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Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Recent Posts
1
FLAC / Re: Multithreading
Last post by ktf -
This is because flaccid doesn't support wav input yet, it treats the wav as raw and sees the wav header as samples. Only using FLAC input is recommended for now.
2
FLAC / Re: Multithreading
Last post by bennetng -
With the above in mind these settings might be reasonable for subset (again I haven't tested queue depth or gasc much, nor chunk with decent tweak/merge, some of these are more guess than knowledge). Seems reasonable to pair stronger flac settings with stronger flaccid settings, it could be tweaked many ways. In probably ascending difficulty

  • Quick
  • --mode gasc --blocksize-limit-lower 1536 --tweak 0 --merge 0 --analysis-comp 6 --output-comp 6
Probably far from optimal but it might be a good starting point. I'd consider the first peakset setting on the list as probably the most balanced.

cmd
Code: [Select]
H:\>flaccid --in phantom.wav --out phantom.flac --mode gasc --blocksize-limit-lower 1536 --tweak 0 --merge 0 --analysis-comp 6 --output-comp 6
phantom.wav     settings        mode(gasc);lax(0);analysis_comp(6);analysis_apod((null));output_comp(6);output_apod((null));tweak(0);merge(0);blocksize_limit_lower(1536);blocksize_limit_upper(4608)
effort  analysis(2.734);tweak(0.000);merge(0.000);output(0.000)  subtiming       analysis(0.00000);tweak(0.00000);merge(0.00000) size    518191375        cpu_time        62.12700

foo_bitcompare
Code: [Select]
Differences found in compared tracks; the tracks became identical after applying offset and truncating first/last samples.
Extra leading/trailing sections contained non-null samples.

Comparing:
"H:\phantom.flac"
"H:\phantom.wav"
Differences found: length mismatch - 1:40:31.333583 vs 1:40:31.333333, 265981811 vs 265981800 samples.
Compared 265981800 samples.
Discarded last 11 samples containing non-null values from the longer file.
Differences found within the compared range: 531100405 values, 0:00.000000 - 1:40:31.333311, peak: 1.631927 (+4.25 dBTP) at 0:40.642426, 1ch
Channel difference peaks: 1.631927 (+4.25 dBTP) 1.481720 (+3.42 dBTP)
File #1 peaks: 0.999969 (-0.00 dBTP) 1.000000 (0.00 dBTP)
File #2 peaks: 0.999969 (-0.00 dBTP) 1.000000 (0.00 dBTP)
Detected offset as -11 samples.

Comparing again with corrected offset...
Compared 265981800 samples, with offset of -11.
Discarded 11 leading samples containing non-null values from file #1.
No differences in decoded data found within the compared range.
Channel peaks: 0.999969 (-0.00 dBTP) 1.000000 (0.00 dBTP)
3
FLAC / Re: Decompressing FLAC files to WAVs
Last post by voxinbrain -
a classic case for people who have nothing to say on the merits of the "geeks press the newbie" series.
unfriendly to the forum - asking a question automatically becomes a beginner, and giving an answer is necessarily a geek.
Just because you know how to use a PC doesn't mean you invented it.
By the way, I'm from the Russian Federation, which means I'm hacking your PC right now.
5
CUETools / Re: how do I turn off .cue file being created?
Last post by c72578 -
@dpr could you please explain briefly, why you want to turn off creating the .cue file?
In the example above you show Encode, Mode: Tracks and Audio Output: Lossless.
Are there other cases, where you would like to turn it off?
6
General - (fb2k) / foobar200 and DSD track - shows "lossy"
Last post by Tuscany -
Hi all, first post here. Tried to do a bit of search but to no avail.

So, I stumbled last night on DSD ... I am still listening to only one track, but I am blown away by the difference. Still cannot believe my ears.
I have one question tho - I installed the plugin as per this article - https://help.nativedsd.com/en/articles/94982-foobar-2000-and-playing-dsf-audio-files

foobar2000 plays the file, reports 11k kbps, sounds amazing, HOWEVER I get the "lossy" rectangular lit up instead of "loseless". When I play regular flac, I get "loseless". Do I miss something or it is just the way foobar2000 interprets based on the file type/codec?
8
General A/V / Re: Blu-Ray and DVD Authoring on Ubuntu Linux using only FOSS
Last post by Octocontrabass -
I suppose some guide as to which bitrates actually work is also going to be helpful. I was assuming there's no limit aside from the top end in the spec, but apparently it's not continuously variable.
The available AC3 bitrates are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, and 640 kbps. All of these should work on Blu-ray.

Maybe there's some limit on the video bit rate choices too?
There's no limit other than the upper limit of 40Mbps.
9
General A/V / Re: Blu-Ray and DVD Authoring on Ubuntu Linux using only FOSS
Last post by CherylJosie -
Then there's this:

"Once again - AC-3 official for DVD Video is 448kbps however all Dolby decoders must decode 512kbps and 640kbps - this is part of the Dolby conformance testing so 448kbps may be seen as soft requirement not hard requirement - authoring software of course can be more or less picky on this."

https://forum.videohelp.com/threads/386092-Encoding-audio-in-DTS-in-multi-channels-with-ffmpeg#post2625252

I suppose some guide as to which bitrates actually work is also going to be helpful. I was assuming there's no limit aside from the top end in the spec, but apparently it's not continuously variable.

Maybe there's some limit on the video bit rate choices too?
10
General A/V / Re: Blu-Ray and DVD Authoring on Ubuntu Linux using only FOSS
Last post by Octocontrabass -
Any idea which of these is most likely to transcode with x264?
It uses the same decoders as ffmpeg, so it should handle all three equally well. However, I notice two of the three have 4:2:2 chroma subsampling, which is incorrect for Blu-ray. As far as I recall, x264 can fix this by itself, but your temporary lossless files will be smaller if you configure Kdenlive to output 4:2:0.

Any idea which audio encoder/frontend to use?
You can configure Kdenlive to render an audio-only file. AC3 with a reasonably high bitrate (320kbps?) should be fine for stereo audio. If you (or your potential future customers) are feeling paranoid, you can use uncompressed 16-bit 48kHz PCM instead, but that's a bit more work since your muxer won't be able to automatically detect what kind of audio it is.

Muxer?
Whatever you use to turn the encoded video and audio into a Blu-ray. You're using tsMuxer, but in theory you could switch to something else without changing the rest of your process.