bugfix update to V2.3HEADmaster

3 files changed, 103 insertions, 68 deletions

diff --git a/Readme.txt b/Readme.txt
index c44a73f..164ecab 100755
--- a/Readme.txt
+++ b/Readme.txt
@@ -1,5 +1,5 @@
- 
-VDR-convert Version 2.2
+
+VDR-convert Version 2.3
 =======================
 vdr-convert is a set of tools to accurately transcode VDR1.x and VDR2.x TV recordings, including all valid streams - video, audio (including AC3/DTS 5.1), Audio Description (AD), and DVB subtitles - into a more compressed and accessible format, while maintaining perceived quality with good compatibility. H264 and AAC are the default codecs for the main streams, but H265 is available from V2 onwards
 
diff --git a/genindex/CVS/Entries b/genindex/CVS/Entries
index fce9f5c..ffdcf56 100644
--- a/genindex/CVS/Entries
+++ b/genindex/CVS/Entries
@@ -7,11 +7,11 @@
 /file.h/1.2/Thu Sep  1 15:37:42 2016//
 /genindex.c/1.2/Thu Sep  1 15:37:42 2016//
 /Makefile/1.2/Thu Sep  1 15:37:42 2016//
-/pes.c/1.2/Thu Sep  1 15:37:42 2016//
 /pes.h/1.2/Thu Sep  1 15:37:42 2016//
 /README/1.2/Thu Sep  1 15:03:50 2016//
 /ringbuffer.c/1.2/Thu Sep  1 15:37:42 2016//
 /ringbuffer.h/1.2/Thu Sep  1 15:37:42 2016//
 /tools.h/1.2/Thu Sep  1 15:37:42 2016//
-/version.h/1.2/Thu Sep  1 15:37:42 2016//
+/pes.c/1.4/Sat Dec 31 16:11:24 2016//
+/version.h/1.4/Sat Dec 31 16:11:24 2016//
 D
diff --git a/vdr-convert b/vdr-convert
index 9b47416..58dc2f1 100755
--- a/vdr-convert
+++ b/vdr-convert
@@ -72,6 +72,13 @@
 #------------------------------------------------------------------------------
 #
 # $Log: vdr-convert,v $
+# Revision 2.3  2019/04/03 13:02:04  richard
+# Fixed support for mono podcasts using HE-AAC
+# Support latest ffmpeg with more relevant/shorter service name+provider metadata
+# AAC streams: Silently use Fraunhofer (libfdk) if available in ffmpeg unless set with -a
+# Improved error reporting tolerance x264/5
+# Improved error reporting tolerance (subtitle stream) with foreign language (hard-subtitled) programmes
+#
 # Revision 2.2  2018/09/12 12:10:10  richard
 # Added --podcast option: podcasts stored in subdirectories, by Title
 # Podcast audio codec config: Can use (HE)AAC, MP3 or Opus
@@ -182,6 +189,7 @@ Alangsurround='eng'
 ADlang='eng'
 Slang='eng'
 framerate=25    # 25 (DVB) or 30 for ATSC
+PROVIDER="unknown"
 
 filesystem=190  # limit filename length to 256 allowing ~60 chars for EP, DOW etc
 # Check behaviour on your filesystem with unicode/UTF characters, may need adjustment
@@ -250,7 +258,7 @@ ext='ts'    # default file format extension
 # libsoxr supposed to be higher quality resampler (sounds a bit better, usu. requires custom ffmpeg build)
 # MKV supports slightly better metadata, LMS doesn't recognise by default.
 
-podcastaudioprofile="libfdk_aac -profile:a aac_he_v2 -b:a 48k -af aresample=resampler=soxr -ar 44100"
+podcastaudioprofile="libfdk_aac -profile:a aac_he_v2 -b:a 48k -af aresample=resampler=soxr -ar 44100"  # (V1 is forced for mono)
 # podcastaudioprofile="libmp3lame -b:a 80k -af aresample=resampler=soxr -ar 44100"  # MP3
 # podcastaudioprofile="libopus -b:a 48k"      # Opus - Can't downsample to 44.1
 
