libavformat/sdsdec.c
7f9978b0
 /*
  * MIDI Sample Dump Standard format demuxer
  * Copyright (c) 2017 Paul B Mahol
  *
  * 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 "libavutil/intreadwrite.h"
 #include "avformat.h"
 #include "internal.h"
 
 typedef struct SDSContext {
     uint8_t data[120];
     int bit_depth;
     int size;
     void (*read_block)(const uint8_t *src, uint32_t *dst);
 } SDSContext;
 
 static int sds_probe(AVProbeData *p)
 {
     if (AV_RB32(p->buf) == 0xF07E0001 && p->buf[20] == 0xF7 &&
         p->buf[6] >= 8 && p->buf[6] <= 28)
         return AVPROBE_SCORE_EXTENSION;
     return 0;
 }
 
 static void byte2_read(const uint8_t *src, uint32_t *dst)
 {
     int i;
 
     for (i = 0; i < 120; i += 2) {
         unsigned sample = (src[i + 0] << 25) + (src[i + 1] << 18);
 
         dst[i / 2] = sample;
     }
 }
 
 static void byte3_read(const uint8_t *src, uint32_t *dst)
 {
     int i;
 
     for (i = 0; i < 120; i += 3) {
         unsigned sample;
 
         sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11);
         dst[i / 3] = sample;
     }
 }
 
 static void byte4_read(const uint8_t *src, uint32_t *dst)
 {
     int i;
 
     for (i = 0; i < 120; i += 4) {
         unsigned sample;
 
         sample = (src[i + 0] << 25) | (src[i + 1] << 18) | (src[i + 2] << 11) | (src[i + 3] << 4);
         dst[i / 4] = sample;
     }
 }
 
 #define SDS_3BYTE_TO_INT_DECODE(x) (((x) & 0x7F) | (((x) & 0x7F00) >> 1) | (((x) & 0x7F0000) >> 2))
 
 static int sds_read_header(AVFormatContext *ctx)
 {
     SDSContext *s = ctx->priv_data;
     unsigned sample_period;
     AVIOContext *pb = ctx->pb;
     AVStream *st;
 
     st = avformat_new_stream(ctx, NULL);
     if (!st)
         return AVERROR(ENOMEM);
 
     avio_skip(pb, 4);
     avio_skip(pb, 2);
 
     s->bit_depth = avio_r8(pb);
     if (s->bit_depth < 8 || s->bit_depth > 28)
         return AVERROR_INVALIDDATA;
 
     if (s->bit_depth < 14) {
         s->read_block = byte2_read;
         s->size = 60 * 4;
     } else if (s->bit_depth < 21) {
         s->read_block = byte3_read;
         s->size = 40 * 4;
     } else {
         s->read_block = byte4_read;
         s->size = 30 * 4;
     }
     st->codecpar->codec_id = AV_CODEC_ID_PCM_U32LE;
 
     sample_period = avio_rl24(pb);
     sample_period = SDS_3BYTE_TO_INT_DECODE(sample_period);
     avio_skip(pb, 11);
 
     st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
     st->codecpar->channels = 1;
     st->codecpar->sample_rate = sample_period ? 1000000000 / sample_period : 16000;
     st->duration = (avio_size(pb) - 21) / (127) * s->size / 4;
 
     avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
 
     return 0;
 }
 
 static int sds_read_packet(AVFormatContext *ctx, AVPacket *pkt)
 {
     SDSContext *s = ctx->priv_data;
     AVIOContext *pb = ctx->pb;
     int64_t pos;
     int ret;
 
     if (avio_feof(pb))
         return AVERROR_EOF;
 
     pos = avio_tell(pb);
     if (avio_rb16(pb) != 0xF07E)
         return AVERROR_INVALIDDATA;
     avio_skip(pb, 3);
 
     ret = av_new_packet(pkt, s->size);
     if (ret < 0)
         return ret;
 
     ret = avio_read(pb, s->data, 120);
 
     s->read_block(s->data, (uint32_t *)pkt->data);
 
     avio_skip(pb, 1); // checksum
     if (avio_r8(pb) != 0xF7)
         return AVERROR_INVALIDDATA;
 
     pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
     pkt->stream_index = 0;
     pkt->pos = pos;
 
     return ret;
 }
 
 AVInputFormat ff_sds_demuxer = {
     .name           = "sds",
     .long_name      = NULL_IF_CONFIG_SMALL("MIDI Sample Dump Standard"),
     .priv_data_size = sizeof(SDSContext),
     .read_probe     = sds_probe,
     .read_header    = sds_read_header,
     .read_packet    = sds_read_packet,
     .extensions     = "sds",
     .flags          = AVFMT_GENERIC_INDEX,
 };