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.
Related
I'm trying to record from the microphone to a wav file as per this example. At the same time, I need to be able to test for input level/volume and send an alert if it's too low. I've tried what's described in this link and seems to work ok.
The issue comes when trying to record and read bytes at the same time using one TargetDataLine (bytes read for monitoring are being skipped for recording and vice-versa.
Another thing is that these are long processes (hours probably) so memory usage should be considered.
How should I proceed here? Any way to clone TargetDataLine? Can I buffer a number of bytes while writing them with AudioSystem.write()? Is there any other way to write to a .wav file without filling the system memory?
Thanks!
If you are using a TargetDataLine for capturing audio similar to the example given in the Java Tutorials, then you have access to a byte array called "data". You can loop through this array to test the volume level before outputting it.
To do the volume testing, you will have to convert the bytes to some sort of sensible PCM data. For example, if the format is 16-bit stereo little-endian, you might take two bytes and assemble to either a signed short or a signed, normalized float, and then test.
I apologize for not looking more closely at your examples before posting my "solution".
I'm going to suggest that you extend InputStream, making a customized version that also performs the volume test. Override the 'read' method so that it obtains the byte that it returns from the code you have that tests the volume. You'll have to modify the volume-testing code to work on a per-byte basis and to pass through the required byte.
You should then be able to use this extended InputStream as an argument when you create the AudioInputStream for the output-to-wav stage.
I've used this approach to save audio successfully via two data sources: once from an array that is populated beforehand, once from a streaming audio mix passing through a "mixer" I wrote to combine audio data sources. The latter would be more like what you need to do. I haven't done it from a microphone source, though. But the same approach should work, as far as I can tell.
I've a stream with contains a audio/video stream. The video encoding is H.264 AVC and the audio encoding is G.711 ยต-law (PCMU). (I have no control over the output format.)
When I try to display the video into a VideoView frame, the device says that it cannot play the video. I guess that's because Android doesn't support the aforementioned audio encoding.
Is there a way to somehow display the selected stream in the VideoView?
Can I just use a Java code snippet to 'compute' the codec? I've seen the class android.net.rtp.AudioCodec, which contains a field PCMU, and I can imagine that class would be useful.
Or do I have to insert a library or some native code (FFmpeg) into my application and use that? If yes, how?
How to fix it?
If you want to manipulate a stream before you display it, you are unfortunately getting into tricky territory on an Android device - if you do have any possibility of doing the conversion on the serve side it will likely by much easier (and you should also usually have more 'horsepower' on the server side).
Assuming that you do need to convert on the server side, then a technique you can use is to stream the video from the server to your app, convert it as needed and then 'stream' from a localhost server in your app to a VideoView in your app. Roughly the steps are:
Stream the encrypted file from the server as usual, 'chunk by chunk'
On your Android device, read from the stream and convert each chunk as it is received
Using a localhost http server on your Android device, now 'serve' the converted chunks to the MediaPlayer (the media player should be set up to use a URL pointing at your localhost http server)
An example of this approach, I believe, is LibMedia: sure: http://libeasy.alwaysdata.net (note, this is not a free library, AFAIK).
For the actual conversion you can use ffmpeg as you suggest - there are a number of ffmpeg wrapper for Android projects that you can either use or look at to design your own. Some examples:
http://hiteshsondhi88.github.io/ffmpeg-android-java/
https://github.com/jhotovy/android-ffmpeg
Take a look at the note about libffmpeginvoke in the second link in particular if you are planning to write your won wrapper.
One thing to be aware of is that compressions techniques, in particular video compression, often use an approach where one packet or frame is compressed relative to the frames before it (and sometimes even the frames after it). For example the first frame might be a 'key frame', the next five frames might just contain data to explain how they differ from the key frame, and the the seventh frame might be another key frame and so on. This means you generally want a 'chunk' which has all the required key frames if you are converting the chunk from one format to another.
I don't think you will have this problem converting from PCM to AAC audio, but it useful to be aware of the concept anyway.
All I need is a peice of java code that can detect DTMF from microphone print out the characters to System.out. I've been searching forever and I couldn't find it.
Oracle Docs on Capturing Audio in Java:
http://docs.oracle.com/javase/tutorial/sound/capturing.html
As discussed in Overview of the Sampled Package, a typical audio-input system in an implementation of the Java Sound API consists of:
An input port, such as a microphone port or a line-in port, which feeds its incoming audio data into:
A mixer, which places the input data in:
One or more target data lines, from which an application can retrieve the data.
(Emphasis Mine)
Also see:
Java (J2SE) DTMF tone detection
I think usually this is done in hardware, so you may end up writing code yourself to analyze the audio you've captured.
Also:
http://sourceforge.net/projects/java-dtmf/
I'd like to modify the audio input stream, the stream that would come
from my microphone.
I have looked through the java.sound package API, but did not entirely understand it,
nor how to modify direct sound input.
Does anyone here know how to do that, or know an API that is capable of doing it?
You want a mixture of things:
The Java Sound system: http://www.oracle.com/technetwork/java/index-139508.html
A trail for it: http://docs.oracle.com/javase/tutorial/sound/index.html
Using audio controls: http://docs.oracle.com/javase/1.5.0/docs/guide/sound/programmer_guide/chapter6.html (part of a wider set of documentation)
If you are able to give more information about what you want to do to the audio stream, it's likely we'll be able to give you more specific advice.
I've written a program that transmits a PCM stream from my pc to another pc or my android(using an AudioTrack). It uses java sound and takes the target and source lines from the Stereo Mix mixer. Everything technically works (sockets, mixers, lines, streams, buffers, etc.) but the output on the remote speakers is very static-y sounding. Even when there isn't sound coming through, there is a constant crackling sound. I've tested the programs on my phone and computer with the same result.
What can I do?
Thanks,
Bill
ps The code is pretty big, and kinda messy, but I can post it somewhere if you ask.
Typically the static means you're feeding incorrect data to the audio subsystem: given what you describe there's probably an error in the audio path, and I'd suspect you are either:
Experiencing some byte alignment issue when reading/transmitting/receiving the audio data from the source.
Inadvertently mixing your network stream with local loopback audio on the receiving end.
Feeding in invalid silence data (for sound subsystems that need to be constantly fed with audio data).
Not feeding in silence data when you should be.