@@ -403,48 +411,52 @@ function map() {
 # v. quick functional testing only
 #		F_VIDEO="-c:v libx264 -preset ultrafast -vf scale=112:63"
 #		F_VIDEO="-c:v mpeg2video"
-      # end video
+    # end video
 	fi
-    if [ $maintain -eq 1 ] && [ $(echo $line |grep -i "audio") ]; then
-      mapst="$mapst B:$((10000 - $bitrate)):$stream_id"
-      AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$Alangstereo"
-      [ $(echo $ext |grep -i "mp3") ] && logit "Warning: audio stream copy may not work with $ext format"
-    # Multichannel audio - just copy to best preserve. Can be AAC 5.1. Always put multichannel ahead of others
-	elif [ $(echo $line |grep -i "audio" |grep -i "ac3\|5.1") ]; then
-      mapst="$mapst C:$((10000 - $bitrate)):$stream_id"
-      AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$Alangsurround"
-      [ $(echo $ext |grep -i "mp3") ] && logit "Fail: multichannel audio will not work with $ext format" "err" && quit 1
-	elif [ $(echo $line |grep -i "audio" |grep -i "aac") ]; then
-      # We don't normally use Opus/HE-AAC as barely better at bitrates we're interested for AV transcodes. Opus is really only 48k
-      # AAC (e.g from HD recording) best left as is. We can't always detect AAC 5.1 (e.g starts in continuity in stereo), sometimes not the bitrate
-      # Doesn't matter how long you probe for, may still miss 5.1 AAC. Only safe way is copy
-
-      # lower the risk - just copy the stream
-      if [ $keep -eq 1 ]; then
-        [ ! $(echo $acodec |grep -i "aac") ] && logit "Fail: unsupported request for non-aac codec with aac source, exiting" "err" && quit 1
-        mapst="$mapst D:$((10000 - $bitrate)):$stream_id"
+	if [ $(echo $line |grep -i "audio") ]; then
+      if [ $maintain -eq 1 ]; then
+        mapst="$mapst B:$((10000 - $bitrate)):$stream_id"
         AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$Alangstereo"
-      else
-        # Take the risk for single-use, still an outside chance it's multichannel and could fail or end up as stereo
-        audio_bitrate $bitrate $3
-        mapst="$mapst E:$((10000 - $outputbitrate)):$stream_id"
-        AUDIO[$stream_id]="-c:a:stream_op $acodec -b:a:stream_op $outputbitrate"k" -metadata:s:a:stream_op language=$Alangstereo"
-      fi
-	else
-      # MP2 usually, sometimes mp3. ffmpeg sometimes says "2 channels". 
-      # AD stream can actually be stereo, esp in HD, or "0 channels" - a null stream, which we drop
-      if [ $(echo $line |grep -i "audio" |grep -i "stereo\|2 channels") ]; then
+        [ $(echo $ext |grep -i "mp3") ] && logit "Warning: audio stream copy may not work with $ext format"
+      # Multichannel audio - just copy to best preserve. Can be AAC 5.1. Always put multichannel ahead of others
+      elif [ $(echo $line |grep -i "ac3\|5.1") ]; then
+        mapst="$mapst C:$((10000 - $bitrate)):$stream_id"
+        AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$Alangsurround"
+        [ $(echo $ext |grep -i "mp3") ] && logit "Fail: multichannel audio will not work with $ext format" "err" && quit 1
+      elif [ $(echo $line |grep -i "aac") ]; then
+        # We don't normally use Opus/HE-AAC as barely better at bitrates we're interested for AV transcodes. Opus is really only 48k
+        # AAC (e.g from HD recording) best left as is. We can't always detect AAC 5.1 (e.g starts in continuity in stereo), sometimes not the bitrate
+        # Doesn't matter how long you probe for, may still miss 5.1 AAC. Only safe way is copy.
+        # lower the risk - just copy the stream
+        if [ $keep -eq 1 ]; then
+          [ ! $(echo $acodec |grep -i "aac\|copy") ] && logit "Fail: unsupported request for non-aac codec with aac source, exiting" "err" && quit 1
+          mapst="$mapst D:$((10000 - $bitrate)):$stream_id"
+          AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$Alangstereo"
+        else
+          # Take the risk for single-use, still a significant chance it's multichannel and could fail or end up as stereo
+          # (In this case use option "-a copy" to force copy the stream)
+          audio_bitrate $bitrate $3
+          mapst="$mapst E:$((10000 - $outputbitrate)):$stream_id"
+          AUDIO[$stream_id]="-c:a:stream_op $acodec -b:a:stream_op $outputbitrate"k" -metadata:s:a:stream_op language=$Alangstereo"
+        fi
+      elif [ $(echo $line |grep -i "stereo\|2 channels") ]; then
+        # MP2 usually, sometimes mp3. ffmpeg sometimes says "2 channels". 
+        # AD stream can actually be stereo, esp in HD, or "0 channels" - a null stream, which we drop
         audio_bitrate $bitrate $3
         mapst="$mapst G:$((10000 - $outputbitrate)):$stream_id"
         AUDIO[$stream_id]="-c:a:stream_op $acodec -b:a:stream_op $outputbitrate"k" -metadata:s:a:stream_op language=$Alangstereo"
-      elif [ $(echo $line |grep -i "audio" |grep -i "mono") ]; then
+      elif [ $(echo $line |grep -i "mono\|1 channel") ]; then
         # Likely the AD stream, at low bitrate. 
         # !!!AAC good at low bitrates. But sparse audio gives Kodi/VLC etc trouble, as does MP3 - so copy MP2: 0.5% hit approx!
         mapst="$mapst J:$((10000 - $bitrate)):$stream_id"
         AUDIO[$stream_id]="-c:a:stream_op copy -metadata:s:a:stream_op language=$ADlang"
       fi
-      # end audio
-	fi
+    
+      # NOTE that HE-AAC V2 cannot (currently) handle mono! As podcast only applies to single stream recordings, this is safe
+      [ $(echo $line |grep -i "mono\|1 channel") ] && podcastaudioprofile=$(echo $podcastaudioprofile |sed -e "s/aac_he_v2/aac_he/")
+      
+    # end audio
+    fi
 
