProbeData *p)
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{
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uint32_t sample_rate = AV_RL32(p->buf + 4);
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uint32_t track_count = AV_RL32(p->buf + 12);
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uint32_t flags = AV_RL32(p->buf + 16);
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if (track_count == 0 || track_count > INT_MAX)
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return 0;
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if ((sample_rate != 5512) && (sample_rate != 11025) &&
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(sample_rate != 22050) && (sample_rate != 44100))
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return 0;
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/* Check the first track header. */
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if (AV_RL32(p->buf + 28) != sample_rate)
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return 0;
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if ((flags & ~PP_BNK_FLAG_MASK) != 0)
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return 0;
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return AVPROBE_SCORE_MAX / 4 + 1;
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}
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static int pp_bnk_read_header(AVFormatContext *s)
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{
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int64_t ret;
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AVStream *st;
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AVCodecParameters *par;
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PPBnkCtx *ctx = s->priv_data;
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uint8_t buf[FFMAX(PP_BNK_FILE_HEADER_SIZE, PP_BNK_TRACK_SIZE)];
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PPBnkHeader hdr;
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if ((ret = avio_read(s->pb, buf, PP_BNK_FILE_HEADER_SIZE)) < 0)
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return ret;
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else if (ret != PP_BNK_FILE_HEADER_SIZE)
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return AVERROR(EIO);
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pp_bnk_parse_header(&hdr, buf);
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if (hdr.track_count == 0 || hdr.track_count > INT_MAX)
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return AVERROR_INVALIDDATA;
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if (hdr.sample_rate == 0 || hdr.sample_rate > INT_MAX)
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return AVERROR_INVALIDDATA;
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if (hdr.always1 != 1) {
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avpriv_request_sample(s, "Non-one header value");
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return AVERROR_PATCHWELCOME;
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}
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ctx->track_count = hdr.track_count;
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if (!(ctx->tracks = av_malloc_array(hdr.track_count, sizeof(PPBnkCtxTrack))))
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return AVERROR(ENOMEM);
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/* Parse and validate each track. */
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for (int i = 0; i < hdr.track_count; i++) {
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PPBnkTrack e;
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PPBnkCtxTrack *trk = ctx->tracks + i;
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ret = avio_read(s->pb, buf, PP_BNK_TRACK_SIZE);
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if (ret < 0 && ret != AVERROR_EOF)
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return ret;
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/* Short byte-count or EOF, we have a truncated file. */
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if (ret != PP_BNK_TRACK_SIZE) {
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av_log(s, AV_LOG_WARNING, "File truncated at %d/%u track(s)\n",
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i, hdr.track_count);
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ctx->track_count = i;
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break;
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}
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pp_bnk_parse_track(&e, buf);
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/* The individual sample rates of all tracks must match that of the file header. */
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if (e.sample_rate != hdr.sample_rate)
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return AVERROR_INVALIDDATA;
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if (e.always1_1 != 1 || e.always1_2 != 1) {
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avpriv_request_sample(s, "Non-one track header values");
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return AVERROR_PATCHWELCOME;
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}
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trk->data_offset = avio_tell(s->pb);
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trk->data_size = e.size;
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trk->bytes_read = 0;
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/*
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* Skip over the data to the next stream header.
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* Sometimes avio_skip() doesn't detect EOF. If it doesn't, either:
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* - the avio_read() above will, or
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* - pp_bnk_read_packet() will read a truncated last track.
