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Last post by Rollin -
In beta 7 playback in exclusive mode is glitchless for local files and streams with fading disabled and enabled on both Realtek and Dr DAC 2 DX. Although, it seems that few milliseconds are "eaten"on manual start of playback for Dr DAC 2 DX (or maybe there is always some glitch on the start of playback). Since it uses standard windows driver for USB audio devices, this should be reproducible with other similar devices, i guess. Noticeable on playback of attached fragment. First drum hit doesn't sound right with new exclusive mode. Playback with old WASAPI exclusive component doesn't have such artifact. I use windows 7.
Last post by josemescud -
I have installed Beta 7 and, for now, everything works excellent, both in my DAC Young MKIII and in my Marshall speaker as in my laptop. I will report in the next few days if there are problems.
Last post by eagleray -
Peter, thanks for the answers. The short buffer has been what it took to get my Xmos USB interface devices to work with event previously. I'm not having the rate switch issue, but I resample everything to 96 khz so I can use the Convolver.
If you want to traverse subdirectories, use FOR /R in place of FOR. If you want to put it in a clickable file, then copy the following to into Notepad and save it as e.g. "aac-remux-to-aacdotm4a.bat" in the folder where your .aac files are
FOR /R %%f IN (*.aac) DO ffmpeg -i "%f" -map 0 -c copy "%%f.m4a"
I omitted the "~n" and so the result isn't that pretty: from file.aac it will produce file.aac.m4a, not file.m4a. But if you do this to folders where there already are .m4a files, then you can use fb2k to search up all ".aac" and you will get both the .aac and the .aac.m4a files and can verify that they are the same before selecting the .aac and deleting them. Then you select the .aac.m4a and rename them to .m4a, and next time you are doing this, all the .aac.m4a will be freshly generated.
Edit: The "-map 0" shouldn't be necessary here. It is for files that have more than one video stream or more than one audio stream. Say, if you want to make mkv out of some video which has selectable audio language.
And I am a bit curious about who uses raw .aac for distributing music :-o
Beta 7 posted, with entirely new implementation of WASAPI Exclusive.
I am seeing a regression compared to beta 6 just using Realtek sound from my mobo. Whenever the player proceeds from a 24bit/48kHz file to 16bit/44.1kHz file, or vice versa, foobar2000 plays extremely choppily until I stop playback and manually open the next file.
This isn't "automated" per se but it's the next best thing; the following command line executed within a folder of *.aac files will wrap them into *.m4a files, which Foobar can then seek, tag, etc. The original *.aac files will be kept. You need to grab ffmpeg.exe if you don't have it (https://www.gyan.dev/ffmpeg/builds/ffmpeg-release-full.7z), and change the path in the command line below to reflect the location of your own ffmpeg.exe:
Last post by ktf -
The last few days I've been busy working on this, and there is some progress. Sadly, the code is very slow, and I don't think this will improve much. This improvement fits a certain niche of FLAC users who want maximum compression within the FLAC format and don't care how long encoding takes, perhaps reencoding in the background.
Attached you'll a 64-bit Windows executable compiled by MinGW with -march=native on a Intel Kaby Lake-R processor and a PDF with a graph of my results. I haven't used -march before, but if I understand correctly, this code should run on fairly recent 64-bit CPUs with AVX2 and FMA3. To make testing a little easier, I have added a new preset, -9, which uses the new code.
The following graph shows the results of encoding with (from left to right) setting -5, -8, -8e, -8ep and finally -9.
The new preset is -8e + a new 'apodization function', irls(2 11). It is technically not an apodization, but in this way, the integration into the FLAC tool is pretty clean. The function has two parameters, the number of iterations per LPC order, and the number of orders. So, with irls(2 11), the encoder does two iterations at order 2, two iterations at order 3, all the way to 2 iterations at order 12. Sadly, there is still something wrong in the code, so I would recommend not using anything else then 11 order at this time. Using more iterations is very well possible, and gives a little gain at the cost of much slowdown.
Last post by Peter -
Event mode with short buffer and super-simple lock-less data feed loop, strictly following Microsoft sample code behaviors.
Some quirks still exist (I wrote entirely new code yesterday....) but I think this is better than what we've been doing until now, hopefully I can get rid of all the options, making the defaults usable for everyone.
Last post by audiomonkey248 -
Thank you! yes for my own collection seems a good solution but what being able to seek to whatever aac file you throw at in other contexts? Also can conversion be automated to m4a? say monitored in real time? (in my ignorance this could a bad idea) but so I don't have to manually convert aac files to m4a?