     # Skip any dvd_sub misidentified in old VDR recordings - e.g. if .vdr files passed unprocessed
     # Can be multiple streams.
@@ -499,6 +511,7 @@ function map() {
           [ $(echo $podcastaudioprofile | grep 'libmp3') ] && ext="mp3" 
           [ $(echo $podcastaudioprofile | grep 'libopus') ] && ext="opus"
         fi
+        #rewrite the audio mapping found above for podcasts
         F_AUDIO=$(echo $F_AUDIO | sed -e "s/\(-c:a:0\).*\(-metadata\)/-c:a:0 $podcastaudioprofile -metadata/")
         concatflags="-movflags +faststart" 	    # help any streaming
         keep=0      # and we not keeping it: VDR may not support HE-AAC V2
@@ -704,7 +717,9 @@ function transcode() {
               echo "Empty subtitles stream for $main, discarding"
             else
               # a few subs means properly suspect
-              delete=0 && logit "Warning: not deleting originals for $main due to suspect subtitle stream size - please check" "err"
+#              delete=0 && logit "Warning: not deleting originals for $main due to suspect subtitle stream size - please check" "err"
+              # no longer error: use later size comparison, ensuring no loss. Often happens in hard-subtitled foreign language films
+              echo
             fi
           fi
         fi
@@ -742,7 +757,7 @@ function transcode() {
 	  return 1
 	fi
     # Now re-test/compare after conversion in case ffmpeg lost some (still keep for examination in case really OK)...
-    # F_SUBS for VDR2 conversions, copyts for VDR1.x. Widened tolerance due to errors when re-transcoding
+    # F_SUBS for VDR2 conversions, copyts for VDR1.x. Widened (empirical) tolerance due to errors when re-transcoding
     if [ "$F_SUBS" ] || [ $copyts ]; then
       subslength $2 "/dev/null"
       [ $? -ne 0 ] && return 1
@@ -765,7 +780,7 @@ function transcode() {
 }
 
 #------------------------------------------------------------------------------
-# Basic QA of the output filesize/duration to make sure it's OK
+# Basic QA - duration to make sure it's OK
 # Return 1 for a fail, else 0
 
