I cannot record audio using monitor source of sink devices,from 2 to 3 days.I have reinstalled Pulseaudio, but the problem remains. I am using ubuntu 12.04 with default pulse audio. few day ago, i had same problem but I reinstalled ubuntu so I overcame problem but now same problem...??
from my point of view, Monitor of internal audio does not seem to get any signal.because
i check Pulse Audio Volume Control (pavucontrol), in which volume bar does not shown volume level in playtab and same case in output Devices tab.However, I can hear audio,and the pavucontrol Play tab shows the name of the applications which is running.
suggest any way to overcome this problem, because my application need audio recording from speaker(from context of pulse audio from sink device).
Thanks...
I got the solution, it was a simple case of the monitor being muted. In pavucontrol go to input devices, then in the show button at the bottom switch it to All input devices. I believe it's normally set to all except monitors, so the monitor doesn't show up. In my case it was this monitor that was muted, but I could still hear sound because the internal audio wasn't muted. sorry for pasting here but lets Hope it helps someone....
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I'm using OpenMusic for the first time on Linux Mint 19.2; and I'm having issues getting any sound out of it from MIDI.
I have it set to Pulse Audio for output, and the PA Volume Control is open and above-other-windows, so I can see whether there's an output to it even if it's silenced. It's not muted, it's just dead silent. When I press the Play button on a note or measure, I still get nothing.
I've set MIDI out to timidity, and verified that it's set to use Pulse Audio. Am I missing something here, like a library or a sound font? How do I change that?
To get the obvious out of the way, my speakers are plugged in and on with the sound up; and other software has no issues with playback. I can play audio if it's from a file like an Ogg Vorbis, so this is likely MIDI related.
I have an audio coming from a radio transceiver on my sound card's microphone input. What i want to make is a simple software-based parrot repeater using Linux CLI tools like the sox suite and arecord. For it to work, i think a flow similar to the following must take place:
The audio that comes on the microphone subdevice is getting recorded in a buffer (file or RAM-based)
When the buffer stops filling (audio stopped), start playing it's content on the audio output device (it is connected to the radio's microphone input)
When it's over, empty the buffer and start expecting step 1 to occur again
I'm looking for an elegant way to implement the logic behind step 2. Is there a CLI tool that i can use for that, so i can pipe the microphone audio taken with arecord to it and play the output of the buffer with sox?
Try looking at this. I did this on a raspberry pi a little while ago, only I made a voice changer.
https://www.instructables.com/Halloween-Voice-Changer-With-Raspberry-Pi/
Basically, play "|rec --buffer 2048 -d" takes recorded sound and puts it in a buffer that is passed in 4096 bit (byte?) chunks to play. -d stands for duration, and if left blank defaults to 0, and will run until killed. If you want to play with the options, there is some helpful info in the links.
Good luck with your project!
The situation goes a little something like this:
I am programming Xcode whilst concurrently listening to music on my Bluetooth headphones... you know to block out the world.
Then, I go to launch my app in the iOS simulator and BOOM all of a sudden my crystal clear music becomes garbled and super low quality like it is playing in a bathtub 2 blocks away... in the 1940s.
Note: the quality deterioration does NOT occur if I am playing music on my laptop or cinema display and I launch the sim. It seems to be exclusively a Sim -> Bluetooth issue.
The problem is more than just annoying. Because often after stopping the simulator the crappy bathtub quality music continues. To fix it I have to open sound preferences in OSX and briefly toggle back to my laptop sound and then back to my Bluetooth headphones.
This is a big deal because I launch the simulator 50x a day and have to do this toggle thing every time as well as suffer listening to 40s era mono ham radio quality music.
For your information, the headphones I am using are Plantronics BackBeat Pro and I am up to date on firmware. I am on OSX 10.11.4 and Xcode 7.3... but this problem has persisted through all versions for 2+ years now. Can you save me from the 1940s?
I've managed to fix it, and it actually seems to be a microphone issue. Go to System Preferences -> Sound, select the Input tab and set Internal Microphone as the input (mine was set with my headphones').
Crappy sound goes way after that =)
EDIT (May 30 2018):
I've found out an easier way to do the same as above. Instead of opening the System Preferences, you can just go to the Mac OSX toolbar, press Option (alt) + click on the sound icon and then select "Internal Microphone" from the "Input Device" list. Print screen as follows.
If you're using Xcode 9 or higher, you can set a default audio input and output for the simulator. This can be done by launching the simulator from Xcode and navigating to I/O > Audio Input within the menu bar and selecting Internal Microphone. This solution will save your audio preference so you won't have to change it on every launch.
On Simulator, Select;
I/O -> Audio Input -> Macbook [Pro]
Done.
Seems like years of suffering are finally over, Xcode 12 Beta Release Notes:
Simulator defaults to the internal microphone unless you explicitly choose a different audio source. This avoids triggering phone call mode on Bluetooth headsets which degrades audio quality while listening to music. (59338925, 59803381)
You can also switch to Mac's internal mic in System Preferences -> Sound, that's how I usually fix this bug (I have Sony Wh-1000XM3)
I am using LENEVO G500 Laptop and my sound card has support for Dolby Advanced Audio v2 that works nicely in Windows OS (i.e. Windows 7, 8 and 8.1). However I have failed to enable Dolby sound effect in my Linux OS (have tried it with Linux Mint 17, Fedora 20).
Does anyone have an idea which linux version has support for this feature or how I can enable in a linux OS.
I would appreciate if you could direct me to the right direction.
