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.
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.
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
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.
I'm trying to set up playback of hi-res files using Foobar2000, but I'm having a really hard time figuring out what all the settings mean, and what options to choose.
I have a Sony CAS-1 DAC/amp, which is powering a pair of LS50's. It supports DSD, FLAC, and PCM up to 32-bit/192khz. Right now the best files I have are DSD 64/128 and WAV 24-bit/192khz. I have a couple 32-bit WAV files too, but they're 48khz. Unfortunately the CAS-1 doesn't support DSD playback beyond 2.8Mhz, so my DSD 128 files can't be played.
Here's a list of issues I've tried to research on my own but can't find answers to:
In Windows, I have the USB audio output of my Surface Pro set to 32-bit/192khz. Is that the right setting to be in? When I run foobar, will it automatically change the output based on the track, or is everything going to get converted to 32/192?
In Foobar, the default maximum sample rate is 88.2khz. Will I affect anything by changing it to 192khz?
I downloaded and installed a few Foobar components to support DSD playback. I have ASIO support, DSD Processor, DSDIFF Decoder, and Super Audio CD Decoder installed into Foobar. Are any of those redundant/unnecessary? I've tried playing around but can't figure it out.
Under preferences>playback>output, I have my output device set to "DSD: ASIO : Sony Audio Driver". I also have the choices of "ASIO: DSD Transcoder (DoP/Native)", "ASIO: Sony Audio Driver", "DS: Primary Audio Driver", and "DS: CAS-1 (Sony Audio)". What do all those settings mean, and do I have it in the right one? I don't want to click through them all and test because last time I tried to, it screwed with playback and I started getting error messages, no matter output I chose after.
Under preferences>playback>output, I have to choose between 8, 16, 24, and 32-bit for output data format. I have it set to 32-bit. Will that make Foobar automatically upconvert anything that's not already 32-bit, and if so does that affect sound quality? Also, does this affect DSD files?
Under preferences>tools, there's a section called "DSD processor" with a box labeled "Use DSD Processor" that I can select. When I select it, nothing changes. What does it do?
Under preferences>tools>SACD, there's an option labeled "output mode". The choices are "PCM", "DSD", and "PCM+DSD". What do each of these do, and what's the best one? Online I found instructions saying to choose "DSD", but does that convert everything to DSD? Does PCM+DSD allow for both codecs to be transmitted without any kind of conversion between the two?
Last question: under that same SACD section, there's another option labeled "DSD processor" with the options "DSD processor". and "none". What does this do?
This is especially frustrating because I have a background working in both tech and audio. If I don't know the answer to something, I know how to find it online. But this has got me beat. I just want my music getting to the DAC as untouched as possible. Thanks everyone.
I have an up-to-date Windows 10 and a multimedia cabled Logitech keyboard and have same issue with foobar2000 v1.4. I use the standard Win10 keyboard driver, not the Logitech driver.
Once I upgraded to v1.4, the play/pause/next/previous buttons worked initially. I also got display notification with song artist and title when I pressed next track button on keyboard. Never had that seen before.
But, next day, supposedly after PC reboot, the mentioned keyboard keys stopped working unless foobar2000 window was active. Also, the notification on display when changing track didn't show up anymore.
Now I downgraded to v1.3.20 and Logitech play/pause/next/previous buttons work again.
Logitech volume button did keep working on v1.4.