 function check_duration() {
@@ -775,11 +790,18 @@ function check_duration() {
     logit "Warning: $outfile, size $(($outputsize/$meg))M, is only $duration sec long, original $(($inputsize/$meg))M, with length detected as $program_duration sec" "err"
     return 1
   fi
+}
+
+#------------------------------------------------------------------------------
+# Basic QA - filesize
+# Return 1 for a fail, else 0
+
+function check_size() {
   # check for silly compression ratio, or larger file, heaven forbid!)
   if [ "$F_VIDEO" ]; then
     [ $vcodec -eq 262 ] && mult=3     # MPEG original
-    [ $vcodec -eq 264 ] && mult=9     # x264: 1/3 of original
-    [ $vcodec -eq 265 ] && mult=12    # x265 1/4 of original
+    [ $vcodec -eq 264 ] && mult=10    # x264: 30% of original
+    [ $vcodec -eq 265 ] && mult=14    # x265: 21% of original
     if [ $debug -eq 0 ] && [ $(($inputsize * 3)) -gt $(($outputsize * $mult)) -o $outputsize -gt $inputsize -a $maintain -eq 0 ]; then
       logit "Warning: Suspect size of $outfile, (new $(($outputsize/$meg))M vs. original $(($inputsize/$meg))M)" "err"
       # keep converted outfile, actually it may be OK
@@ -870,10 +892,10 @@ function escape_apostrophe() {
    echo $($1 | sed 's/"/\\\"/g')
 }
 
-# return key=value, escaped, but only if the value exists (alert missing ones)
+# return key=value, escaped, but only if the value exists (alert missing ones).
 function create_meta() {
 #   [ "$2" ] && echo "-metadata $1=\"$(echo $2|sed -e 's/[;\]//g'|sed "s/'/\\\'/g"|sed 's/"/\\\"/g')"\"||echo "NOTE: -metadata $1 is missing" 1>&2
-   [ "$2" ] && echo "-metadata $1=\"$(echo $2|sed -e "s/"/\\\'/g"|sed -e 's/[;\]//g'|sed "s/\"/'/g")"\"||echo "NOTE: -metadata $1 is missing" 1>&2
+  [ "$2" ] && echo "-metadata $1=\"$(echo $2|sed -e "s/"/\\\'/g" -e 's/[;\]//g' -e "s/[\`\"]/'/g")"\"||echo "NOTE: -metadata $1 is missing" 1>&2
 }
 
 function Repeat() {
@@ -927,6 +949,7 @@ while [ "$1" != "" ]; do
                                 top=$(echo "$1"|sed -e 's/[^0-9]//g')
                                 ;;
         -a | --acodec )         shift
+                                acodec_set=1
                                 acodec=$(echo "$1"|sed -e 's/[^a-z0-9_]//g')
                                 ;;
         -v | --vcodec )         shift
@@ -998,23 +1021,29 @@ done
 Repeat - 120 
 echo
 
-[ $vcodec -lt 264 ] || [ $vcodec -gt 265 ] && usage && exit 1
-[ $ext == "mp3" ] && acodec="libmp3lame"
+[ ! -d "$input" ] && logit "Recording directory '$input' doesn't exist!" && exit 1
 
 # ffmpeg exists?
 command -v $ffmpeg >/dev/null 2>&1 || { echo "$0 requires 'ffmpeg' but it's not in the path/config,  Aborting."; exit 1; }
 ffmpegbuild=$($ffmpeg 2>&1)  # get build options to check feature compatibility
 
-[ $vcodec -eq 264 ] && [ ! $(echo $ffmpegbuild | grep 'libx264') ] && logit "$0 requires ffmpeg built with at least libx264" && exit 1
-[ $vcodec -eq 265 ] && [ ! $(echo $ffmpegbuild | grep 'libx265') ] && logit "ffmpeg is not built with libx265" && exit 1
-[ $acodec = "libfdk_aac" ] && [ ! $(echo $ffmpegbuild | grep 'libfdk') ]  && logit "ffmpeg is not built with libfdk_aac for AAC" && exit 1
-[ $acodec = "libmp3lame" ] && [ ! $(echo $ffmpegbuild | grep 'libmp3lame') ]  && logit "ffmpeg is not built with libmp3lame for MP3" && exit 1
-
 # create deep probes
 ffmpeg="nice -n 19 "$ffmpeg" -y -hide_banner -nostats -probesize 250M -analyzeduration 600M -copytb 1"
 
