FLAC is supposed to reduce the size of the audio source. For some reason though, whenever I try to use it to encode this .DTS audio source from my Blu-Ray, it is taking a 269MB .DTS file and converting it into a 629MB audio file.
I have tried all the different compression levels and the size only varies by a couple MB. It's still coming out about 630MB on average. Whats going on here? Why is this making it larger and not shrinking it like it is supposed to be doing?
I have tried 3 different softwares, and all of them are giving me these same results. I even tried updating FLAC to the newest release, and this is still happening.
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FLAC is supposed to reduce the size of the audio source.
If you take a compressed format like .ac3, .dts, .mp3,... decode it and then encode it to a lossless format like FLAC, file size naturally will increase.
=> There is no problem with FLAC. Just what you expected it to do is wrong.
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Exactly, and as a result it gives a file with a lower size because it compressed it better. Thus reducing the size of the source audio file like I said.
DTS is lossless audio already. I am working with a lossless audio source. FLAC is supposed to be compressing it. The size is not supposed to be increasing by 3x. Even if it were one of the other compressed formats, that is way too much of a size increase.
I just asked a friend if they would try it, and it came out smaller for them. They encode FLAC all the time because it comes out smaller than DTS.
My issue was my encoders were outputting 24-bit FLAC audio files when the source was only 16-bit. As soon as they are set to be 16-bit instead of 24-bit, the 3x file size is greatly reduced, and it comes out smaller than the DTS file. It was padding them or something, and making much larger when set for 24-bit.
There was a problem with it. The problem was my settings.Last edited by killerteengohan; 30th Oct 2022 at 08:18.
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Okay,... but DTS is not lossless. DTS-HD is,...
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That's what Blu-Ray audio is according to the information for it on the case. The file on the computer ends with .DTS regardless though. Its not gonna be "Audio.DTS-HD", or at least I have never seen that file extension before. That's why I called it .DTS.
I apologize if that was confusing.
Thanks for the answer! -
I apologize if that was confusing.users currently on my ignore list: deadrats, Stears555
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Would you happen to know the command line for FLAC to make it output 16-bit? I found this "-sample_fmt s16" and tried to use it, but no matter what I try, it will not work with MeGui.
I don't even know if thats the correct command line usage. The error I get just keeps saying "ERROR: --channels not allowed with --decode"
Audacity I can just choose 16-Bit in settings, but MeGui defaults to 24-Bits. I would use Audacity, but I have avisynth syncing involved that can't be used with Audacity. -
No clue about MeGui, but 'flac --explain' reports:
Code:=============================================================================== flac - Command-line FLAC encoder/decoder version 1.4.0 Copyright (C) 2000-2009 Josh Coalson Copyright (C) 2011-2022 Xiph.Org Foundation This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program 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 General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. =============================================================================== Usage: Encoding: flac [<general/encoding/format options>] [INPUTFILE [...]] Decoding: flac -d [<general/decoding/format options>] [FLACFILE [...]] Testing: flac -t [<general options>] [FLACFILE [...]] Analyzing: flac -a [<general/analysis options>] [FLACFILE [...]] Be sure to read the list of known bugs at: http://xiph.org/flac/documentation_bugs.html For encoding: The input file(s) may be a PCM WAVE, Wave64, RF64 file, AIFF (or uncompressed AIFF-C) file, or raw samples. The output file(s) will be in native FLAC or Ogg FLAC format For decoding, the reverse is true. A single INPUTFILE may be - for stdin. No INPUTFILE implies stdin. Use of stdin implies -c (write to stdout). Normally you should use: flac [options] -o outfilename or flac -d [options] -o outfilename instead of: flac [options] > outfilename or flac -d [options] > outfilename since the former allows flac to seek backwards to write the STREAMINFO or WAVE/AIFF header contents when necessary. general options: -v, --version Show the flac version number -h, --help Show basic usage a list of all options -H, --explain Show this screen -d, --decode Decode (the default behavior is to encode) -t, --test Same as -d except no decoded file is written -a, --analyze Same as -d except an analysis file is written -c, --stdout Write output to stdout -s, --silent Do not write runtime encode/decode statistics --totally-silent Do not print anything of any kind, including warnings or errors. The