avformat: add msf demuxer

Signed-off-by: Paul B Mahol <onemda@gmail.com>

1 files changed, 95 insertions, 0 deletions

diff --git a/libavformat/msf.c b/libavformat/msf.c
new file mode 100644
index 0000000000..754ad44408
--- /dev/null
+++ b/libavformat/msf.c
@@ -0,0 +1,95 @@
+/*
+ * MSF demuxer
+ * Copyright (c) 2015 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"
+
+static int msf_probe(AVProbeData *p)
+{
+    if (memcmp(p->buf, "MSF", 3))
+        return 0;
+
+    if (AV_RB32(p->buf+8) <= 0)
+        return 0;
+
+    if (AV_RB32(p->buf+16) <= 0)
+        return 0;
+
+    return AVPROBE_SCORE_MAX / 3 * 2;
+}
+
+static int msf_read_header(AVFormatContext *s)
+{
+    unsigned codec, align, size;
+    AVStream *st;
+
+    avio_skip(s->pb, 4);
+
+    st = avformat_new_stream(s, NULL);
+    if (!st)
+        return AVERROR(ENOMEM);
+
+    st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
+    codec                  = avio_rb32(s->pb);
+    st->codec->channels    = avio_rb32(s->pb);
+    if (st->codec->channels <= 0)
+        return AVERROR_INVALIDDATA;
+    size = avio_rb32(s->pb);
+    st->codec->sample_rate = avio_rb32(s->pb);
+    if (st->codec->sample_rate <= 0)
+        return AVERROR_INVALIDDATA;
+    align = avio_rb32(s->pb) ;
+    if (align > INT_MAX / st->codec->channels)
+        return AVERROR_INVALIDDATA;
+    st->codec->block_align = align;
+    switch (codec) {
+    case 0: st->codec->codec_id = AV_CODEC_ID_PCM_S16BE; break;
+    case 3: st->codec->block_align = 16 * st->codec->channels;
+            st->codec->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
+    case 7: st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
+            st->codec->codec_id = AV_CODEC_ID_MP3;       break;
+    default:
+            avpriv_request_sample(s, "Codec %d", codec);
+            return AVERROR_PATCHWELCOME;
+    }
+    st->duration = av_get_audio_frame_duration(st->codec, size);
+    avio_skip(s->pb, 0x40 - avio_tell(s->pb));
+    avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+
+    return 0;
+}
+
+static int msf_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+    AVCodecContext *codec = s->streams[0]->codec;
+
+    return av_get_packet(s->pb, pkt, codec->block_align ? codec->block_align : 1024 * codec->channels);
+}
+
+AVInputFormat ff_msf_demuxer = {
+    .name           = "msf",
+    .long_name      = NULL_IF_CONFIG_SMALL("MSF"),
+    .read_probe     = msf_probe,
+    .read_header    = msf_read_header,
+    .read_packet    = msf_read_packet,
+    .extensions     = "msf",
+};