-[ ! -d "$input" ] && logit "Recording directory '$input' doesn't exist!" && exit 1
-[ $dif ] && [ $dif -gt 1 ] && echo "deinterlacer config must be 0 or 1" && usage && exit 1
+# === video ===
+[ $vcodec -lt 264 ] || [ $vcodec -gt 265 ] && usage && exit 1
+[ $vcodec -eq 264 ] && [ ! $(echo $ffmpegbuild | grep 'libx264') ] && logit "$0 requires ffmpeg built with at least libx264" && exit 1
+[ $vcodec -eq 265 ] && [ ! $(echo $ffmpegbuild | grep 'libx265') ] && logit "ffmpeg is not built with libx265" && exit 1
+
+# === audio ===
+[ $ext = "mp3" ] && acodec="libmp3lame"
+# native aac and copy are built ins.
+[ $acodec != "aac" ] && [ $acodec != "copy" ] && [ ! $(echo $ffmpegbuild | grep -w $acodec) ] && logit "ffmpeg is not built with codec library '$acodec' - use exact library name" && exit 1
+ 
+# Silently use Fraunhofer in pref to built-in aac (if not explicitly set on cmd line)
+[ $acodec = "aac" ] && [ ! $acodec_set ] && [ $(echo $ffmpegbuild | grep 'libfdk') ] && acodec='libfdk_aac'
+# help Fraunhofer codec produce full(er) range - default cutoff 14kHz @low bitrates.
+[ $acodec = "libfdk_aac" ] && acodec="$acodec -cutoff 17000"
 
 if [ $keep -eq 1 ]; then
   command -v vdr >/dev/null 2>&1 || { echo "$0 requires 'vdr' but it's not in the path,  Aborting."; exit 1; }
@@ -1027,14 +1056,13 @@ if [ $xmit -gt 0 ]; then
   [ $xmit -gt 1 ] && { echo "$0 Cannot FTP and Redo! - Aborting."; exit 1; }
 fi
 
+[ $dif ] && [ $dif -gt 1 ] && echo "deinterlacer config must be 0 or 1" && usage && exit 1
+
 # *** Adjustments to interdependent parameters ***
 
 [ $ext == "mp4" ] || [ $ext == "avi" ] && [ $subs -eq 1 ] && subs=0 && logit "Warning: subs not supported in $ext format: discarded if present" "warn"
 [ $top -gt 0 ] && [ $maintain -eq 1 ] && echo "Can't crop without re-encoding" && exit 1
 
-# help Fraunhofer codec produce full(er) range - default cutoff 14kHz @low bitrates.
-[ $acodec == "libfdk_aac" ] && acodec="$acodec -cutoff 17000"
-
 if [ "$podcast" ]; then
   [ ! -d "$podcast" ] && logit "Podcast directory '"$podcast"' doesn't exist!" && exit 1
   outputdir=$podcast
@@ -1123,6 +1151,7 @@ do
   ID[2]=$(echo "$line" |grep "^E" |cut -f3 -d" " |awk '{print strftime("%F-%a_%H-%M",$1)}')
  
   #resolution. Just look for HD in station name - a bit simplistic, but only used for file naming
+  PROVIDER=$(echo "$line" |grep "^C" |cut -f2 -d" ")
   CHAN=$(echo "$line" |grep "^C" |cut -f3-7 -d" ")
   HD=$(echo "$CHAN" |grep -o 'HD')
   fr=$(echo "$line" |grep "^F"|sed 's/[^0-9]*//g')
@@ -1148,8 +1177,7 @@ do
   pod_genrenum=$(echo "$line" |grep "^G" |cut -f2 -d" ")
   pod_presenters=$(echo "$line" |grep "^D" |grep -oP '(?<=Presenters: ).*' |cut -f1 -d"|")   # rarely
   pod_actors=$(echo "$line" |grep "^D" |grep -oP '(?<=Actors: ).*' |cut -f1 -d"|")   # often
-  pod_desc=$(echo "$line" |grep -oP '(?<=^D ).*' |cut -f1 -d"|")  # almost always, sadly usu. freeform!
-  
+  pod_desc=$(echo "$line" |grep -oP '(?<=^D ).*' |cut -f1 -d"|")  # almost always, sadly usu. freeform! 
   lifetime=$(echo "$line" |grep "^L" |sed 's/[^0-9]*//g')
 done
 
