[Ffmpeg-devel] [PATCH] demuxer for dcinema audio

[Reimar Döffinger]

Hi,
I'd like to suggest the attached patch for demuxing D-Cinema audio. It's
not exactly beautiful since it does something like decoding (changing
the bit order), but I though it to be the easier solution over adding a
special decoder (since it isn't used anywhere else).
The smil file that came with it said it was of the format "audio/302m",
and I assume it is SMPTE 302M uncompressed audio, but I don't have the
specs.
Please tell me what you think about it...
Oh, some samples (I already extended the MPlayer mpeg demuxer to play
the gxf file):
Address: ftp://www.ropa-net.de
Username: web49f4
Password: 123456

Greetings,
Reimar D?ffinger
-------------- next part --------------
Index: libavformat/Makefile
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/Makefile,v
retrieving revision 1.98
diff -u -r1.98 Makefile
--- libavformat/Makefile	13 Aug 2005 13:45:42 -0000	1.98
+++ libavformat/Makefile	27 Aug 2005 14:51:27 -0000
@@ -17,7 +17,7 @@
       yuv4mpeg.o 4xm.o flvenc.o flvdec.o movenc.o psxstr.o idroq.o ipmovie.o \
       nut.o wc3movie.o mp3.o westwood.o segafilm.o idcin.o flic.o \
       sierravmd.o matroska.o sol.o electronicarts.o nsvdec.o asf.o asf-enc.o \
-      ogg2.o oggparsevorbis.o oggparsetheora.o oggparseflac.o
+      ogg2.o oggparsevorbis.o oggparsetheora.o oggparseflac.o s302m.o
 AMROBJS=
 ifeq ($(AMR_NB),yes)
 AMROBJS= amr.o
Index: libavformat/allformats.c
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/allformats.c,v
retrieving revision 1.48
diff -u -r1.48 allformats.c
--- libavformat/allformats.c	15 Jul 2005 12:50:00 -0000	1.48
+++ libavformat/allformats.c	27 Aug 2005 14:51:27 -0000
@@ -109,6 +109,7 @@
     sol_init();
     ea_init();
     nsvdec_init();
+    s302m_init();
 
 #ifdef CONFIG_ENCODERS
     /* image formats */
Index: libavformat/avformat.h
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/avformat.h,v
retrieving revision 1.132
diff -u -r1.132 avformat.h
--- libavformat/avformat.h	21 Aug 2005 22:31:00 -0000	1.132
+++ libavformat/avformat.h	27 Aug 2005 14:51:30 -0000
@@ -546,6 +546,9 @@
 /* nsvdec.c */
 int nsvdec_init(void);
 
+/* s302m.c */
+int s302m_init(void);
+
 #include "rtp.h"
 
 #include "rtsp.h"
Index: libavformat/s302m.c
===================================================================
RCS file: libavformat/s302m.c
diff -N libavformat/s302m.c
--- /dev/null	1 Jan 1970 00:00:00 -0000
+++ libavformat/s302m.c	27 Aug 2005 14:51:30 -0000
@@ -0,0 +1,99 @@
+/*
+ * SMPTE 302M decoder.
+ * Copyright (c) 2005 Reimar D?ffinger.
+ *
+ * This library 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 of the License, or (at your option) any later version.
+ *
+ * This library 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 this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+#include "avformat.h"
+
+static int s302m_header(AVFormatContext *s, AVFormatParameters *ap) {
+  AVStream *st = av_new_stream(s, 0);
+  st->codec->codec_type = CODEC_TYPE_AUDIO;
+  st->codec->codec_tag = 0x1;
+  st->codec->channels = 6;
+  st->codec->sample_rate = 96000;
+  st->codec->bit_rate = 3 * 6 * 96000 * 8;
+  st->codec->block_align = 4 * 6;
+  // we internally convert to 32 bit since that's easier and faster
+  st->codec->bits_per_sample = 32;
+  return 0;
+}
+
+// This bitorder is the result of pure guesswork (counting how often
+// the bits flip), so it might be wrong.
+static inline uint32_t get_sample(ByteIOContext *pb) {
+  int i;
+  uint32_t in = (get_be16(pb) << 8) | get_byte(pb);
+  uint32_t out = 0;
+  uint32_t in_back = in;
+  uint8_t sync = in & 0x0f; // sync flags
+  in >>= 4;
+  for (i = 0; i < 20; i++) {
+    out <<= 1;
+    out |= (in & 1);
+    in >>= 1;
+  }
+  out <<= 12;
+  out |= sync; // sync flags go into lowest bits
+  // It seems those might also contain audio data,
+  // don't know where they belong then, samples welcome.
+  // These might be also useful to detect the number
+  // of channels in case there are files with != 6 channels
+  return out;
+}
+
+static int s302m_packet(AVFormatContext *s, AVPacket *pkt) {
+  ByteIOContext *pb = &s->pb;
+  uint32_t *buf;
+  int i;
+  int size = get_be16(pb);
+  get_be16(pb); // unkown
+  if (size % 3) {
+    av_log(s, AV_LOG_ERROR, "Packet size must be divisible by sample size.\n");
+    return AVERROR_IO;
+  }
+  size /= 3;
+  if (av_new_packet(pkt, size * 4))
+    return AVERROR_IO;
+  pkt->flags |= PKT_FLAG_KEY;
+  pkt->stream_index = 0;
+
+  buf = (uint32_t *)pkt->data;
+  for (i = 0; i < size; i++) {
+    buf[i] = get_sample(pb);
+    if (url_feof(pb))
+      break;
+  }
+  return i * 4;
+}
+
+static AVInputFormat s302m_iformat = {
+  "302",
+  "SMPTE 302M format",
+  0,
+  NULL,
+  s302m_header,
+  s302m_packet,
+  NULL,
+  NULL,
+  .extensions = "302",
+};
+
+int s302m_init(void)
+{
+  av_register_input_format(&s302m_iformat);
+  return 0;
+}
+