Last post by kode54 -
As I PMed to you, this sounds more like something that would be useful for mastering audio.
I have never tried Sonar in recent years, but I have used both Logic Pro X and Reaper. You may have better luck with the real time auditioning in Reaper?
What are your machine specs, if I may ask? That may go a long way to identifying whether it's even worth switching to a different DAW.
Rendering with tracked tools in a text or json based format, using either rendered MIDI with foo_midi or any other plugin, or using a MIDI with no notes as a CC system to apply controls to the audio streams you're rendering from, may be possible, but I haven't tried something that bold as far as projects go.
Again, no amount of upsampling will improve the audio quality, unless the hardware is incredibly terrible at playing lower sample rates, which is highly unlikely.
There is no grand extrapolator which can intelligently reproduce information that never existed in the original signal, or was lost due to recording at a lower sample rate or downsampling. A neural network could be trained to try to replicate lost information, using original and downsampled audio, but this sort of replication is not totally reliable, and has only really been trained and demonstrated on visual data, not really on audio. And considering how slow it is on just images, it would be insane to expect it to be usable on massive quantities of samples like you have with audio. Well, not entirely. Figure 12 million samples in one dimension instead of the multiple of two dimensions. Still, don't expect miracles like this to actually produce anything you can hear, if your original signal is already at least Redbook quality.
Sorry for my late reply.
I was coding with WASAPI in exclusive mode, and i've tried to play 24bit audio with it.
My code failed to work on my own computer (with realtek audio chip), but worked fine on my friend's computer with a XMOS audio device.
I just sent him Binary Exe for testing, not source code. You may say Realtek chips need word padding, yes, i understand.
But the tricky part is, i can hear the music from the noise, and it has been speed up, maybe 10% or somewhat.
To figure out the byte padding pattern, I even tried to fill the buffer with only 8bit data, and only left channel data.
However, no matter how hard i tried, the best result i got was the SPEED UP audio.
If it was just a byte padding or byte order problem, the out put sound SHOULDN'T be speed up, as each time you GetBuffer(), you can know the buffer size in SAMPLEs.
That is why i say Realtek seems having a buggy driver, however, maybe i'm still wrong in some parts.
Actually, I was coding a Emulator of a CPU (or you can say a Computer) that is designed by myself.
I use WASAPI to emulate the audio device, and it's quite important to use EVENT mode, since many real world audio chips works in this way (Interrupt).
On the other hand, exclusive mode gives a constant audio buffer length, which is also similar to real world audio chips.
(when using shared mode, you have to check the buffer length each time after call to GetBuffer().)
At the first time i come to the audio part, i tried WaveOut() and it has very large latency, for about half a second.
Half a second may sounds not bad, but i want the latency smaller than 30ms, meaning about 30Hz around or even higher, close to real world audio chips.
You can guess that i want the audio device IRQ as a timer, so that it would be quite straightforward to implement A/V synchronize when doing video playing back.
If the audio latency is large, i'll have to prepare another timer and the video may become jerky.
Now i've almost finished coding my Emulator and i have implemented simple audio/video playback on it.
For the audio part, i just expose a WAVEFORMATE struct to the Guest System in the form of PCI bus configuration space registers,
and just copy data from Guest System (DMA) to fill the audio buffer without padding/ordering.
As you may expect, it can't play 24bit audio properly on my computer, i just round the audio data to 16bit when i have to work with it.
Did you use something else than foobar2000 to ReplayGain scan the Opus files? Opus specs forbid the use of old ReplayGain tags as it has its own R128 gain tags. Header gain adjustment is also part of the specifications. Decoders are supposed to always apply the header gain and optionally R128 Gain from tags as an additional adjustment. Since not all players support tag based ReplayGain foobar2000 allows writing the desired RG info to header. That feature is supposed to give ReplayGain with Opus everywhere where the format can be decoded.
That said I get the same ReplayGained loudness with Opus with all header writing options when I use foobar2000 to do the tagging and playback.
Last post by eric.w -
Yes - click "Reply", then below the text box for composing your post there's "Add files by dragging & dropping"
Rolled back more, to v. 1.3.14.
The same situation.
Experimentally found out that Preferences : Advanced : Tagging : Opus : Header gain option was "Use Track Gain", switched to "Leave null". The playback is still as without RG info. Good news: scan runs with stable and correct result now.
UPD: switched back to v. 1.3.17. ReplayGain seems to function OK with Opus now. Remained "Leave null" option for mentioned parameter.
Head-spinning issue related to small preferences option.
Last post by Zip -
Is there anyway to embed a couple midi channels into a Wave file using fubar2000 your midi plugin?
Or to record midi info into the audio? Possibly a new fumat :-)
I do this sonar but hate their playlist view for live performance. (unstable and hard to read & few options)
All I want to do is run a few midi CCs over two channels in-sync with the audio.
Any ideas or suggestions would be greatly appreciated.
Version 1.3.15 does the same. I discovered that RG info being not read and Opus files are considered as files without RG info.
Hi, I'm having a couple of issues and am hoping someone might have a tip or two.
I've created a random pool to add a random album to a playlist, which seems to be a common use of this component. It was working well for a while, but now whenever I add an album it is pre-shuffled (see image).
It only happens in this context--in all other playlists and filters album tracks are in their proper numerical order.
Any idea why this might happen?
I've tried re-creating the pool, and I've also tried adjusting Playlist Attribute parameters to no avail (thinking it might be causing the issue).
I've attached images of my random pool and playlist attribute settings.
Is there a way to make the random album auto-play when its added to the playlist?