@@ -1296,7 +1324,7 @@ done
 
 # Map the last output file
 # Why? Sometimes a stream actually has no content (e.g. AD or subs),
-# even if the stream notionally existed when probed, ffmpeg usually drops it silently if empty,
+# even if the stream notionally existed when probed, ffmpeg can drop it silently if actually empty,
 # resulting in a different map in the output file. The transcode function may also drop null streams
 # ffmpeg still needs a specific map to copy more than just "the best" single audio & video stream.
 # For VDR1.x conversions, this also fixes the output stream mapping order 
@@ -1352,7 +1380,9 @@ else
       logit "Cannot create directory $outputdir/$TITLE, falling back to $outputdir" "warn"
     else
       outputdir="$outputdir/$TITLE"
-    fi      
+    fi
+    # Title has description concatenated so it shows in LMS scrolling title display
+    LONGTITLE="$LONGTITLE: $pod_desc"
   fi 
   for ((i=0; i<${#formats[*]}; i++));
   do
@@ -1361,7 +1391,7 @@ else
   # if marks present, convert to an EDL file
   [ -f "$input/marks$VDRtype" -a $redo -eq 0 ] && edl "$input/marks$VDRtype" "$outputdir/$BASENAME"
 fi
-
+   
 if [ $MERGEFILES ]; then
   result=0; results=0
   # Do merge(s)
@@ -1371,12 +1401,14 @@ if [ $MERGEFILES ]; then
       # add tag parameters for all recordings now (not just podcasts)
       # A number of these are gross approximations as metadata is poor - e.g. Album artist
       # language a bit approximate: individual streams have language tags, but not always shown
-      # Title has description concatenated so it shows in LMS scrolling title display - as some podcasts
       # MP4 likes "date" not "year" (ffmpeg docs wrong atm)
-      # ts has no metadata, ffmpeg silently drops
+      # ts/m2ts has only service provider + name, ffmpeg silently drops the rest
+      # https://patchwork.ffmpeg.org/patch/12047/ limits these to 255 bytes
+
       metadata=" \
-      "$(create_meta "service_provider" $CHAN)"\
-      "$(create_meta "title" "$LONGTITLE: $pod_desc")"\
+      "$(create_meta "service_provider" $PROVIDER)"\
+      "$(create_meta "service_name" $CHAN)"\
+      "$(create_meta "title" $LONGTITLE)"\
       "$(create_meta "year" $pod_year)"\
       "$(create_meta "date" $pod_year)"\
       "$(create_meta "album" $pod_album)"\
@@ -1401,6 +1433,7 @@ if [ $MERGEFILES ]; then
       cmd="$ffmpeg -i concat:"\"$MERGEFILES"\" $metadata -c copy $concatflags $map "\"$outfile"\" 2>> $LOGFILE"
       echo; Repeat - 120
       logit "Creating output with $cmd" "info"
+#      echo $cmd > "command.txt"
       sh -c $cmd
       result=$?
       [ $result -eq 0 ] && msg="OK" || msg="Fail"
@@ -1413,10 +1446,12 @@ if [ $MERGEFILES ]; then
     fi
   done
   if [ $results -eq 0 ]; then
-    check_duration       # basic QA
+    check_duration       # basic QA only. Skip size as may have used higher Q which is not tracked. Duration detects most errors
     [ $? -ne 0 ] && cleanup && quit 1
     # Extensive QA testing only when keeping
     if [ $keep -eq 1 ]; then
+      check_size
+      [ $? -ne 0 ] && cleanup && quit 1
       # Save video, force a VDR reindex and check it was really successful before deleting original files
       # VDR2 will only recognise .ts's in the NEW format of directory name
       if [ $VDRtype ]; then
@@ -1448,7 +1483,7 @@ if [ $MERGEFILES ]; then
         done
         mv -n "$input/index" "$input/index.orig" || logit "could not save index" "warn"
       fi
-      # Only a single ts after transcoding, (VDR will only check the file having replaced the originals!)
+      # Only a single file after transcoding, (VDR will only check the file having replaced the originals!)
       mv -f "$outfile" "$RECDIR/00001.ts" || logit "could not move .ts" "err"
       # Neatness, but importantly a check that VDR will be reasonably happy to play the file - otherwise bail
       sh -c "vdr --genindex "\"$RECDIR"\""
@@ -1513,4 +1548,4 @@ fi
 # remove lockfile
 quit 0
 
-# --------- $Id: vdr-convert,v 2.2 2018/09/12 12:10:10 richard Exp $ ---------- END
+# --------- $Id: vdr-convert,v 2.3 2019/04/03 13:02:04 richard Exp $ ---------- END