Thanks.
I've googled out a very good advise on forums that helps me to achieve a Dolby like sound on my Kubuntu 19.04 with Lenovo g780.
Install PulseEffects https://github.com/wwmm/pulseeffects
(repos with deb files are here: https://github.com/wwmm/pulseeffects/wiki/Package-Repositories#debian--ubuntu)
Restart the user session or reboot after this, because PulseAudio will be upgraded, and it may cause problems if you don't restart.
Run PulseEffects and close it. It'll create all settings dirs on first launch. They required for next step.
Install PulseEffects-Presets from here: https://github.com/JackHack96/PulseEffects-Presets
(I've used the suggested script that automatically downloads them to PulseEffects import dirs, it will require flatpak that could be got from repos with sudo apt install flatpak)
Launch PulseEffects again. Select Convolver. Enable it. Click on wave button. You'll see a list of presets. Enable:
Dolby ATMOS ((128K MP3)) 1.Default.irs
Close the dialog and that's it. You can toggle Convolver in PulseEffects on and off while playng music to compare results. You may play with other presets as well.
For improving sound on a notebook or a tablet PulseEffects help pages come with a tuorial about how to achieve this.
App can be minimized to tray on GTK-enabled desktops with an additional application: https://github.com/boomshop/pulseffectstray
It's better enable autostart in app settings (it will copy it's desktop file to ~/.config/autostart with --gapplication-service command line. So next time start without GUI).
It is possible to get reasonably close to the Dolby Advanced Audio output on Linux.
TLDR:
Record the result of playing a -0dBFS impulse in Windows with all effects enabled. Save that as a wav file and use it as input to the PulseEffects Convolver.
Step-by-step:
Install Audacity in Windows.
Configure Audacity to use WASAPI
Select the loopback device as input
Select your laptop speakers as output, making sure all Dolby Advanced Audio effects are enabled.
Start recording
Play an impulse audio file (you might need to do this twice, Audacity often doesn't pick up the first impulse.
Zoom in and select the area around the recorded impulse (see screenshot below)
Export the selection as a WAV file and change the extension to irs
Import this irs file into the Convolver.
Some notes:
Audacity isn't required, presumably any software capable of recording from the output device will be fine.
To avoid any changes introduced by sample rate conversion, set the sample rate of the output and input devices in Windows to be the same.
When recording, in Step 5, Audacity would not record unless audio was playing. This is probably due to using WASAPI. Just start recording, play the impulse and if you don't see it in the recording output as a single spike, play it again.
The screenshot is quite heavily zoomed in so that you can see the area where there is data. When selecting what to export, try to make sure the selection is roughly centered around the central peak. It doesn't have to be perfect.
As a useful check to make sure what you are recording has been processed by Dolby Advanced Audio, you can disable all effects on the output device in Windows and record the impulse a second time. This should show up as a single peak sample and not the symmetric pattern.
After a bit of research I found this explanation that seems to have satisfied my query. It generally says ...
There isn't going to be an easy fix for this, unless Dolby releases a Linux driver or publishes more information on what exactly their software is doing (which is unlikely).
Haswell-ThinkPad-problems, linux-low-audio-quality
Beware that recently PulseEffects has changed it's name to EasyEffects, but PulseEffects-Presets hasn't updated it's config files to cover this change; Therefore this answer might not be applicable anymore.
I have a DirectShow application that I built with Delphi 6 using the DSPACK component library. For two days I have been trying to solve a problem with audio playback. When I run the filter graph I create I hear repetitive clicks in the playback. What was really confusing was that the audio file I created simultaneously with my filter graph had clean continuous audio, not gaps. So I knew that the audio buffers were being delivered properly but something I was doing was "jamming up" the "live" playback. Or so I thought. I spent two days diagnosing the problem looking for semaphores being held too long (locks) or perhaps timestamp problems, which I documented in this other Stack Overflow post:
Getting stuttering during rendering of my DirectShow filter despite output file being "smooth"
A few minutes ago I decided to try a test with the Graph Edit utility. I created a dead simple graph consisting of just the capture device I was using (VOIP phone microphone), and the renderer device I was using (HD ATI Rear Audio output to headphones). Two filters total. Much to my surprise I heard the same clicking. So here was a case that did not involve my code at all and I heard clicking.
Then I changed the audio renderer in the Graph Edit created filter graph to the VOIP phone ear piece. The clicking went away.
Now I know there's a way to get smooth audio on ut the ATI Rear Audio device since its the preferred audio output device and everything from videos I play on my PC to wave files I play on it sound flawless. So are the other software programs doing something different than just connecting filters? I am wondering if perhaps the default mode for the HD ATI Rear Audio is without double-buffering and perhaps those other software programs know how to enable that feature? Or are they doing something else, perhaps using another DirectShow or DirectSound filter or technique for example, to make the audio play smoothly on the HD ATI Rear Audio renderer?
What you possibly having (depends on actual stuttering though) is that when you are using capture and playback devices backed by different hardware, their sampling rates slightly differ. For example, you capture 22050 Hz at actual rate of (22050 - 2%) Hz and you play it back with hardware consuming bytes at (22050 + 2%) Hz.
Now obviously this won't work out smooth: eventually playback will experience data underlow... If you save into file and play back from file, it will go smooth as the file will be able to supply data at the rate of playback device. If capture and playback devices are the same hardware, they are likely to use shared "hardware" clock and rates match.
The problem is known as "rate matcing" and is discussed on MSDN in Live Sources section.