How to properly detect, decode and play a radio stream? - java

I am currently trying to write a jukebox-like application in Java that is able to play any audio source possible, but encountered some difficulties when trying to play radio streams.
For playback I use JLayer from JavaZoom, that works fine as long as the target is a direct media file or a direct media stream (I can play PCM, MP3 and OGG just fine). However I encounter difficulties when trying to play radio streams which either contain pre-media data like a m3u/pls file (which I could fix by adding a detection beforehand), or data that is streamed on port 80 while a web-page exists at the same location and the media transmitted depends on the type of request. In the later case, whenever I try to stream the media, I instead get the HTML data.
Example link of a stream that is hidden behind a web-page: http://stream.t-n-media.de:8030
This is playable in VLC, but if you put it into a browser or my application you'll receive an HTML file.
Is there:
A ready-made, free solution that I could use in place of JLayer? Preferably open source so I can study it?
A tutorial that can help me to write a solution on my own?
Or can someone give me an example on how to properly detect/request a media stream?
Thanks in advance!


import java.io.*;
import java.net.*;
import javax.sound.sampled.*;
import javax.sound.midi.*;
/**
* This class plays sounds streaming from a URL: it does not have to preload
* the entire sound into memory before playing it. It is a command-line
* application with no gui. It includes code to convert ULAW and ALAW
* audio formats to PCM so they can be played. Use the -m command-line option
* before MIDI files.
*/
public class PlaySoundStream {
// Create a URL from the command-line argument and pass it to the
// right static method depending on the presence of the -m (MIDI) option.
public static void main(String[ ] args) throws Exception {
if (args[0].equals("-m")) streamMidiSequence(new URL(args[1]));
else streamSampledAudio(new URL(args[0]));
// Exit explicitly.
// This is needed because the audio system starts background threads.
System.exit(0);
}
/** Read sampled audio data from the specified URL and play it */
public static void streamSampledAudio(URL url)
throws IOException, UnsupportedAudioFileException,
LineUnavailableException
{
AudioInputStream ain = null; // We read audio data from here
SourceDataLine line = null; // And write it here.
try {
// Get an audio input stream from the URL
ain=AudioSystem.getAudioInputStream(url);
// Get information about the format of the stream
AudioFormat format = ain.getFormat( );
DataLine.Info info=new DataLine.Info(SourceDataLine.class,format);
// If the format is not supported directly (i.e. if it is not PCM
// encoded), then try to transcode it to PCM.
if (!AudioSystem.isLineSupported(info)) {
// This is the PCM format we want to transcode to.
// The parameters here are audio format details that you
// shouldn't need to understand for casual use.
AudioFormat pcm =
new AudioFormat(format.getSampleRate( ), 16,
format.getChannels( ), true, false);
// Get a wrapper stream around the input stream that does the
// transcoding for us.
ain = AudioSystem.getAudioInputStream(pcm, ain);
// Update the format and info variables for the transcoded data
format = ain.getFormat( );
info = new DataLine.Info(SourceDataLine.class, format);
}
// Open the line through which we'll play the streaming audio.
line = (SourceDataLine) AudioSystem.getLine(info);
