eac3to - audio conversion tool

[Thunderbolt8]

is there a way to losslessly change wave/pcm files from 176kHz to flac? when I use the switches -override and -192000 nothing happens, the output flac file still has a sample rate of 176kHz

I can use -ResampleTo192000 but then I get the message "Reducing depth from 64 to 24 bits..." so I dont know if the outcome can still be considered lossless.

[LigH]

I don't really understand your intentions ... FLAC is a lossless compressor for integer PCM samples without changing the attributes as long as they are supported. If it gets 176 kHz PCM as input, why should the FLAC compressed result have any other sampling rate than 176 kHz? There should be no reason that FLAC would support 192 kHz, but not 176 kHz.

If you wanted to resample 176 kHz to 192 kHz, this resampling won't be lossless. There will probably be a conversion using floating point values intermediately.

[Keiyakusha]

Thunderbolt8
It will not be lossless. It is similar to converting pcm to MP3 then to FLAC (but far less destructive) where result is a lossless format that contains data that is no longer lossless. And btw, if you started with 16bit pcm, your result will not only be lossy, but an upscale too (in terms of bitdepth, not only sample count)
That said, I have no idea whether flac supports 176kHz, but I don't really see why it wouldn't. If for some reason it does not have support for it, you better use a different format that can handle it.

[hello_hello]

According to the info here, the maximum bitdepth supported by flac is 32 bit. https://xiph.org/flac/faq.html#general__samples
It also states: FLAC supports linear sample rates from 1Hz - 655350Hz in 1Hz increments

Only I couldn't make a 32 bit flac file no matter what I did. I tried the command line and several GUIs and the best I could manage was an error message stating 32 bits per sample is unsupported.
Creating a 176kHz wave file was easy. Creating a 176kHz, 64 bit (float) wave file wasn't much harder (adding -full to the command line):

eac3to v3.29
command line: "C:\Program Files\MeGUI\tools\eac3to\eac3to.exe" "E:\test.mkv" 2:"D:\T2_Audio - English.wav" -full -resampleTo176400 -progressnumbers
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:01:18, 24p /1.001
1: h264/AVC, English, 928x696 24p /1.001 (4:3)
2: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
[a02] ac3, 48000, 2.0
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Resampling to 176.4kHz...
[a02] Writing WAV...
[a02] Creating file "D:\T2_Audio - English.wav"...
Video track 1 contains 1874 frames.
eac3to processing took 5 seconds.
Done.

But as soon as you tell eac3to to output a flac file (I tried different sample rates and it doesn't make any difference):

eac3to v3.29
command line: "C:\Program Files\MeGUI\tools\eac3to\eac3to.exe" "E:\test.mkv" 2:"D:\T2_Audio - English.flac" -full -resampleTo176400 -progressnumbers
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 0:01:18, 24p /1.001
1: h264/AVC, English, 928x696 24p /1.001 (4:3)
2: AC3, English, 2.0 channels, 192kbps, 48kHz, dialnorm: -27dB
[a02] ac3, 48000, 2.0
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Resampling to 176.4kHz...
[a02] Reducing depth from 64 to 24 bits...
[a02] Encoding FLAC with libFlac...
[a02] Creating file "D:\T2_Audio - English.flac"...
Video track 1 contains 1874 frames.
eac3to processing took 14 seconds.
Done.

Reducing the bitdepth to something flac will play with makes sense, but is there any way to encode a 32 bit flac file as the flac help documents suggests it can?

[LigH]

FLAC may be able to support up to 32 bit integer resolution by specification. But eac3to may limit the internal resolution to a sane amount of 24 bit:

Most PCM audio streams may have only 16 or at most 24 bit per sample integer resolution. If you have a lossy format based on frequency spectrum subbands, you will usually have encoded 32-bit floating point values, which have a mantissa precision of 24 bit (see IEEE floating point specs regarding "Single Precision Float"). Even if eac3to converts audio to "Double Precision Float" (64 bit overall, 53 bit mantissa), the original samples still had at most 24 bit precision (rather less in less-than-maximum-volume scenes), therefore it doesn't make sense to waste more than 24 bits after any conversion.

