Re: Usb Soundcard For Mac
Leigha Keplinger
If so, would it be possible to write a script that reads an input of a soundcard and bounces it to an output on the same card and, if this input is silent for more than ten minutes, reroutes it to the output?
I'm still kind of newish to linux, and I installed panon to find that it's not working. I found a command that fixes it, but I need to run sed -i 's/collections.Iterable/collections.abc.Iterable/g' *.py in the directory with my soundcard, and I don't know where that is.
This is the soundcard I would be using and this is the bluetooth receiver I want to power. I want to use a USB 2.0 Passive Hub (Power to the Hub comes from the connected port). Will a USB 2.0 port deliver enough power to the hub or do I need to use an USB 3.0 port or even an external power source?
If you're using your Helix as an interface for recording (or the Scarlet) your computer is treating that as it's soundcard (as it is right now, anyways) and the output level of the master volume control will raise or lower the level over EVERYTHING playing back through the Helix, including the playback from Reaper. The scarlet would function the same way without changing any settings (and that's how most soundcards work). If you have the analog outputs of the Helix connected to the Scarlet, turning up the guitar master volume will change the level going into Reaper, and may clip your inputs on the Scarlet. All of this is expected behavior. I'm not familiar with the Focusrite Software, but I would assume you can route/setup monitor mixes just like you would with a normal mixer, so you'd have to go into that software, and adjust the Helix levels to whichever output its is connected to. I would agree with Zolko, though. Pick one interface at a time to use, and make sure you have the proper settings in Reaper assigned.
When using my DAW, if I just play a track, the halos on Scarlett 1/2 light up so somehow the DAW is getting sent to the Helix and back into the Scarlett. I have my computer soundcard set to Scarlett playback and record.
His only problem is his insistence on using the Helix as his soundcard. With my setup, I get to use all of the Scarlett inputs (many more than Helix) AND use HX Edit AND re-amp, all at the same time with NO GRIEF EVER!!
I'm not insisting on using Helix as soundcard. I don't want to use Helix as soundcard. I've just sold a Radial Twin City AB Y splitter because I was getting exactly what I wanted before I went to the 2dnd gen Scarlett
Throw away ASIO4All. The dedicated drivers for either device are going to infinitely more stable. Or at least DISABLE HELIX in the Asio4All menu. You only want the Scarlet as your soundcard, and Helix will talk to Edit without being used as the soundcard.
The MOTIF XS8 has an inbuilt Steinberg Yamaha FW audio interface which serves as my main soundcard connection. I am shortly going to simplify my set-up, dispose of the MOTIF XS8 and switch to a MIDI keyboard/controller (Maybe a Novation Launchkey MK3 or similar) as my future day to day keyboard. This leaves me with a gap of a soundcard/audio interface.
Unlike my Helix (which you can also use as a pc soundcard) the pc sound is running through all of the FX and amp blocks (I'm guessing this is due to the reamping nature of the whole affair). Sure you have to rememebr to switch them off. But it makes adding reverb, pitch and EQ super easy (not as easy with Helix ). I'm currently listening to Youtube in a concert hall.
My Question or Issue
Hello Spotify community. I have a big issue with my app. Usually it didn't happen so often but now it happens everytime i open my laptop.
I'm using an external sound card (Focusrite Scarlett 2i4 2nd gen with all drivers updated!!!) and for about 1-2 month i experience this thing: the audio playback is slowed down, and there are cracks and pops throughout the entire song. sometimes this thing lasted for..10-20 seconds and then dissparears but now everytime i play a song those cracks and pops annoy the h*** out of me. I didn't watched the clock to see for how much these things happen but i can clearly tell you that it's more than 5 minutes. Right now i'm playing a 3 minute song and changed like 2 tracks and the problem persists. Even though i reconnect the usb cable and so there's no improvement, however if i unplugg the external soundcard, the sound coming from the laptop's soundcard is just fine. I will attach a video with my problem so you can understand what i'm talking about.
Thank you.