line.open(format);
// Allocate a buffer for reading from the input stream and writing
// to the line. Make it large enough to hold 4k audio frames.
// Note that the SourceDataLine also has its own internal buffer.
int framesize = format.getFrameSize( );
byte[ ] buffer = new byte[4 * 1024 * framesize]; // the buffer
int numbytes = 0; // how many bytes
// We haven't started the line yet.
boolean started = false;
for(;;) { // We'll exit the loop when we reach the end of stream
// First, read some bytes from the input stream.
int bytesread=ain.read(buffer,numbytes,buffer.length-numbytes);
// If there were no more bytes to read, we're done.
if (bytesread == -1) break;
numbytes += bytesread;
// Now that we've got some audio data to write to the line,
// start the line, so it will play that data as we write it.
if (!started) {
line.start( );
started = true;
}
// We must write bytes to the line in an integer multiple of
// the framesize. So figure out how many bytes we'll write.
int bytestowrite = (numbytes/framesize)*framesize;
// Now write the bytes. The line will buffer them and play
// them. This call will block until all bytes are written.
line.write(buffer, 0, bytestowrite);
// If we didn't have an integer multiple of the frame size,
// then copy the remaining bytes to the start of the buffer.
int remaining = numbytes - bytestowrite;
if (remaining > 0)
System.arraycopy(buffer,bytestowrite,buffer,0,remaining);
numbytes = remaining;
}
// Now block until all buffered sound finishes playing.
line.drain( );
}
finally { // Always relinquish the resources we use
if (line != null) line.close( );
if (ain != null) ain.close( );
}
}
// A MIDI protocol constant that isn't defined by javax.sound.midi
public static final int END_OF_TRACK = 47;
/* MIDI or RMF data from the specified URL and play it */
public static void streamMidiSequence(URL url)
throws IOException, InvalidMidiDataException, MidiUnavailableException
{
Sequencer sequencer=null; // Converts a Sequence to MIDI events
Synthesizer synthesizer=null; // Plays notes in response to MIDI events
try {
// Create, open, and connect a Sequencer and Synthesizer
// They are closed in the finally block at the end of this method.
sequencer = MidiSystem.getSequencer( );
sequencer.open( );
synthesizer = MidiSystem.getSynthesizer( );
synthesizer.open( );
sequencer.getTransmitter( ).setReceiver(synthesizer.getReceiver( ));
// Specify the InputStream to stream the sequence from
sequencer.setSequence(url.openStream( ));
// This is an arbitrary object used with wait and notify to
// prevent the method from returning before the music finishes
final Object lock = new Object( );
// Register a listener to make the method exit when the stream is
// done. See Object.wait( ) and Object.notify( )
sequencer.addMetaEventListener(new MetaEventListener( ) {
public void meta(MetaMessage e) {
if (e.getType( ) == END_OF_TRACK) {
synchronized(lock) {
lock.notify( );
}
}
}
});
// Start playing the music
sequencer.start( );
// Now block until the listener above notifies us that we're done.
synchronized(lock) {
while(sequencer.isRunning( )) {
try { lock.wait( ); } catch(InterruptedException e) { }
}
}
}
finally {
// Always relinquish the sequencer, so others can use it.
if (sequencer != null) sequencer.close( );
if (synthesizer != null) synthesizer.close( );
}
}
}
I have used this piece of code in one of my projects that deal with Audio streaming and was working just fine.
Furthermore, you can see similar examples here:
Java Audio Example