exit code will be the only way to determine successful completion. --no-utf8-convert Do not convert tags from local charset to UTF-8. This is useful for scripts, and setting tags in situations where the locale is wrong. This option must appear before any tag options! -w, --warnings-as-errors Treat all warnings as errors -f, --force Force overwriting of output files -o, --output-name=FILENAME Force the output file name; usually flac just changes the extension. May only be used when encoding a single file. May not be used in conjunction with --output-prefix. --output-prefix=STRING Prefix each output file name with the given STRING. This can be useful for encoding or decoding files to a different directory. Make sure if your STRING is a path name that it ends with a '/' slash. --delete-input-file Automatically delete the input file after a successful encode or decode. If there was an error (including a verify error) the input file is left intact. --preserve-modtime Output files have their timestamps/permissions set to match those of their inputs (this is default). Use --no-preserve-modtime to make output files have the current time and default permissions. --keep-foreign-metadata If encoding, save WAVE or AIFF non-audio chunks in FLAC metadata. If decoding, restore any saved non-audio chunks from FLAC metadata when writing the decoded file. Foreign metadata cannot be transcoded, e.g. WAVE chunks saved in a FLAC file cannot be restored when decoding to AIFF. Input and output must be regular files, not stdin/out. --keep-foreign-metadata-if-present As previous option, but do not throw an error in case no foreign metadata is found, the wrong kind of foreign metadata is found (on decoding) or if the foreign could not be parsed, i.e. all foreign metadata related errors are treated as warnings. --skip={#|mm:ss.ss} Skip the first # samples of each input file; can be used both for encoding and decoding. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second. --until={#|[+|-]mm:ss.ss} Stop at the given sample number for each input file. The given sample number is not included in the decoded output. The alternative form mm:ss.ss can be used to specify minutes, seconds, and fractions of a second. If a `+' sign is at the beginning, the --until point is relative to the --skip point. If a `-' sign is at the beginning, the --until point is relative to end of the audio. --ogg When encoding, generate Ogg FLAC output instead of native FLAC. Ogg FLAC streams are FLAC streams wrapped in an Ogg transport layer. The resulting file should have an '.oga' extension and will still be decodable by flac. When decoding, force the input to be treated as Ogg FLAC. This is useful when piping input from stdin or when the filename does not end in '.oga' or '.ogg'. --serial-number Serial number to use for the FLAC stream. When encoding and no serial number is given, flac uses a random one. If encoding to multiple files the serial number is incremented for each file. When decoding and no number is given, flac uses the serial number of the first page. analysis options: --residual-text Include residual signal in text output. This will make the file very big, much larger than even the decoded file. --residual-gnuplot Generate gnuplot files of residual distribution of each subframe decoding options: -F, --decode-through-errors By default flac stops decoding with an error and removes the partially decoded file if it encounters a bitstream error. With -F, errors are still printed but flac will continue decoding to completion. Note that errors may cause the decoded audio to be missing some samples or have silent sections. --cue=[#.#][-[#.#]] Set the beginning and ending cuepoints to decode. The optional first #.# is the track and index point at which decoding will start; the default is the beginning of the stream. The optional second #.# is the track and index point at which decoding will end; the default is the end of the stream. If the cuepoint does not exist, the closest one before it (for the start point) or after it (for the end point) will be used. The cuepoints are merely translated into sample numbers then used as --skip and --until. A CD track can always be cued by, for example, --cue=9.1-10.1 for track 9, even if the CD has no 10th track. encoding options: -V, --verify Verify a correct encoding by decoding the output in parallel and comparing to the original --lax Allow encoder to generate non-Subset files --ignore-chunk-sizes Ignore data chunk sizes in WAVE/AIFF files; useful when piping data from programs which generate bogus data chunk sizes. --sector-align Align encoding of multiple CD format WAVE files on sector boundaries. This option is DEPRECATED and may not exist in future versions of flac. shntool offers similar functionality. --replay-gain Calculate ReplayGain values and store them as FLAC tags. Title gains/peaks will be computed for each file, and an album gain/peak will be computed for all files. All input files must have the same resolution, sample rate, and number of channels. The sample rate must be one of 