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*/
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if ((ret = avio_skip(s->pb, e.size)) == AVERROR_EOF) {
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ctx->track_count = i + 1;
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av_log(s, AV_LOG_WARNING,
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"Track %d has truncated data, assuming track count == %d\n",
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i, ctx->track_count);
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break;
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} else if (ret < 0) {
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return ret;
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}
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}
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/* File is only a header. */
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if (ctx->track_count == 0)
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return AVERROR_INVALIDDATA;
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ctx->is_music = (hdr.flags & PP_BNK_FLAG_MUSIC) &&
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(ctx->track_count == 2) &&
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(ctx->tracks[0].data_size == ctx->tracks[1].data_size);
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/* Build the streams. */
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for (int i = 0; i < (ctx->is_music ? 1 : ctx->track_count); i++) {
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if (!(st = avformat_new_stream(s, NULL)))
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return AVERROR(ENOMEM);
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par = st->codecpar;
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par->codec_type = AVMEDIA_TYPE_AUDIO;
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par->codec_id = AV_CODEC_ID_ADPCM_IMA_CUNNING;
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par->format = AV_SAMPLE_FMT_S16P;
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av_channel_layout_default(&par->ch_layout, ctx->is_music + 1);
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par->sample_rate = hdr.sample_rate;
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par->bits_per_coded_sample = 4;
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par->block_align = 1;
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par->bit_rate = par->sample_rate * (int64_t)par->bits_per_coded_sample *
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par->ch_layout.nb_channels;
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avpriv_set_pts_info(st, 64, 1, par->sample_rate);
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st->start_time = 0;
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st->duration = ctx->tracks[i].data_size * 2;
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}
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return 0;
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}
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static int pp_bnk_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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PPBnkCtx *ctx = s->priv_data;
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/*
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* Read a packet from each track, round-robin style.
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* This method is nasty, but needed to avoid "Too many packets buffered" errors.
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*/
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for (int i = 0; i < ctx->track_count; i++, ctx->current_track++)
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{
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int64_t ret;
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int size;
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PPBnkCtxTrack *trk;
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ctx->current_track %= ctx->track_count;
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trk = ctx->tracks + ctx->current_track;
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if (trk->bytes_read == trk->data_size)
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continue;
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if ((ret = avio_seek(s->pb, trk->data_offset + trk->bytes_read, SEEK_SET)) < 0)
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return ret;
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else if (ret != trk->data_offset + trk->bytes_read)
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return AVERROR(EIO);
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size = FFMIN(trk->data_size - trk->bytes_read, PP_BNK_MAX_READ_SIZE);
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if (!ctx->is_music) {
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ret = av_get_packet(s->pb, pkt, size);
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if (ret == AVERROR_EOF) {
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/* If we've hit EOF, don't attempt this track again. */
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trk->data_size = trk->bytes_read;
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continue;
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}
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} else {
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if (!pkt->data && (ret = av_new_packet(pkt, size * 2)) < 0)
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return ret;
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ret = avio_read(s->pb, pkt->data + size * ctx->current_track, size);
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if (ret >= 0 && ret != size) {
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/* Only return stereo packets if both tracks could be read. */
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ret = AVERROR_EOF;
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}
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}
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if (ret < 0)
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return ret;
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trk->bytes_read += ret;
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pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
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pkt->stream_index = ctx->current_track;
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pkt->duration = ret * 2;
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if (ctx->is_music) {
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if (pkt->stream_index == 0)
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continue;
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pkt->stream_index = 0;
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}
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ctx->current_track++;
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return 0;
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}
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/* If we reach here, we're done. */
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return AVERROR_EOF;
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}
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static int pp_bnk_read_close(AVFormatContext *s)
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{
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PPBnkCtx *ctx = s->priv_data;
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av_freep(&ctx->tracks);
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return 0;
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}
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static int pp_bnk_seek(AVFormatContext *s, int stream_index,
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int64_t pts, int flags)
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{
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PPBnkCtx *ctx = s->priv_data;
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if (pts != 0)
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return AVERROR(EINVAL);
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if (ctx->is_music) {
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av_assert0(stream_index == 0);
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ctx->tracks[0].bytes_read = 0;
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ctx->tracks[1].bytes_read = 0;
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} else {
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ctx->tracks[stream_index].bytes_read = 0;
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}
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return 0;
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}
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const AVInputFormat ff_pp_bnk_demuxer = {
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.name = "pp_bnk",
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.long_name = NULL_IF_CONFIG_SMALL("Pro Pinball Series Soundbank"),
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.priv_data_size = sizeof(PPBnkCtx),
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.flags_internal = FF_FMT_INIT_CLEANUP,
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.read_probe = pp_bnk_probe,
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.read_header = pp_bnk_read_header,
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.read_packet = pp_bnk_read_packet,
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.read_close = pp_bnk_read_close,
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.read_seek = pp_bnk_seek,
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};
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