Add AC3 support to esmerge.

--HG--
extra : convert_revision : svn%3Aeff31bef-be4a-0410-a8fe-e47997df2690/trunk%4061

Makefile

@@ -90,6 +90,7 @@ BINDIR = bin
 # All of our non-program source files
 SRCS = \
  accessunit.c \
+ ac3.c \
  adts.c \
  avs.c \
  bitdata.c \
@@ -111,6 +112,7 @@ SRCS = \
 OBJS = \
  accessunit.o \
  avs.o \
+ ac3.o \
  adts.o \
  bitdata.o \
  es.o \
@@ -149,6 +151,7 @@ PROG_OBJS = \
   $(OBJDIR)/tsreport.o \
   $(OBJDIR)/tsserve.o \
   $(OBJDIR)/ts_packet_insert.o \
+  $(OBJDIR)/m2ts2ts.o
   $(OBJDIR)/pcapreport.o 
 #\
 #  $(OBJDIR)/test_ps.o
@@ -183,6 +186,7 @@ PROGS = \
   $(BINDIR)/tsplay \
   $(BINDIR)/tsserve \
   $(BINDIR)/ts_packet_insert \
+  $(BINDIR)/m2ts2ts
   $(BINDIR)/pcapreport 
 #\
 #  $(BINDIR)/test_ps
@@ -266,6 +270,8 @@ $(BINDIR)/ts2ps:		$(OBJDIR)/ts2ps.o $(LIB)
 $(BINDIR)/ts_packet_insert:	$(OBJDIR)/ts_packet_insert.o $(LIB)
 		$(CC) $< -o $(BINDIR)/ts_packet_insert $(LDFLAGS) $(LIBOPTS)
 
+$(BINDIR)/m2ts2ts:		$(OBJDIR)/m2ts2ts.o $(LIB)
+		$(CC) $< -o $(BINDIR)/m2ts2ts $(LDFLAGS) $(LIBOPTS)
 $(BINDIR)/pcapreport:	$(OBJDIR)/pcapreport.o $(LIB)
 		$(CC) $< -o $(BINDIR)/pcapreport $(LDFLAGS) $(LIBOPTS)
 
@@ -295,7 +301,7 @@ H262_H = h262_fns.h h262_defns.h
 TSWRITE_H = tswrite_fns.h tswrite_defns.h
 REVERSE_H = reverse_fns.h reverse_defns.h
 FILTER_H = filter_fns.h filter_defns.h $(REVERSE_H)
-AUDIO_H = adts_fns.h l2audio_fns.h audio_fns.h audio_defns.h adts_defns.h
+AUDIO_H = adts_fns.h l2audio_fns.h ac3_fns.h audio_fns.h audio_defns.h adts_defns.h
 
 # Everyone depends upon the basic configuration file
 $(LIB)($(OBJS)) $(TEST_OBJS) $(PROG_OBJS): compat.h
@@ -344,6 +350,8 @@ $(OBJDIR)/tsplay.o:       tsplay.c $(TS_H) misc_fns.h $(PS_H) $(PES_H) version.h
 	$(CC) -c $< -o $@ $(CFLAGS)
 $(OBJDIR)/tswrite.o:      tswrite.c misc_fns.h version.h
 	$(CC) -c $< -o $@ $(CFLAGS)
+$(OBJDIR)/m2ts2ts.o:	  m2ts2ts.c $(TS_H) misc_fns.h version.h
+	$(CC) -c $< -o $@ $(CFLAGS)
 $(OBJDIR)/pcapreport.o:      pcapreport.c pcap.h version.h misc_fns.h
 	$(CC) -c $< -o $@ $(CFLAGS)
 

Makefile.w32

@@ -74,6 +74,7 @@ PROGS = \
     $(EXEDIR)\ts2es.exe       \
     $(EXEDIR)\es2ts.exe       \
     $(EXEDIR)\esreverse.exe   \
+    $(EXEDIR)\m2ts2ts.exe     \
     $(EXEDIR)\ps2ts.exe       \
     $(EXEDIR)\psreport.exe    \
     $(EXEDIR)\stream_type.exe \
@@ -85,6 +86,7 @@ PROGS = \
 # Object files for the library
 LIB_OBJS = \
  $(OBJDIR)\accessunit.obj \
+ $(OBJDIR)\ac3.obj \
  $(OBJDIR)\adts.obj \
  $(OBJDIR)\avs.obj \
  $(OBJDIR)\audio.obj \
@@ -111,6 +113,7 @@ PROG_OBJS = \
   $(OBJDIR)/esmerge.obj \
   $(OBJDIR)/esreport.obj \
   $(OBJDIR)/esreverse.obj \
+  $(OBJDIR)/m2ts2ts.obj \
   $(OBJDIR)/ps2ts.obj \
   $(OBJDIR)/psreport.obj \
   $(OBJDIR)/stream_type.obj \
@@ -153,6 +156,9 @@ $(EXEDIR)\esreport.exe: $(OBJDIR)\esreport.obj $(LIBFILE)
 $(EXEDIR)\esreverse.exe: $(OBJDIR)\esreverse.obj $(LIBFILE)
 	link /out:$@ $(LOPT) $** wsock32.lib
 
