Is there any way to detect system sound instead of microphone sound? I want to be able to detect whenever my system makes a sound instead of when the microphone picks up the actual sound.
One way I found to do this use an "audio loop-back in either software or hardware (e.g. connect a lead from the speaker 'out' jack to the microphone 'in' jack)."
Capturing speaker output in Java
I am building a program that plays an mp3 file whenever a system sound happens but I don't want it to go off if the dog barks.
Thanks!
What about something with pyaudio (http://people.csail.mit.edu/hubert/pyaudio/)
Like this:
import pyaudio
chunk = 1024
p = pyaudio.PyAudio()
stream = p.open(format=pyaudio.paInt16,
channels=1,
rate=44100,
input=True,
frames_per_buffer=chunk)
data = stream.read(chunk)
And then you could calculate the root-mean-square(RMS) of the audio sample and go from there.
Edited:
You can see what kind of devices you can use by doing something like the following. (http://people.csail.mit.edu/hubert/pyaudio/docs/#pyaudio.PyAudio.get_device_info_by_index)
import pyaudio
p = pyaudio.PyAudio()
for i in xrange(0,10):
try:
p.get_device_info_by_index(i)
except Exception,e:print e
Related
I am trying to perform a simple task, select an input device and set the output device.
The use case is as follows, I have 3.5mm jacks and my user can select the output device (headphones or speaker) from a list.
I can play a sound on a given device (with clip), I can control the input device (mute/volume), but I haven't found any way to specify the target line, it's always the system default.
I can get the mixer
Optional<Mixer.Info> optJackInMixerInfo = Arrays.stream(AudioSystem.getMixerInfo())
.filter(mixerInfo -> {
// Filter based on the device name.
})
.findFirst();
Mixer m = AudioSystem.getMixer(jackInMixerInfo);
// The target
Line.Info[] lineInfos = m.getTargetLineInfo();
for (Line.Info lineInfo : lineInfos) {
m.getLine(lineInfo);
System.out.println("ici");
}
I got only the "master volume control".
How can I select the output device ? I can be happy with changing the system default device too.
The naming of TargetDataLine and SourceDataLine is kind of backwards. Outputs to the local sound system for playback are directed to a SourceDataLine and inputs to Java like microphone lines use TargetDataLine. I used to know why they were named this way but it's slipped my mind at the moment.
There is a tutorial Accessing Audio System Resources with specifics.
Most computers only have a limited number of float controls available, with "master volume" being the one most likely to be implemented. You would use this to alter the volume of the output. Another tutorial in the series, Processing Audio with Controls covers this topic. For myself, I generally convert the audio stream to PCM and handle volume directly (multiply each value by a factor that ranges from 0 to 1) and then convert back to a byte stream, rather than rely on controls which may or may not be present.
I have a webpage with a video. I have to protect this video from capturing it from browser with video capturing programms. I think that for this task i need to check process list or something like this, but to do this i have to use Java. Could anyone give advice how to create this kind of programm. thanks!
I think that it is not possible to prevent user from video capturing. You can make it harder but you will never prevent user from capture screen of his computer. Even if you will control process list of computer (which i guess impossible or impossible for most users) You still cant prevent video capture from computer's video output.
You asking about thing that looks like DRM. History shows that task unsolveable.
You can try identify capturing users (if they will drop your video to torrent). With special unique marks which you can add to video for user. google: steganography
So, I went over the Java's sound tutorial and I did not find it all so helpful.
Anyways, what I understood from the tutorial for recording sound from a mic is this:
Although they do show how to get a target data line and so on, they do not tell how you can actually record sound [or maybe I didn't get it all well].
My understanding so far has been this:
Mixer can be your sound card or sound software drivers that can be used to process the sound, whether input or output
TargetDataLine is used when you want to output your sound into the computer. Like save it to the disk
Port is where your external devices like mic, etc are connected
Problems that remain
How do I select the proper mixer? Java's tut says that you get all the available mixers and query each one to see if it has what you want. That's quite vague for a beginner
How do I get the port on which my integrated mic is? Specifically, how do I get input from it into the mixer?
How do I output this to the disk?
Using the AudioSystem.getTargetDataLine(AudioFormat format) method you will get
... a target data line that can be used for recording audio data in the format specified by the AudioFormat object. The returned line will be provided by the default system mixer, or, if not possible, by any other mixer installed in the system that supports a matching TargetDataLine object.
See the accepted answer for Java Sound API - capturing microphone for an example of this.
If you want more control of which data line to use you can enumerate all the mixers and the data lines they support and pick the one you want. Here is some more information regarding how you would go about doing that: Java - recording from mixer
Once you've obtained the TargetDataLine you should open() it, and then call read() repeatedly to obtain data from that data line. The byte[] that you fill up with data with each call to read() can be written to disk e.g. through a FileOutputStream.
If any of my fellow Xuggler users can tell me what I'm doing wrong, that would be awesome! I am doing the following:
Reading from ogv (ogg video)
Queueing the audio and video
Writing back to ogv
sounds simple right? The problem I am experiencing in my QueueMixer class is that the output file plays the audio 2x too fast, and no matter what I check or change with regards to pts it doesnt improve.
The eclipse project and all files being used are at this link:
http://dl.dropbox.com/u/7316897/paul-xuggled.zip
To run a test, compile and execute the StreamManager class; a test ogv file is included. Before anyone asks, yes I have to queue the data as it will be mixed with other data in a future version.
Audio has a sample rate. Make sure that you copy the sample rate from the input to the output, otherwise the system might use a default (like 44KHz while the original data was sampled at 22KHz which would create such an effect).
The fix is to multiply the sample count by two when extracting them from the ShortBuffer.
samples = new short[(int) audioSamples.getNumSamples() * 2];
audioSamples.getByteBuffer().asShortBuffer().get(samples);
Having half the samples causes the audio to play twice as fast.
I'm using the javax.sound.sampled package in a radio data mode decoding program. To use the program the user feeds audio from their radio receiver into their PC's line input. The user is also required to use their mixer program to select the line in as the recording input. The trouble is some users don't know how to do this and also sometimes other programs alter the recording input setting. So my question is how can my program detect if the line in is set as the recording input ? Also is it possible for my program to change the recording input setting if it detects it is incorrect ?
Thanks for your time.
Ian
To answer your first question, you can check if the Line.Info object for your recording input matches Port.Info.LINE_IN like this:
public static boolean isLineIn(Line.Info lineInfo) {
Line.Info[] detected = AudioSystem.getSourceLineInfo(Port.Info.LINE_IN);
for (Line.Info lineIn : detected) {
if (lineIn.matches(lineInfo)) {
return true;
}
}
return false;
}
However, this doesn't work with operating systems or soundcard driver APIs that don't provide the type of each available mixer channel. So when I test it on Windows it works, but not on Linux or Mac. For more information and recommendations, see this FAQ.
Regarding your second question, you can try changing the recording input settings through a Control class. In particular, see FloatControl.Type for some common settings. Keep in mind that the availability of these controls depends on the operating system and soundcard drivers, just like line-in detection.