I should want to detect frequencies in music with java.
I use TarsosDSP, which work ok, but I'm afraid there no active community (I think).
Is there other ?
On stackoverflow I've seen Sound processing: mixing two audio files, phase shifting and peak controller, Capturing audio and processing rythm in realtime, audio search library... but there's no accepted answer, so I try me too.
Related
My overarching goal: I'm looking for a way to grab current system sound and run it through a visualizer in Processing 3. Currently I have found a way to do this grabbing the mic input:
function setup() {
sound = new p5.AudioIn();
sound.start();
fft = new p5.FFT();
fft.setInput(sound);
}
But I have yet to find a way to do this with system sound (ie. A youtube video, Spotify, an MP3 file playing)
As well i'm not even sure if this is possible with some programs like spottily as they have built in security.
All in all I think the solution to this problem is probably similar to how you would go about capturing system audio in a screen recording program.
Note: The captured audio is being pipped into the Minim Library for visual processing.
Capturing system output in Processing is a bit tricky. In fact, even being able to record system output is a demon of its own.
I managed to accomplish this task on my MacBook Pro in Processing by using Soundflower (Mac) as a workaround. This application acts as a MIDI device to route your sound output to your sound input. Once installed, open Audio MIDI Setup and select Soundflower as your sound input.
When you run your Processing script, p5.AudioIn() will take the Soundflower audio input channel and use it get the frequency bands of all sounds coming out of your computer alone.
Best of luck!
I need to implement a voice activity detection algorithm in Java so that I can know when to start and/or stop recording audio. I am looking for an algorithm that can take either a byte[], a target-data-line, or an audio file as input. Also, a solution would preferably not use external dependencies.
Give a look at TarsosDSP as source of inspiration: It is so far the best open source Java library to deal with Audio Detection. It is purely written in Java and briefly provides:
SoundDetection
PitchDetection
PercussionDetection
Audio Time Stretching
Pitch Shifting
IIR- filters
Check also the official paper and manual for a better grasp on the topic.
My application takes a long time to prepare and buffer an audio stream. I have read this question Why does it take so long for Android's MediaPlayer to prepare some live streams for playback?, however it just says people have experienced this issue, it does not state how to improve the problem.
I am experiencing this in all versions of Android, tested from 2.2 - 4.1.2.
The streams are in a suitable bit-rate for mobile and 3G connection. The same stream takes less than a second to start buffering in the equivalent iOS app.
Is there a way to specify the amount of time that should be buffered? I know that the Tune In radio application offers this feature ( https://play.google.com/store/apps/details?id=tunein.player ).
Thanks.
Edit: I've tested again and found that it only happens on devices running Gingerbread and above (>=2.3). I know that Android changed the underlying framework from OpenCore to StageFright. So how can I optimise the media framework? It just seems wrong that the old HTC Wildfire can prepare, stream and play, literally 10x faster than the brand new HTC One X and Nexus 7.
I have struggled with this question for months. Finally i found the solution.
The real problem is in the implementation of the MediaPlayer class. Particularly with the way MediaPlayer buffers the data. This is why the solution is to create your own buffering, save it to a temp file and feed that to MediaPlayer.
This tutorial and source code explain exactly how. http://androidstreamingtut.blogspot.nl/2012/08/custom-progressive-audio-streaming-with.html
By adapting this code, it is easy to create a much better streaming player.
Google Developers really screwed up here.
EDIT : This answer is rather old. Nowdays i would recommend not using MediaPlayer and use ExoPlayer instead. It is extendable, stable and can play many different types of media. You can find it here: https://github.com/google/ExoPlayer/
There really isn't much you can do since the Android MediaPlayer class doesn't provide access to lower level settings such as buffer size. The only alternative would be to make your own player using AudioTrack and a library like FFmpeg to do the decoding.
The one thing I'd recommend is to play around with encoding. For instance, for MP4s, ensure that the MOOV Atom is located at the beginning of the file (there are enough questions on S/O regarding how to do this with ffmpeg, etc). With MP3s, you can look at different codecs or bitrates for instance.
You can, for instance, try a number of audio files you find online, and if you see one that doesn't take a long time to buffer, try to encode your files in the same way.
I read this code to play video files using xuggler as a library.When i run this code , the video without sound plays.Why is that ?
If not what should i do to play audio with video.?
Quite simply, it's because that code doesn't play any audio. It was never written to do so!
If you want to play audio and video, check out the example here:
http://xuggle.googlecode.com/svn/trunk/java/xuggle-xuggler/src/com/xuggle/xuggler/demos/DecodeAndPlayAudioAndVideo.java
The question to me suggests that you've just grabbed the source without looking at the tutorials - I'd strongly suggest doing so so you can understand the code behind what's going on rather than just throwing it in there and expecting it to work - especially if you're planning on using Xuggler in any great depth!