+$(EXEDIR)\m2ts2ts.exe: $(OBJDIR)\m2ts2ts.obj $(LIBFILE)
+	link /out:$@ $(LOPT) $** wsock32.lib
+
 $(EXEDIR)\stream_type.exe: $(OBJDIR)\stream_type.obj $(LIBFILE)
 	link /out:$@ $(LOPT) $** wsock32.lib
 

ac3.c

@@ -0,0 +1,171 @@
+/*
+ * Support for ATSC Digital Audio Compression Standard, Revision B
+ * (AC3) audio streams.
+ *
+ * ***** BEGIN LICENSE BLOCK *****
+ * Version: MPL 1.1
+ *
+ * The contents of this file are subject to the Mozilla Public License Version
+ * 1.1 (the "License"); you may not use this file except in compliance with
+ * the License. You may obtain a copy of the License at
+ * http://www.mozilla.org/MPL/
+ *
+ * Software distributed under the License is distributed on an "AS IS" basis,
+ * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
+ * for the specific language governing rights and limitations under the
+ * License.
+ *
+ * The Original Code is the MPEG TS, PS and ES tools.
+ *
+ * The Initial Developer of the Original Code is Amino Communications Ltd.
+ * Portions created by the Initial Developer are Copyright (C) 2008
+ * the Initial Developer. All Rights Reserved.
+ *
+ * Contributor(s):
+ *   Kynesim Ltd, Cambridge UK
+ *
+ * ***** END LICENSE BLOCK *****
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <string.h>
+
+#include "compat.h"
+#include "misc_fns.h"
+#include "ac3_fns.h"
+
+static const unsigned int // Table 5.18, frame sizes
+l_frmsizecod[19][3] = {
+  { 64, 69, 96 },
+  { 80, 87, 120 },
+  { 96, 104, 144 },
+  { 112, 121, 168 },
+  { 128, 139, 192 },
+  { 160, 174, 240 },
+  { 192, 208, 288 },
+  { 224, 243, 336 },
+  { 256, 278, 384 },
+  { 320, 348, 480 },
+  { 384, 417, 576 },
+  { 448, 487, 672 },
+  { 512, 557, 768 },
+  { 640, 696, 960 },
+  { 768, 835, 1152 },
+  { 896, 975, 1344 },
+  { 1024, 1114, 1536 },
+  { 1152, 1253, 1728 },
+  { 1280, 1393, 1920 }
+};
+
+
+/*
+ * Read the next AC3 frame.
+ *
+ * Assumes that the input stream is synchronised - i.e., it does not
+ * try to cope if the next two bytes are not '0000 1011 0111 0111'
+ *
+ * - `file` is the file descriptor of the AC3 file to read from
+ * - `frame` is the AC3 frame that is read
+ *
+ * Returns 0 if all goes well, EOF if end-of-file is read, and 1 if something
+ * goes wrong.
+ */
+#define SYNCINFO_SIZE 5
+
+int read_next_ac3_frame(int            file,
+			audio_frame_p *frame)
+{
+  int   i, err;
+  byte  sync_info[SYNCINFO_SIZE];
+  byte *data = NULL;
+  int   fscod;
+  int   frmsizecod;
+  int   frame_length;
+
+  offset_t posn = tell_file(file);
+
+  err = read_bytes(file, SYNCINFO_SIZE, sync_info);
+  if (err == EOF)
+    return EOF;
+  else if (err)
+  {
+    fprintf(stderr, "### Error reading syncinfo from AC3 file\n");
+    fprintf(stderr, "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
+    return 1;
+  }
+
+  if (sync_info[0] != 0x0b || sync_info[1] != 0x77)
+  {
+    fprintf(stderr, "### AC3 frame does not start with 0x0b77"
+            " syncword - lost synchronisation?\n"
+            "    Found 0x%02x%02x instead of 0x0b77\n",
+            (unsigned)sync_info[0], (unsigned)sync_info[1]);
+    fprintf(stderr, "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
+    return 1;
+  }
+
+  fscod = sync_info[4] >> 6;
+  if (fscod == 3)
+  {
+    // Bad sample rate code
+    fprintf(stderr, "### Bad sample rate code in AC3 syncinfo\n");
+    fprintf(stderr, "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
+    return 1;
+  }
+
+  frmsizecod = sync_info[4] & 0x3f;
+  if (frmsizecod > 37)
+  {
+    fprintf(stderr, "### Bad frame size code %d in AC3 syncinfo\n",
+            frmsizecod);
+    fprintf(stderr, "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
+    return 1;
+  }
+
+  frame_length = l_frmsizecod[frmsizecod >> 1][fscod];
+  if (fscod == 1)
+    frame_length += frmsizecod & 1;
+  frame_length <<= 1;  // Convert from 16-bit words to bytes
+
+  data = malloc(frame_length);
+  if (data == NULL)
+  {
+    fprintf(stderr, "### Unable to extend data buffer for AC3 frame\n");
+    return 1;
+  }
+  for (i = 0; i < SYNCINFO_SIZE; i++)
+    data[i] = sync_info[i];
+
+  err = read_bytes(file, frame_length - SYNCINFO_SIZE,
+                   &data[SYNCINFO_SIZE]);
+  if (err)
+  {
+    if (err = EOF)
+      fprintf(stderr, "### Unexpected EOF reading rest of AC3 frame\n");
+    else
+      fprintf(stderr, "### Error reading rest of AC3 frame\n");
+    free(data);
+    return 1;
+  }
+
+  err = build_audio_frame(frame);
+  if (err)
+  {
+    free(data);
+    return 1;
+  }
+  (*frame)->data = data;
+  (*frame)->data_len = frame_length;
+
+  return 0;
+}
+
+
+// Local Variables:
+// tab-width: 8
+// indent-tabs-mode: nil
+// c-basic-offset: 2
+// End:
+// vim: set tabstop=8 shiftwidth=2 expandtab:

