I am using staxrip version 1.3.1.6 and I have a Sony hx50v camera that creates MTS files. The audio is 2 channel ac3. Staxrip just cannot seem to do anything with the audio. I cannot figure out what I need to do to get it to get audio to show up in that bottom section. Here is the log, it sees it, but doesn't extract the audio. Any help would be great. Thanks.
Audio
ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Format settings, Endianness : Big
Codec ID : 129
Duration : 13s 536ms
Bit rate mode : Constant
Bit rate : 256 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 423 KiB (1%)
Text
ID : 4608 (0x1200)
Menu ID : 1 (0x1)
Format : PGS
Codec ID : 144
Duration : 12s 948ms
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Index LWLibav
------------------------------------------------------------
"D:\_Backup data 3\Download\_video processing\Staxrip\Apps\ffmpeg\ffmpeg.exe" -i "C:\_kill\videos hold\00264 temp files\00264.avs"
ffmpeg version N-74700-g628a73f Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.9.3 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --enable-decklink --enable-zlib
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 59.100 / 56. 59.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 39.100 / 5. 39.100
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 2.101 / 1. 2.101
libpostproc 53. 3.100 / 53. 3.100
Input #0, avisynth, from 'C:\_kill\videos hold\00264 temp files\00264.avs':
Duration: 00:00:13.51, start: 0.000000, bitrate: 0 kb/s
Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1920x1080, 59.94 fps, 59.94 tbr, 59.94 tbn, 59.94 tbc
At least one output file must be specified
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I may not be able to help you easily with the original files; ffmpeg seems to miss the audio stream here.
But try to remux the MTS to MKV using mkvmerge (or MKVtoolnix GUI); this should be handled more certainly. -
I notice that if I copy the filename and drop it into the audio path, it sees the acs and then can do everything fine. The problem is, I need it to do that automatically. I have a zillion files to process. If it could just batch these all, it would be perfect.
I don't think I can post a sample file on this forum. Is there a place I can post a sample file? -
I uploaded the file 00312.MTS let me know if that is there
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