I doubt you will ever get your hands on PCM samples with true 32 bit integer resolution.

[madshi]

Originally Posted by Smithy

Terminator 2 Judgment Day Skynet Edition 1991 Blu-ray 1080p EUR VC-1 DTS-HD MA

eac3to v3.29 (dcadec)

M2TS, 2 video tracks, 9 audio tracks, 19 subtitle tracks, 2:17:19, 24p /1.001
6: DTS Master Audio, German, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
[a06] dts, 48000, 7.1
[a06] Extracting audio track number 6...
[a06] Decoding with libDcaDec DTS Decoder...
[a06] Writing WAVs...
[a06] The libDcaDec DTS Decoder reported the error "Bitstream navigation error" while decoding. <ERROR>
Aborted at file position 16039026688. <ERROR>

Can you provide a small sample with which I could reproduce the issue? Then I can report this to the dcadec developer.

[Smithy]

Originally Posted by madshi

Can you provide a small sample with which I could reproduce the issue? Then I can report this to the dcadec developer.

maybe, i will find and check the runtime of aborted position.

the demuxxed dtshd has other file aborted position. ^^

eac3to v3.29

DTS Master Audio, 7.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
dts, 48000, 7.1
Decoding with libDcaDec DTS Decoder...
Writing WAV...
Creating file "V:\00018.mpls_6ger.dtshd_.wav"...
The libDcaDec DTS Decoder reported the error "Bitstream navigation error" while decoding. <ERROR>
Aborted at file position 1368129536. <ERROR>

[Thunderbolt8]

Originally Posted by LigH

I don't really understand your intentions ... FLAC is a lossless compressor for integer PCM samples without changing the attributes as long as they are supported. If it gets 176 kHz PCM as input, why should the FLAC compressed result have any other sampling rate than 176 kHz? There should be no reason that FLAC would support 192 kHz, but not 176 kHz.

If you wanted to resample 176 kHz to 192 kHz, this resampling won't be lossless. There will probably be a conversion using floating point values intermediately.

I want to concert a DSD (.DFF) audio file losslessly to Flac, because Winamp cant play DSD and the wasapi Plugin cannot play flac files with 176 kHz (only 96 and 192; also no 32-bit flac files; and no, i dont want to change my audio player)

[madshi]

Originally Posted by Smithy

maybe, i will find and check the runtime of aborted position.

If all else fails, you can encrypt and upload the whole DTS file and PM me the download address. I can then cut a sample for the dcadec dev.

[nevcairiel]

Originally Posted by Thunderbolt8

I want to concert a DSD (.DFF) audio file losslessly to Flac, because Winamp cant play DSD and the wasapi Plugin cannot play flac files with 176 kHz (only 96 and 192; also no 32-bit flac files; and no, i dont want to change my audio player)

You cannot convert DSD to FLAC lossless. There is always going to be a loss when converting DSD to PCM, or vice-versa.

[Smithy]

Originally Posted by madshi

If all else fails, you can encrypt and upload the whole DTS file and PM me the download address. I can then cut a sample for the dcadec dev.

i found the position and here is 1 min sample.
http://workupload.com/file/5ppYHVxe

[sneaker_ger]

Originally Posted by LigH

FLAC may be able to support up to 32 bit integer resolution by specification. But eac3to may limit the internal resolution to a sane amount of 24 bit:

I think the libflac encoder eac3to uses is limited to 24 bit in the first place. Does an alternative encoder to that even exist?
But as you say: it's probably that way because it's sane.

[madshi]

Originally Posted by Smithy

i found the position and here is 1 min sample.
http://workupload.com/file/5ppYHVxe

Thanks. I've reported it to the dcadec dev.

[LigH]

Originally Posted by sneaker_ger

I think the libflac encoder eac3to uses is limited to 24 bit in the first place. Does an alternative encoder to that even exist?

Any independent flac.exe or libflac.dll based on official sources:

https://sourceforge.net/projects/flac/
http://www.rarewares.org/lossless.php

[sneaker_ger]

I meant an alternative encoder that encodes to FLAC format but is not based on libFLAC. (And that supports 32 bit encoding)

[LigH]

Please excuse the counter-question ... but: Would there be any reason to spend any time on re-programming an OpenSource software with a rather tolerant free license?