ac3_fns.h

@@ -0,0 +1,59 @@
+/*
+ * Support for ATSC Digital Audio Compression Standard, Revision B
+ * (AC3) audio streams.
+ *
+ * ***** BEGIN LICENSE BLOCK *****
+ * Version: MPL 1.1
+ *
+ * The contents of this file are subject to the Mozilla Public License Version
+ * 1.1 (the "License"); you may not use this file except in compliance with
+ * the License. You may obtain a copy of the License at
+ * http://www.mozilla.org/MPL/
+ *
+ * Software distributed under the License is distributed on an "AS IS" basis,
+ * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
+ * for the specific language governing rights and limitations under the
+ * License.
+ *
+ * The Original Code is the MPEG TS, PS and ES tools.
+ *
+ * The Initial Developer of the Original Code is Amino Communications Ltd.
+ * Portions created by the Initial Developer are Copyright (C) 2008
+ * the Initial Developer. All Rights Reserved.
+ *
+ * Contributor(s):
+ *   Kynesim Ltd, Cambridge UK
+ *
+ * ***** END LICENSE BLOCK *****
+ */
+
+#ifndef _ac3_fns
+#define _ac3_fns
+
+#include "audio_fns.h"
+
+
+/*
+ * Read the next AC3 frame.
+ *
+ * Assumes that the input stream is synchronised - i.e., it does not
+ * try to cope if the next two bytes are not '0000 1011 0111 0111'
+ *
+ * - `file` is the file descriptor of the AC3 file to read from
+ * - `frame` is the AC3 frame that is read
+ *
+ * Returns 0 if all goes well, EOF if end-of-file is read, and 1 if something
+ * goes wrong.
+ */
+extern int read_next_ac3_frame(int            file,
+                               audio_frame_p *frame);
+
+
+#endif // _ac3_fns
+
+// Local Variables:
+// tab-width: 8
+// indent-tabs-mode: nil
+// c-basic-offset: 2
+// End:
+// vim: set tabstop=8 shiftwidth=2 expandtab:

audio.c

@@ -35,6 +35,7 @@
 #include "audio_fns.h"
 #include "adts_fns.h"
 #include "l2audio_fns.h"
+#include "ac3_fns.h"
 
