If you run eac3to with the -demux flag it will run a deeper analysis of the sub titles. I ran the following command to extract the necessary tracks: eac3to "f:\Avatar (2009)\RED_BIRD_2D_WW" 1) -demux These are some of the most annoying to deal with. Let’s take a look at Avatar (2009) for these sub titles. If we inspect the sup files we’ll find that one is a normal sub title track while the other is a closed caption sub title track. Our output will look the same as before and we can drop these files into mkvmerge GUI. Using Underworld (2003) as an example we can see the audio.wav used to extract the audio into the wav container. So instead of converting it to FLAC we can just pop it into a wav container. This option isn’t always a solution for those of us that want to bitstream audio to our receiver. This will take a little longer to extract as it will encode the HD audio to FLAC during the process. That can be done during the eac3to extraction with the 3:audio.flac command. A lot of people choose to convert the LPCM audio track to FLAC which is a free lossless compression format. So the quality is great but it typically takes up a lot of space. Instead of using a licensed compression format like TrueHD or DTS-HD the LPCM track is a raw uncompressed track. Video:0kB audio:10336kB subtitle:0 global headers:0kB muxing overhead 0.099745%ĭuration: 00:01:00.00, start: 0.LPCM audio is the inexpensive way of getting HD audio onto Blu-ray discs. $ fmpeg -i InputFile.flac -y -acodec pcm_s16be OutputFile.flv Impossible to read with software players (tested with VLC and Foobar2000) Seen as LPCM_S16LE by FFmpeg (Little Endian !!!) With the help of FFmpeg forum I also tried to change the FFmpeg commandīut when asking to transcode to Big-Endian (pcm_16be) the conversion in a FLV container with "-acodec pcm_s16be" create a file which is: OutputFile.flv: Invalid data found when processing input Video:0kB audio:10336kB subtitle:0 global headers:0kB muxing overhead 0.000000% $ ffmpeg -i InputFile.flac -y -threads 4 -ar 44100 -ac 2 -f s16be OutputFile.flvįfmpeg version git-b821def Copyright (c) 2000-2014 the FFmpeg developersīuilt on 20:15:19 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)Ĭonfiguration: -disable-opencl -enable-gpl -enable-libfaac -enable-libmp3lame -enable-libopencore-amrnb -enable-libopencore-amrwb -enable-libtheora -enable-libvorbis -enable-libx264 -enable-nonfree -enable-postproc -enable-version3 -enable-x11grab -enable-librtmp -enable-libxvid -enable-libass -enable-libvpx My understanding of "network byte order" is "Big-Endian" (ie: LPCM_S16BE) According to DLNA specification, LPCM "denotes uncompressed audio data, using 16-bit signed representation in two’s-complement notation and network byte order". I cannot use another encoder (flac for example) because my DLNA media server (Serviio) has an internal use of FFmpeg I need to convert audio FLAC files to LPCM with FFmpeg but the result is an invalid file I think now it's better to use this mailing list I've started a tread on FFmpeg Support Forum: Next message: Impossile to convert pure audio files to LPCM (S16BE).Previous message: Questions about w3fdif deinterlacing.Impossile to convert pure audio files to LPCM (S16BE) pperroux at pperroux at Impossile to convert pure audio files to LPCM (S16BE)
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