I don't know any source adoption as freely available as the reference implementation by Xiph.org yet.

[nevcairiel]

Originally Posted by LigH

Would there be any reason to spend any time on re-programming an OpenSource software with a rather tolerant free license?

Often its a good idea to have independent implementations, as it can drive new ideas and improvements. For FLAC, FFmpeg has an independent decoder and encoder, but it seems limited to 24-bit as well.

[hello_hello]

Originally Posted by LigH

FLAC may be able to support up to 32 bit integer resolution by specification. But eac3to may limit the internal resolution to a sane amount of 24 bit....

I doubt you will ever get your hands on PCM samples with true 32 bit integer resolution.

What's the definition of "true 32 bit integer resolution"?
I can make a 32 bit wave file easily enough with Audacity. I think. This is 32 bit integer isn't it?

General
CompleteName : D:\test.wav
Format : Wave
FileSize/String : 12.2 MiB
Duration/String : 9s 59ms
OverallBitRate_Mode/String : Constant
OverallBitRate/String : 11.3 Mbps

Audio
Format : PCM
Format_Settings_Endianness : Little
Format_Settings_Sign : Signed
CodecID : 1
Duration/String : 9s 59ms
BitRate_Mode/String : Constant
BitRate/String : 11.3 Mbps
Channel(s)/String : 2 channels
SamplingRate/String : 176.4 KHz
BitDepth/String : 32 bits
StreamSize/String : 12.2 MiB (100%)

I tried both versions of flac you linked to as well as the version on the flac website, and wherever version foobar2000 is using that doesn't seem to like 32 bit integer either.



Audacity is the only program I have installed that'll output a 32 bit integer wave file. The other programs such as foobar2000 seem to want to output a 32 bit float wave file.
You're right though, anything over 24 bit for a flac file would definitely be overkill. I was just curious to try it when I read Thunderbolt8's question.

I don't know much about DSD, but according to Wikipedia it's comparable to 20 bit, 96kHz PCM, so a 24 bit flac file should be quite adequate.

[LigH]

Originally Posted by hello_hello

What's the definition of "true 32 bit integer resolution"?
I can make a 32 bit wave file easily enough with Audacity. I think. This is 32 bit integer isn't it?

So you want to convert 24 bit PCM to 32 bit PCM ... You can stuff the 8 lsb with 0-bits. The result still has at most 24 bit precision.

I rather mean: I wonder if there is any hardware recording audio with up to 32 bit precision. But I doubt that there are many 32-bit ADC (analogue-digital convertors) available, as well as I doubt there are many microphones with a sensitivity required to record with 32 bit precision.

IMHO, there are physical and electronical limits which would make audio recording with 32 bit precision very improbable. And even if, the "noise carpet" on any realistic movie set (not in "deaf rooms", not for synthetic sounds) would probably be high enough to return a signal-to-noise ratio even below the 24 bit treshold (don't remember exactly where it was, 120 dB?).

So in most practical cases, 32 bit precision would be an illusion, lying on a big fluffy carpet of noise.

[ZMachine95]

hello guys and girls, I would like to convert some BDMV's folders. I would like to use eac3to to to get the correct mpls file and send all tracks on ffmpeg stdin and convert.

I have thought about using something like that..

eac3to J:\BDMV\ 1) stdout.mkv | ffmpeg.exe -hwaccel auto -y -i - -map 0:v:0 -c:v libx265 -crf 20.0 -preset veryfast -map 0:a:0 -c:a:0 libvorbis -b:a:0 192k -map 0:a:1 -c:a:1 libvorbis -b:a:1 192k -map 0:s:0 -c:s:0 copy -map 0:s:1 -c:s:1 copy -map 0:s:2 -c:s:2 copy "H:\output.mkv"

but it doesn't work. If I use only stdout.h264 the video track is piped to ffmpeg and converted.

What am I doing wrong? or there is a fast way to do that?... I don't actually have to use only ffmpeg and eac3to..

thanks guys