 /*
  * Build a new generic audio frame datastructure
@@ -106,6 +107,8 @@ extern int read_next_audio_frame(int            file,
     return read_next_adts_frame(file,frame,0);
   case AUDIO_L2:
     return read_next_l2audio_frame(file,frame);
+  case AUDIO_AC3:
+    return read_next_ac3_frame(file, frame);
   default:
     fprintf(stderr,"### Unrecognised audio type %d - cannot get next audio frame\n",
             audio_type);

audio_defns.h

@@ -45,6 +45,7 @@ typedef struct audio_frame *audio_frame_p;
 #define AUDIO_UNKNOWN   0               // which is a reserved value
 #define AUDIO_ADTS      ADTS_AUDIO_STREAM_TYPE
 #define AUDIO_L2        MPEG2_AUDIO_STREAM_TYPE
+#define AUDIO_AC3       ATSC_DOLBY_AUDIO_STREAM_TYPE
 
 #define AUDIO_ADTS_MPEG2 0x100
 #define AUDIO_ADTS_MPEG4 0x101
@@ -54,6 +55,7 @@ typedef struct audio_frame *audio_frame_p;
                          (x)==AUDIO_ADTS_MPEG2?"ADTS-MPEG2": \
                          (x)==AUDIO_ADTS_MPEG4?"ADTS-MPEG4": \
                          (x)==AUDIO_L2        ?"MPEG2":      \
+                         (x)==AUDIO_AC3       ?"ATSC-AC3":   \
                          "???")
 
 

esmerge.c

@@ -55,6 +55,8 @@
 #define ADTS_SAMPLES_PER_FRAME  1024
 // For MPEG-1 audio layer 2, this is 1152
 #define L2_SAMPLES_PER_FRAME    1152
+// For AC-3 this is 256 * 6
+#define AC3_SAMPLES_PER_FRAME   (256 * 6)
 
 // ------------------------------------------------------------
 #define TEST_PTS_DTS 0
@@ -186,6 +188,9 @@ static int merge_with_avs(avs_context_p  video_context,
   case AUDIO_L2:
     prog_type[1] = MPEG2_AUDIO_STREAM_TYPE;
     break;
+  case AUDIO_AC3:
+    prog_type[1] = ATSC_DOLBY_AUDIO_STREAM_TYPE;
+    break;
   default:              // what else can we do?
     prog_type[1] = ADTS_AUDIO_STREAM_TYPE;
     break;
@@ -419,6 +424,9 @@ static int merge_with_h264(access_unit_context_p  video_context,
   case AUDIO_L2:
     prog_type[1] = MPEG2_AUDIO_STREAM_TYPE;
     break;
+  case AUDIO_AC3:
+    prog_type[1] = ATSC_DOLBY_AUDIO_STREAM_TYPE;
+    break;
   default:              // what else can we do?
     prog_type[1] = ADTS_AUDIO_STREAM_TYPE;
     break;
@@ -610,6 +618,7 @@ static void print_usage()
     "  -l2               The audio stream is MPEG layer 2 audio\n"
     "  -mp2adts          The audio stream is MPEG-2 style ADTS regardless of ID bit\n"
     "  -mp4adts          The audio stream is MPEG-4 style ADTS regardless of ID bit\n"
+    "  -ac3              The audio stream is Dolby AC-3 in ATSC\n"
     "\n"
     "  -patpmtfreq <f>    PAT and PMT will be inserted every <f> video frames. \n"
     "                     by default, f = 0 and PAT/PMT are inserted only at  \n"
@@ -618,8 +627,8 @@ static void print_usage()
     "Limitations\n"
     "===========\n"
     "For the moment, the video input must be H.264 or AVS, and the audio input\n"
-    "ADTS or MPEG layer 2. Also, the audio is assumed to have a constant\n"
+    "ADTS, AC-3 ATSC or MPEG layer 2. Also, the audio is assumed to have a\n"
-    "number of samples per frame.\n"
+    "constant number of samples per frame.\n"
     );
 }
 
@@ -712,6 +721,10 @@ int main(int argc, char **argv)
       {
         audio_type = AUDIO_L2;
       }
+      else if (!strcmp("-ac3", argv[ii]))
+      {
+        audio_type = AUDIO_AC3;
+      }
       else if (!strcmp("-h264",argv[ii]))
       {
         video_type = VIDEO_H264;
@@ -851,6 +864,9 @@ int main(int argc, char **argv)
   case AUDIO_L2:
     audio_samples_per_frame = L2_SAMPLES_PER_FRAME;
     break;
+  case AUDIO_AC3:
+    audio_samples_per_frame = AC3_SAMPLES_PER_FRAME;
+    break;
   default:              // hmm - or we could give up...
     audio_samples_per_frame = ADTS_SAMPLES_PER_FRAME;
     break;