Just reading the javadoc of AudioSystem give me an idea.
There is an other signature for getAudioInputStream: you can give it an InputStream instead of a URL.
So, try to manage to get the input stream by yourself and add the needed headers so that you get the stream instead the html content:
URLConnection uc = url.openConnection();
uc.setRequestProperty("<header name here>", "<header value here>");
InputStream in = uc.getInputStream();
ain=AudioSystem.getAudioInputStream(in);
Hope this help.


I know this answer comes late, but I had the same issue: I wanted to play MP3 and AAC audio and also wanted the user to insert PLS/M3U links. Here is what I did:
First I tried to parse the type by using the simple file name:
import de.webradio.enumerations.FileExtension;
import java.net.URL;
public class FileExtensionParser {
/**
*Parses a file extension
* #param filenameUrl the url
* #return the filename. if filename cannot be determined by file extension, Apache Tika parses by live detection
*/
public FileExtension parseFileExtension(URL filenameUrl) {
String filename = filenameUrl.toString();
if (filename.endsWith(".mp3")) {
return FileExtension.MP3;
} else if (filename.endsWith(".m3u") || filename.endsWith(".m3u8")) {
return FileExtension.M3U;
} else if (filename.endsWith(".aac")) {
return FileExtension.AAC;
} else if(filename.endsWith((".pls"))) {
return FileExtension.PLS;
}
URLTypeParser parser = new URLTypeParser();
return parser.parseByContentDetection(filenameUrl);
}
}
If that fails, I use Apache Tika to do a kind of live detection:
public class URLTypeParser {
/** This class uses Apache Tika to parse an URL using her content
*
* #param url the webstream url
* #return the detected file encoding: MP3, AAC or unsupported
*/
public FileExtension parseByContentDetection(URL url) {
try {
HttpURLConnection connection = (HttpURLConnection) url.openConnection();
InputStream in = connection.getInputStream();
BodyContentHandler handler = new BodyContentHandler();
AudioParser parser = new AudioParser();
Metadata metadata = new Metadata();
parser.parse(in, handler, metadata);
return parseMediaType(metadata);
} catch (IOException e) {
e.printStackTrace();
} catch (TikaException e) {
e.printStackTrace();
} catch (SAXException e) {
e.printStackTrace();
}
return FileExtension.UNSUPPORTED_TYPE;
}
private FileExtension parseMediaType(Metadata metadata) {
String parsedMediaType = metadata.get("encoding");
if (parsedMediaType.equalsIgnoreCase("aac")) {
return FileExtension.AAC;
} else if (parsedMediaType.equalsIgnoreCase("mpeg1l3")) {
return FileExtension.MP3;
}
return FileExtension.UNSUPPORTED_TYPE;
}
}
This will also solve the HTML problem, since the method will return FileExtension.UNSUPPORTED for HTML content.
I combined this classes together with a factory pattern and it works fine. The live detection takes only about two seconds.
I don't think that this will help you anymore but since I struggled almost three weeks I wanted to provide a working answer. You can see the whole project at github: https://github.com/Seppl2202/webradio

Related

PDF file encode to base64 take more time if 100k documents are to be encode

Am trying to encode pdf documents to base64, If it is less in number ( like 2000 documents) its working nicely. But am having 100k plus doucments to be encode.
Its take more time to encode all those files. Is there any better approach to encode large data set.?
Please find my current approach
String filepath=doc.getPath().concat(doc.getFilename());
file = new File(filepath);
if(file.exists() && !file.isDirectory()) {
try {
FileInputStream fileInputStreamReader = new FileInputStream(file);
byte[] bytes = new byte[(int) file.length()];
fileInputStreamReader.read(bytes);
encodedfile = new String(Base64.getEncoder().encodeToString(bytes));
fileInputStreamReader.close();
} catch (FileNotFoundException e) {
e.printStackTrace();
}
}

Try this:
Figure out how many files you need to encode.
int files = Files.list(Paths.get(directory)).count();
Split them up into a reasonable amount that a thread can handle in java. I.E) If you have 100k files to encode. Split it into 1000 lists of 1000, something like that.
int currentIndex = 0;
for (File file : filesInDir) {
if (fileMap.get(currentIndex).size() >= cap)
currentIndex++;
fileMap.get(currentIndex).add(file);
}
/** Its going to take a little more effort than this, but its the idea im trying to show you*/
Execute each worker thread one after another if the computers resources are available.
for (Integer key : fileMap.keySet()) {
new WorkerThread(fileMap.get(key)).start();
}
You can check the current resources available with:
public boolean areResourcesAvailable() {
return imNotThatNice();
}
/**
* Gets the resource utility instance
*
* #return the current instance of the resource utility
*/
private static OperatingSystemMXBean getInstance() {
if (ResourceUtil.instance == null) {
ResourceUtil.instance = ManagementFactory.getOperatingSystemMXBean();
}
return ResourceUtil.instance;
}

Java play AAC encoded audio ( JAAD decoder )

I have struggled with playing aac encoded audio files with Java a while now.
We had a group project at the end of our first semester and wanted to have a background music and few soundeffects in there.
At the end we used WAV files, as we couldn't get the AAC's to play.

