Last post by andy o -
So something I've been annoyed about, that I haven't really heard anyone mention about this trend of "truly wireless" BT headsets, is that the mic(s) is/are pretty much on your ear(s). My thinking is that there really is no substitute for good mic placement even in headsets, and something like the wired Apple earpods' mic sounds a LOT better in calls, no matter how fancy the noise reduction algorithms and quantity of mics there are in the BT headset. And it's not just wired headsets, but also Bluetooth headsets which have wires and inline mics.
My experience is with a few Sony BT headsets. Even something as fancy as the WI-1000X, which has a huge neckband where they could easily have put a mic on, uses the noise-cancelling mics on the ears for calls. And it sounds awful. I guess if you speak loudly (and annoyingly to other people) it will work well enough.
Has anyone tested any of these BT headsets that actually don't compromise much on mic audio quality? I'm curious about how the Airpods fare on this, compared to wired or inline mics.
I don't know what's the reason to not have foobar2000 not have a function to undo tag.
That's easy, most people don't have issues with the problem you're describing and/or have preventive measures in place to help with such problems (backups). This seems to be a niche fringe request imho and considering the response in this thread you shouldn't expect anyone to jump in and make something and dedicate their time to it.
Last post by fohrums -
I'm surprised this hasn't been brought up enough. I would love this feature too even if that means a component based method of doing-so. I'm with wcs13 when dealing with many tracks it's impossible to not make a mistake. Whether the mistake was you thought the select-all or CTRL-A function for the current playlist is the album list you wanted to re-edit or it was your 50,000 track library and all of them now have the same Artist Name = YIKES!
I don't know what's the reason to not have foobar2000 not have a function to undo tag changes especially with a great history support for that matter, too. Again, Foobar2000 needs this safety function feature and not more anymore aesthetic ones.
Last post by rutra80 -
Nothing is perfect, FLAC predictors aren't either. There surely exist data patterns that you can add to sound file, which don't add actual meaningful information, but will cause FLAC files to bloat because its predictors won't handle them efficiently. FLAC isn't a format that aims at maximum compression.
Last post by kaiser.lima -
My experience in upload to youtube shows me that if you use V1 HE-AAC, or >190 LE-AAC, for example, youtube will convert to 120k LE-AAC and its crops the sound above 16kHz . The best result I observed in streams sites was upload mp3lame VBR (0) audio streams at 48100 Hz encoded inside the H.264 video, them the server choose the better compress method, like done in .wav files. Some youtube accounts appear have special permissions that allows the 190k AAC ( mp4a.40.2@192k), but its apear to be automatic based in popularity (like the VEVO i.e.). The mp3lame VBR (0) at least "brings" the sound until 20 kHz to youtube server, i think, in "sane" internet bandwidth . HE-AAC appear to have problems in youtube because the codec licences related at this format. LE-AAC are 99% converted to poor 120k.
Last post by kaiser.lima -
I use ffmpeg with aac_fdk from Zeranoe's FFmpeg Builds Home Page: <http://ffmpeg.zeranoe.com/builds/> I found better result with this bat command, bring the files you want to the flie .bat you create, the files wil place C:Temp in this example.
------------------ @ECHO ON
FOR %%A IN (%*) DO ( "C:\Program Files\ffmpeg-20151003-git-061b67f-win32-shared\bin\ffmpeg" -i %%A -vn -c:a libfdk_aac -profile:a aac_he -ar 88200 -b:a 160k -y "C:\Temp\%%~nA.m4a" ) --------- -The result is a 160 kbps he-aac file that sounds equivalent at 320 kbps lame mp3 (saved at C:\Temp in this example) -if the original file have 48 kHz the result are yet better, you can use a video file i. e. -Original studio editions (masters) when done at 60, 88 kHz have good results yet in 20-30 KHz audio spectrum! (observed in Adobe Audition audiograms and some mastering i done). Good to producers! -the "ffmpeg-20151003-git-061b67f-win32-shared" part must be the folder name of bin/ffmpeg.exe that you have after download and place the program in "Program Files" folder, it may change in your version! -some audio systems (or old tv media boxes) can't handle the HE part or files at 88,2 kHz, if you have them, sell it!!
If you're fine with that bitrate you may as well just use a time domain subband codec like MusePack or hell, even MP2, and enjoy the perfect temporal resolution. Throwing more bits at a transform codec doesn't do much to fix their fundamental shortcomings beyond 192 kbps.
Still does not change the fact that i love to convert with Opus - it has become my favorite Open Source codec. Flawless in every kind of ways
Last post by tedsmith -
When in foo_input_sacd's Output Mode of "DSD+PCM" you can set the "PCM Samplerate"to any of the options just fine. They will all work essentially identically for most systems. They don't affect the bits going to the DAC so don't worry about it. (They matter more when you convert DSD to PCM for your DAC...) Similarly the DSD2PCM can be any filter you like when you are using "DSD+PCM".
From what I can understand, by putting output in DSD + PCM, Foobar will send my DAC a pure DSD stream when playing DSD files, but convert a PCM version to use internally for visualizations. But then I have the option of selecting a samplerate which doesn't align with with 192 without downconversion. I have 44.1, 88.2, 176.4, and 352.8.