(So far i uninstalled spotify twice, reinstalled the audio drivers twice and also the windows is freshly installed with the HDD formated (i had a problem with the windows and i had to format the entire HDD like 1-2 months ago MAX)
*untill i managed to upload a file with the problem, the audio streaming is WORKING. and i didn't do nothing. luckily i managed to record the thing that's happening.
**ok so i can't shrink the mp4 file size, please make a 10MB limit for file uploads..
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The problem was present but not like this. it lasted first 10-20 seconds and then it was ok. Also there was one time when the problem was the same even without the soundcard. I have spotify for 6-7 months or so and it didn't happen to me until october or november. The soundcard doesn't make this kind of sound if i playback something from youtube or facebook or mp3 from my HDD. so i guess it just adjust it's buffer size automatically.
*i found the asio panel and the buffer was at maximum. 1024 at 44.1 khz. changed it to 512 and now there's no problem whatsoever. maybe the buffer size was too large for spotify?
Why do the 26mV and 12mV readings remain constant on the line out when I adjust the signal from my sine wave amplitude or change the volume and balance control of my PC soundcard? The voltmeter reading never changes. The voltage changes on the scope, but not on the voltmeter. What am I missing?
My second question has to do with the soundcard max voltage readings I am getting. When I boost the frequency or amplitude of the wave it saturates at 2 volts. I am guessing +2 and -2 is the max input/output of my soundcard? The research I have done would agree with this conclusion but I would appreciate some input! If this is the case why can't I get a 2 volt reading on the voltmeter instead of this micro volt jitter?
If the software is any good it should have a calibration setting which you can use to set things up correctly. This will probably involve feeding a signal of a known level into the card and telling the software what the level is, then it can work the rest out. Since most soundcards have a DC blocking capacitor you will need an AC signal for this.
Since audio hardware can change frequently when new audio devices areplugged in or out, it is a good idea to always retrieve new instancesof soundcard._Speaker and soundcard._Microphoneinstead of keeping old references around for a long time.
The internal soundcard is driven by the kernel module in:/lib/modules/4.9.59-v7+/kernel/sound/arm/snd-bcm2835.koUnless you are planning to use the audio jack (in 1) or HDMI (in 2),you should disable this kernel module. There are two methods to do this.
I have heard that you can quite happily run logic without a soundcard I onlt intend to use headphones and I work "inside the box" is there an advantage of using a soundcard other than outputs or will it improve sound, power ect ?
For the sake of precise terminology, yes, you do need an audio interface, which is the same thing as a soundcard, but usually soundcard is a term reserved for inexpensive cards inside a PC to play video games. Audio pros usually talk about an "audio interface".
I'm working on creating a simple soundcard with a simple codec called pcm5102a which is a part of the kernel source. My board is the STM32MP135F-DK. I2S seems to be enough for this purpose however if you think that SAI will work better, then I'll take advice on that happily as well.
The current configuration is as in the attached audio_test.zip with I2S1 enabled and SPI1 Clock Mux set to PLL3Q instead of the default PLL4P (all set in the STM32CubeMX). The configuration is inserted into the build with a new machine and it does build without any problems. It's only after the boot that aplay -l doesn't list any soundcards, moreover below message is visible in the dmesg output:
The input is provided on a BNC jack. I chose BNC because it's small andis easy to adapt to whatever you need (RCA, bananas, etc.). The outerconductor of the BNC is not grounded (unless you want it to be, by pressing the"GND" button). The input can also be connected to the"generator" output, to monitor the level of the signal going out ofthe soundcard.
Because of the input configuration, the negative side of the input must bewith +/-15V of ground (the soundcard ground). Any signal over about 10VRMS will cause the input amp to clip. So the negative side is reallyintended to be connected to ground at the DUT. You can successfullymeasure differential signals as long as they are low voltage, though,like a balanced line-level signal. Note however that the input impedanceis 100k resistance between the - input and ground, and 100k between the + inputand the - input. Not balanced...
Protection is provided by resistors and diode clamps. Both sides of theinput canwithstand up to 200V RMS referenced to soundcard ground, even if the range is set to 200mV. The input caps block DC up to 400V, again, on both the + and - inputs. Soyou could, for example, make a measurement referenced to B+ (as long as B+is under 400V, and has less than 10V ripple).
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