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, or 48 kHz. NOTE: this option may also leave a few extra bytes in the PADDING block. --cuesheet=FILENAME Import the given cuesheet file and store it in a CUESHEET metadata block. This option may only be used when encoding a single file. A seekpoint will be added for each index point in the cuesheet to the SEEKTABLE unless --no-cued-seekpoints is specified. --picture=SPECIFICATION Import a picture and store it in a PICTURE block. More than one --picture command can be specified. The SPECIFICATION can either be a simple filename for the picture file, or a complete specification whose parts are separated by | characters. Some parts may be left empty to invoke default values. Using a filename is shorthand for "||||FILE". The SPECIFICATION format is: [TYPE]|[MIME-TYPE]|[DESCRIPTION]|[WIDTHxHEIGHTxDEPTH[/COLORS]]|FILE TYPE is optional; it is a number from one of: 0: Other 1: 32x32 pixels 'file icon' (PNG only) 2: Other file icon 3: Cover (front) 4: Cover (back) 5: Leaflet page 6: Media (e.g. label side of CD) 7: Lead artist/lead performer/soloist 8: Artist/performer 9: Conductor 10: Band/Orchestra 11: Composer 12: Lyricist/text writer 13: Recording Location 14: During recording 15: During performance 16: Movie/video screen capture 17: A bright coloured fish 18: Illustration 19: Band/artist logotype 20: Publisher/Studio logotype The default is 3 (front cover). There may only be one picture each of type 1 and 2 in a file. MIME-TYPE is optional; if left blank, it will be detected from the file. For best compatibility with players, use pictures with MIME type image/jpeg or image/png. The MIME type can also be --> to mean that FILE is actually a URL to an image, though this use is discouraged. DESCRIPTION is optional; the default is an empty string The next part specifies the resolution and color information. If the MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave this empty and they can be detected from the file. Otherwise, you must specify the width in pixels, height in pixels, and color depth in bits-per-pixel. If the image has indexed colors you should also specify the number of colors used. FILE is the path to the picture file to be imported, or the URL if MIME type is --> -T, --tag=FIELD=VALUE Add a FLAC tag. Make sure to quote the comment if necessary. This option may appear more than once to add several comments. NOTE: all tags will be added to all encoded files. --tag-from-file=FIELD=FILENAME Like --tag, except FILENAME is a file whose contents will be read verbatim to set the tag value. The contents will be converted to UTF-8 from the local charset. This can be used to store a cuesheet in a tag (e.g. --tag-from-file="CUESHEET=image.cue"). Do not try to store binary data in tag fields! Use APPLICATION blocks for that. -S, --seekpoint={#|X|#x|#s} Include a point or points in a SEEKTABLE # : a specific sample number for a seek point X : a placeholder point (always goes at the end of the SEEKTABLE) #x : # evenly spaced seekpoints, the first being at sample 0 #s : a seekpoint every # seconds; # does not have to be a whole number You may use many -S options; the resulting SEEKTABLE will be the unique- ified union of all such values. With no -S options, flac defaults to '-S 10s'. Use -S- for no SEEKTABLE. Note: -S #x and -S #s will not work if the encoder can't determine the input size before starting. Note: if you use -S # and # is >= samples in the input, there will be either no seek point entered (if the input size is determinable before encoding starts) or a placeholder point (if input size is not determinable) -P, --padding=# Tell the encoder to write a PADDING metadata block of the given length (in bytes) after the STREAMINFO block. This is useful if you plan to tag the file later with an APPLICATION block; instead of having to rewrite the entire file later just to insert your block, you can write directly over the PADDING block. Note that the total length of the PADDING block will be 4 bytes longer than the length given because of the 4 metadata block header bytes. You can force no PADDING block at all to be written with --no-padding. The encoder writes a PADDING block of 8192 bytes by default, or 65536 bytes if the input audio is more than 20 minutes long. -b, --blocksize=# Specify the blocksize in samples; the default is 1152 for -l 0, else 4096; for subset streams this must be <= 4608 if the samplerate <= 48kHz, for subset streams with a higher samplerates it must be <= 16384. -0, --compression-level-0, --fast Synonymous with -l 0 -b 1152 -r 3 --no-mid-side -1, --compression-level-1 Synonymous with -l 0 -b 1152 -M -r 3 -2, --compression-level-2 Synonymous with -l 0 -b 1152 -m -r 3 -3, --compression-level-3 Synonymous with -l 6 -b 4096 -r 4 --no-mid-side -4, --compression-level-4 Synonymous with -l 8 -b 4096 -M -r 4 -5, --compression-level-5 Synonymous with -l 8 -b 4096 -m -r 5 -5 is the default setting -6, --compression-level-6 Synonymous with -l 8 -b 4096 -m -r 6 -A tukey(0.5) -A partial_tukey(2) -7, --compression-level-7 Synonymous with -l 12 -b 4096 -m -r 6 -A tukey(0.5) -A