But when analogue represents a continuous and interrupted signal irrespective of playback speedAnalog media isn't "continuous" as such, everything has a certain resolution, due to physical characteristics.
Analog-y in digital:
Why on earth spend 8" on 80 kilobytes of data when everyone above a certain age have had 1.44 megabytes fit on a much smaller area using the same principle? Because it is much more technically challenging to put the same signal on the same area (assuming you "read the surface", it is area).
Now fix the technology: area is proportional to the signal size you can write.
Audio playback is naturally constrained to 1 second of music per second (surprise!), so for a given tape or vinyl groove, lower speed means less area per second to carry the information, and less quality.
Thank for the answer, interesting
For Reaper : I set the Reaper's project sample rate (in Project Settings tab) according to the sample rate of the content. This way Reaper forces soundcard sr switch, if needed.
Indeed if a resampler dsp is used, there will be obviously no sr switch... but it's no more the originial content.
Will try with ASIO4all to see if it changes something.
I also tried with Winyl and observed the same behavior than Foobar (cutoffs).
I didn't compile my own Audacity to get ASIO support so I tested REAPER instead. Granted I'm a total newbie with the program but I didn't see any option there to alter sample rate on the fly. The rate set in preferences was always used for playback and audio is resampled as required.
I tested my Asus soundcard's native ASIO and ASIO4ALL with the integrated sound chip on the motherboard (using Windows' driver). With REAPER I always lost the first samples on both sound devices and I tested WASAPI too and the same happened there. It also didn't matter if I used 44.1 kHz/48 kHz test file or 44.1 kHz/48 kHz samples. All combinations produced same results. And I did disable the fade effects from the file and the mixer showed action, just nothing came out of speakers.
With foobar2000 the native Asus ASIO drivers ate beginning of the samples when samplerate switched at track change. With ASIO4ALL beginning wasn't missing on either sound device.
But the very end of the previous track gets cut off with both ASIO implementations when samplerate changes. Same thing also happens with WASAPI event. WASAPI event over here allows hearing the very first samples of the next track though.
WASAPI push doesn't cut off the last samples on track change but it on the other hand snips away the beginning of the next track.
It seems like sound API/driver is doing the decision whether to play old buffered samples when stream format changes and just ignore new data until then or to drop buffered samples and start playing the new stream. It can't do both without something adding a delay.
Best solution to this problem is of course to use the DirectSound output method and utilize the existing resampler in Windows mixer. That way no samples will be missing in any situation. Another option is to use a resampler DSP so that DAC doesn't need to reinitialize itself in the middle of the playback.
Last post by Porcus -
Hold shift while pasting to forcibly paste at the end of playlist. This is hinted about in the statusbar when mouse hovering over 'paste' so it's not that hidden.What about spending a menu item accessible with Shift + right mouse? Like, underneath Paste?
Thank you for your comment. Still I would be glad if simple "Ctrl+V" would paste new tracks always below selected item om playlist.Then one would have a choice to make: above or below? I guess that most who select the first ("first of many", obviously) would want them pasted on top?
Then it appears the url doesn't return an actual audio stream or playlist. I'm guessing it's actually a web page with a player on it? What is the address?
It's a web page with a player on it.
[Thanks a lot! This is the kind of starting point I'm looking for, and thanks for pointing me to the utils interface which for some reason I always fail to take into account. I still can't figure out a reasonable way to determine the character code, though. The combination of keycode and utils.IsKeyPressed is actually enough, but do I really have to go through all possible combinations one by one with a switch statement? Isn't there something like a keyboard mapping? Perhaps some array of values in the registry key?on_key_down returns standard ascii character codes. So values 65-90 correspond to A-Z (you need to check if VK_SHIFT is down for upper/lower though). If you want to output the key pressed to the screen you can do something like:
var str = String.fromCharCode(65); // str now contains 'A'
Thanks for screen reader improvements in beta 3.
Would it be possible to add something like Add to active DSPs / Remove from active DSPs to the DSP Manager context menus? I figured out that Enter and Delete works but it might be perhaps more user friendly for some users relying on keyboard-only operation.
Is there a kind Foobar developer in the vicinity ?
I've been experiencing this behavior for a long time now.
Library Tree (component, not script)
Active DSPs: Skip silence / SoundTouch / tempo shift
Liric show 3
One way sync
Tipical use is to create a playlist with mostly Library Tree or Facets or Quicksearch (no Albumlist and no Search)
or dragging files directly from a folder into a playlist.
Problem encountered only in playlist.
Sometimes, when pressing "play" button or even double clicking on a track,
won't start the track shown as selected but it starts a track even from another
playlist. I will check this better but it seems that the track started is the last one
played in the previous "session" of Foobar. I say session but in fact I never shut FB down
so it's always on. Many times it seems to happen when I go and use FB after some time
nothing has been played.
It happens regularly on two PCs with two similar FB installations and configs.
One on Seven and the other one on XP.
If you have any other hint that would help to find the culprit I'll be glad
to do some testing.
From now on I will try to pay attention to the actions taken right before this happens
so to be able to write them down.
Also the sound doesn't come from a point like regular tweeter speakers, it "shoots" from the whole surface of the panel, so I think (and others can correct me if I'm wrong) that volume should fall off with distance differently than a regular speaker.I'd guess that a speaker of the shape of a Martin-Logan might have the same properties as a more conventionally-driven "vertical line array" speaker like the ones made by Townshend, Scaena. or even the Grateful Dead's legendarily humungous 'Wall of Sound'. At least when it comes to volume/distance matters.