I'm trying to make an audio visualizer on Linux but I can't seem to find any library that allows me to read the sound levels of the audio of the system. All the libraries I seem to come across seem to either be for reading the microphone or for recording the system audio.
I've tried using pyaudio but for some reason, I've only been able to open the microphone and while the program is running the audio coming out of the speakers then gets distorted.
edit: I'm using PulseAudio but I would be fine switching to another sound driver like alsa, etc.
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My goal is to be able to write sheet music in Musescore and then have the audio output of the playback routed to Ableton Live.
I've tried using loopMIDI audio and LoopBe1 as virtual midi cables.
I have the Jack audio driver set in Ableton's audio preferences under ASIO drivers. As seen in the photo, it seems that Ableton is recognizing the virtual midi cables as an input. I have Musescore's Jack audio settings enabled. I have a midi instrument set up in Ableton. However, when I play back audio in Musescore Ableton doesn't seem to be recognizing any input.
I was trying to follow along with this tutorial. However, they seemed to omit certain details. For example, as seen in my image I was only able to route general sound/midi devices together not specific [left1,right1] to another [in1,in2]
I am new to PulseAudio and ALSA, so please go easy on me. This might seem like a dumb question, but it is quite important to have it answered.
I am developing application on ARM imx6 board (lets call it BOARD1), with built-in sound card support. With ALSA, I am able to play audio throgh Headset_OUT. But now, we want to move to a new board (lets call it BOARD2), which does not have built in soundcard. But the idea is to connect a bluetooth module to the BOARD2 and have the audio streamed to the bluetooth speaker.
My question is, is it possible to use PULSEAUDIO to send/receive audio to external (bluetooth) audio device without local embedded soundcard (i.e. is it possible to do audio encoding/decoding in just software with pulseaudio and gstreamer combination) ?
Regards
I am to test voice recognition programs. Some which I have access to the code and others where I don't.
Sadly my (beautiful) voice is not perfect, so when I am reading a text it sounds slightly different each time. Which makes the testing difficult and time consuming. Giving that I can tweak a lot of parameters.
So I was wondering if there was a way to record my own voice (already done). And then play it as normal microphone input so the voice recognition program I am testing will see it as microphone input.
This would also help greatly if it could be done programatically in C#. So I can in my own code specify when to play what.
To play it from speakers and have the voice recognition programs listen to the microphone is not an option, because it is not the same sound on different computers/speakers/microphones.
Thanks.
Edit:
What i have found so far is to use a software sound Card simulator. But I haven't been able to find a suitable one.
Just as there are printer drivers that do not connect to a printer at all but rather write to a PDF file, analogously there are virtual audio drivers available that do not connect to a physical microphone at all but can pipe input from other sources such as files or other programs.
I hope I'm not breaking any rules by recommending free/donation software, but VB-Audio Virtual Cable should let you create a pair of virtual input and output audio devices. Then you could play an MP3 into the virtual output device and then set the virtual input device as your "microphone". In theory I think that should work.
If all else fails, you could always roll your own virtual audio driver. Microsoft provides some sample code but unfortunately it is not applicable to the older Windows XP audio model. There is probably sample code available for XP too.
I have developed a pretty complex audio software for my client with plugins for Winamp, Windows Media player and VST. Now the client is interested in some method to avoid maintaining the multitude of plugins, we have no way to support all the media players out there.
The client does not care for Unix/Mac yet, so I can look only at Windows XP and Vista/7/
Basically, what we need is a way to always reliably intercept as much audio stream protocols as possible (well, except maybe ASIO, that's another story, I guess), then pass this audio through our custom effects engine and then route back to the default audio device, whatever it is.
Now I am thinking, what options do I have (theoretically).
I could use hooks. I need to hook globally older vaweOut and also DirectSound.
But will this still work on Vista/7?
I could use a virtual driver, like the author of the Virtual Audio Cable did:
http://software.muzychenko.net/eng/vac.htm
Seems a pretty daunting task. Anyway, the client will contact the author of VAC to see if he agrees to sell his source code for a reasonable price.
This driver could install itself as a default audio output device, intercept the audio stream from Windows, and pass it back to default device. Hmm, but what about various DirectSound audio buffers, do I have to mix them myself or is there any way I could tell Windows mixer to mix all for me and pass a single mixed audio stream?
It seems, this custom driver will of course kill all the hardware audio acceleration, but we can live with that, if we warn our customers about this issue.
As I understand, the most current Windows driver standard is WDF.
But maybe it does not work for audio on Windows Vista/7?
I know, Vista/7 has a different audio stack from XP.
If I can do it using WDF, what driver should I write - kernel mode or user mode?
Maybe I am missing more elegant and simple options to intercept, process and route audio on Windows?
Try Virtual Audio Streaming SDK. Also virutal sound card and let you read/process audio data in realtime.
http://www.virtualaudiostreaming.net/sdk-license.html
Currently In my embedded linux I can not open multiple ALSA sound stream. What I need is to have different sound playing at a time. Like one application is playing video and other playing some wav file at a same time.
Thanks,
Sunny.
Is dmix enabled? If it is not possible, you need a software mixing daemon, like pulseaudio.