partial_tukey(2) -8, --compression-level-8, --best Synonymous with -l 12 -b 4096 -m -r 6 -A tukey(0.5) -A partial_tukey(2) -A punchout_tukey(3) -m, --mid-side Try mid-side coding for each frame (stereo only) -M, --adaptive-mid-side Adaptive mid-side coding for all frames (stereo only) -e, --exhaustive-model-search Do exhaustive model search (expensive!) -A, --apodization="function" Window audio data with given the function. The functions are: bartlett, bartlett_hann, blackman, blackman_harris_4term_92db, connes, flattop, gauss(STDDEV), hamming, hann, kaiser_bessel, nuttall, rectangle, triangle, tukey(P), welch, partial_tukey(n) punchout_tukey(n) and subdivide_tukey(n). More than one may be specified but encoding time is a multiple of the number of functions since they are each tried in turn. The encoder chooses suitable defaults in the absence of any -A options. -l, --max-lpc-order=# Max LPC order; 0 => only fixed predictors. Must be <= 12 for Subset streams if sample rate is <=48kHz. -p, --qlp-coeff-precision-search Do exhaustive search of LP coefficient quantization (expensive!); overrides -q; does nothing if using -l 0 -q, --qlp-coeff-precision=# Specify precision in bits of quantized linear-predictor coefficients; 0 => let encoder decide (the minimum is 5, the default is -q 0) -r, --rice-partition-order=[#,]# Set [min,]max residual partition order (# is 0 to 15 inclusive; min defaults to 0; the default is -r 0; above 4 does not usually help much) --limit-min-bitrate Limit minimum bitrate by not allowing frames consisting of only constant subframes. This ensures a bitrate of at least 1 bit/sample, for example 48kbit/s for 48kHz input. This is mostly beneficial for internet streaming. format options: --force-raw-format Force input (when encoding) or output (when decoding) to be treated as raw samples --force-aiff-format Force the decoder to output AIFF format. This option is not needed if the output filename (as set by -o) ends with .aif or .aiff; this option has no effect when encoding since input AIFF is auto-detected. --force-rf64-format Force the decoder to output RF64 format. This option is not needed if the output filename (as set by -o) ends with .rf64; this option has no effect when encoding since input RF64 is auto-detected. --force-wave64-format Force the decoder to output Wave64 format. This option is not needed if the output filename (as set by -o) ends with .w64; this option has no effect when encoding since input Wave64 is auto-detected. raw format options: --endian={big|little} Set byte order for samples --channels=# Number of channels --bps=# Number of bits per sample --sample-rate=# Sample rate in Hz --sign={signed|unsigned} Sign of samples (the default is signed) --input-size=# Size of the raw input in bytes. If you are encoding raw samples from stdin, you must set this option in order to be able to use --skip, --until, --cuesheet, or other options that need to know the size of the input beforehand. If the size given is greater than what is found in the input stream, the encoder will complain about an unexpected end-of-file. If the size given is less, samples will be truncated. negative options: --no-adaptive-mid-side --no-cued-seekpoints --no-decode-through-errors --no-delete-input-file --no-preserve-modtime --no-keep-foreign-metadata --no-exhaustive-model-search --no-lax --no-mid-side --no-ogg --no-padding --no-qlp-coeff-prec-search --no-residual-gnuplot --no-residual-text --no-ignore-chunk-sizes --no-sector-align --no-seektable --no-silent --no-force --no-verify --no-warnings-as-errors
users currently on my ignore list: deadrats, Stears555 -
It ran, but unfortunately that parameter added to the command line did not change anything. It still came out as 24-bit.
It says it ran this command line in the log.
--force --force-raw-format --endian=little --sign=signed -4 --bps=16 - -o "{0}" -
I think you may be right.
I noticed in log it's also running an avisynth script right before encoding and using the command line. This is the sync info.
Code:-[NoImage] LoadPlugin("C:\Users\Aiden\Desktop\MeGUI\tools\lsmash\LSMASHSource.dll") -[NoImage] LWLibavAudioSource("C:\Users\Aiden\Desktop\Encoding\Blu-Ray\Episode 1\T2_Audio - English.dts") -[NoImage] __film = last -[NoImage] __just_audio = __film -[NoImage] __blank = BlankClip(length=44834, fps=29.97) -[NoImage] __film = AudioDub(__blank, __film) -[NoImage] __t0 = __film.trim(0, 29) -[NoImage] __t1 = __film.trim(0, 2787) -[NoImage] __t2 = __film.trim(2794, 44834) -[NoImage] __t0 ++ __t1 ++ __t2 -[NoImage] AudioDubEx(__just_audio, last) -[NoImage] # detected channels: 6 -[NoImage] # detected channel positions: 3/2/0.1 -[NoImage] AudioBits(last)>24?ConvertAudioTo24bit(last):last -[NoImage] return last
An older version of MeGui however uses this in it's script.
Code:AudioBits(last)>24?ConvertAudioTo16bit(last):last
I think I can just make an avisynth script with the ConvertAudioTo16bit in it, and run the script instead of the built in encoder and settings. I will give that a try later when I have the time.
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