As a further note, if all you want to do is play videos I probably wouldn't advise using Xuggler because it's just not really designed for that sort of thing. Sure it will play them, but you have to manage all the sync issues yourself (which the above example doesn't do a great job of), it will only use your CPU and therefore really eat into its consumption especially if you're doing things like HD video, and the sheer amount of background code is just overkill (getting the streams, finding the stream IDs, extracting each packet, determining the type of the packet, dealing with it appropriately, managing errors, etc. etc.) It's a great piece of software, but if you're only using it to play stuff then it'll probably give you far more trouble than its worth.
Instead I'd advocate having a play with VLCJ - it's a bit complicated to get set up to work reliably (out of process players are required for rock solid stability) but when you've got that far it plays near any file type under the sun and manages all the issues above for you much, much better than you'll ever cope with by hand!
My guess is that you have a missing audio codec. I've had this same issue but vice versa; the sound played but the video didn't.
I'm trying to record/process some audio from three usb microphones with Java Sound on Snow Leopard (but can switch to Windows if it fixes things). Problem is, when I try to use the mixer that corresponds to the usb mic, Java Sound tells me that the line isn't supported. Specifically, it says this...
Available mixers:
Java Sound Audio Engine
USBMIC Serial# 041270067
Built-in Input Built-in Microphone
Soundflower (2ch)
Soundflower (16ch)
Exception in thread "AWT-EventQueue-0"
java.lang.IllegalArgumentException:
Line unsupported: interface
TargetDataLine supporting format
PCM_SIGNED 96000.0 Hz, 8 bit, stereo,
2 bytes/frame,
...when I ask it to select the USBMIC mixer:
Mixer mixer = AudioSystem.
getMixer(mixerInfo[1]);
I have tried matching the audio format to the exact specifications of the microphones (16-bit, 44100Hz, stereo) and it didn't make any difference.
The problem is cropping up here:
final TargetDataLine line = (TargetDataLine)
mixer.getLine(info);
It would seem that the mixer and the TargetDataLine don't like each other. Is there some way to get these two to 'match' and get along?
The microphones that I'm using are admittedly a bit strange. They were made to be used in a karaoke video game called SingStar. The mics themselves have standard mono line-in connectors that plug into a little hub (two to a hub) that converts them into a single male usb connector. Strangeness aside, though, they seem to work perfectly fine with Audacity as separate channels, so multichannel recording with them is clearly possible, just maybe not in Java.
I've also considered using a program like Soundflower that shares audio between different programs. However, I'm not sure this will work as I can't see how to make the USB mics inputs to Soundflower and then pipe them into a Java. A quick experiment showed me that I could record audio in Audacity from the mics, pipe it out through Soundflower, and then process in my Java program. Still, what I would like to do is have it all happen in real time in Java.
Anybody familiar with this kind of problem?
I think that a simple way to do this would be using Soundflower and Soundflowerbed.
I can't see how to make the USB mics inputs to Soundflower and then pipe them into a Java.
It sounds like you have Soundflower installed already. Soundflowerbed is found in the same disk image as Soundflower and is a menubar application. It lets you route sound between applications which don't have controls built in for selecting sound devices. Install that from the disk image and click it to run.
All of the following will be using my Echo Audiofire 4 but in principle should work on any audio device.
Using Soundflowerbed
Open Soundflower and tick the audio device you want to use under Soundflower (16ch). As I'm a new user I can't post images but they are linked below. If I get the bounty then I will edit the post to include the images inline.
From here you would use Soundflower (16ch) as your audio input device in Java sound.
Creating an aggregate audio device
An alternative way to solve this if that didn't work is to create an aggregate device. Open Applications > Utilities > Audio Midi Setup and click the plus sign to create a new aggregate device.
Tick the device that you want to aggregate. You only want your USBMIC (As I'm a new SO user I can only post two images per answer so the next two are linked here).
The key part which may be giving you trouble is the clock on the device. If you select the Mac as the clock source then that may be more stable.
If this still doesn't work then you could try adding the Mac built-in audio to the aggregate device and making it the master clock by right clicking on the device you want to be the master.
Other options
Finally, I haven't used this before but Pulse Audio (Google it, I can't insert more links in this post) might be a possible solution for mixing your audio streams together. It looks quite heavyweight though.
According to my research, especially threads like this, the microphone you are using is most likely causing the problem. The thread states that the microphone is even a problem when it comes to switching games, so I am guessing that it will be a problem when switching platforms, too.
My suggestion is - if you have not tried this already - to use a different microphone! Most microphones I have messed around with have special chip controllers that convert data into the data compatible for the game system. Being that you are using this on an operating system for the computer, you are probably getting some very odd effects that you wouldn't get on a game system like Playstation or others.
Hopefully this helps! Happy coding!
The AudioFormat doesn't match the TargetDataLine's supported format. I don't know if that was a typo or not but the Exception thrown says the TargetDataLine supports 8 bit audio and right below that you said you're using a 16 bit AudioFormat. It also supports up to 2 bytes per frame, how quickly and in what size chunks are you trying to read the data? Sorry if that doesn't help but I thought I'd point that out in case it was overlooked.