This weekend I gave it another try and searched along again, and have got a working code searched together from different sites, but nowhere was a complete working solution.
For a more comfortable usage in future projects I made me a small library for aac playback.
As it was hard to find the working solution, I wanted to share it with you, in the hope some one having the same problem in the future will have it a bit easier.
The following code is a snippet from the lib I wrote. You can have a look onto the whole lib on GitLab at
Java AAC-Player.
This lib/solution uses the aac decoder JAAD.
The player I wrote around the JAAD library is available as Maven artifact:
<dependency>
<groupId>com.gitlab.9lukas5</groupId>
<artifactId>jaad</artifactId>
<!--
check latest version on https://search.maven.org
or try the latest tag you find in git
-->
</dependency>
public static void play(File[] files)
{
// local vars
byte[] b; // array for the actual audio Data during the playback
AudioTrack track; // track we are playing atm
AudioFormat af; // the track's format
SourceDataLine line; // the line we'll use the get our audio to the speaker's
Decoder dec; // decoder to get the audio bytes
Frame frame; //
SampleBuffer buf; //
int currentTrack; // index of current track from playlist
MP4Container cont; // container to open the current track with
Movie movie; // and get the content from the container
try
{
// for-next loop to play each titel from the playlist once
for (currentTrack = 0; currentTrack < files.length; currentTrack++)
{
cont = new MP4Container(new RandomAccessFile(files[currentTrack], "r")); // open titel with random access
movie = cont.getMovie(); // get content from container,
List<Track> content = movie.getTracks();
if (content.isEmpty()) // check if container HAS content
throw new Exception ("insert error message here"); // if so,
track = (AudioTrack) movie.getTracks().get(0); // grab first track and set the audioformat
af = new AudioFormat(track.getSampleRate(), track.getSampleSize(), track.getChannelCount(), true, true);
line = AudioSystem.getSourceDataLine(af); // get a DataLine from the AudioSystem
line.open(); // open and
line.start(); // start it
dec = new Decoder(track.getDecoderSpecificInfo());
buf = new SampleBuffer();
while(track.hasMoreFrames()) // while we have frames left
{
frame = track.readNextFrame(); // read next frame,
dec.decodeFrame(frame.getData(), buf); // decode it and put into the buffer
b = buf.getData(); // write the frame data from the buffer to our byte-array
line.write(b, 0, b.length); // and from there write the byte array into our open AudioSystem DataLine
while (paused) // check if we should pause
{
Thread.sleep(500); // if yes, stay half a second
if (Thread.interrupted()) // check if we should stop possibly
{
line.close(); // if yes, close line and
return; // exit thread
}
}
if (Thread.interrupted()) // if not in pause, still check on each frame if we should
{ // stop. If so
line.close(); // close line and
return; // exit thread
}
}
line.close(); // after titel is over, close line
if (loop) // if we should loop current titel, set currentTrack -1,
currentTrack--; // as on bottom of for-next it get's +1 and so the same titel get's played again
else if (repeat && (currentTrack == files.length -1)) // else check if we are at the end of the playlist
currentTrack = -1; // and should repeat the whole list. If so, set currentTrack -1, so it get's 0 on for-next bottom
}
}
catch (LineUnavailableException | IOException | InterruptedException e)
{
e.printStackTrace();
}
}

Video Streaming on network using java sockets

I am developing a networking java application. I wanted to stream a video from network (maybe using sockets). I search on the internet but I didnt find any working server and client code to stream video from a server to a client.
Can anyone find streaming server and client or code a simple program so that I can understand how streaming is done using java.
PS. I fount an assignment related to this on internet. But it has error and some methods are also unimplemented. If you can remove errors and complete the methods it will also be helpful..
http://cs.anu.edu.au/student/comp3310/2004/Labs/lab6/lab5.html

See: Any simple (and up to date) Java frameworks for embedding movies within a Swing Application?, just refer to the JavaFX only code sample (you don't need any Swing code).
import javafx.application.Application;
import javafx.scene.*;
import javafx.scene.media.*;
import javafx.stage.Stage;
public class VideoPlayerExample extends Application {
public static void main(String[] args) throws Exception { launch(args); }
#Override public void start(final Stage stage) throws Exception {
final MediaPlayer oracleVid = new MediaPlayer(
new Media("http://download.oracle.com/otndocs/products/javafx/oow2010-2.flv")
);
stage.setScene(new Scene(new Group(new MediaView(oracleVid)), 540, 208));
stage.show();
oracleVid.play();
}
}
So, encode your video to a format understood by JavaFX (e.g. h264 encoded mp4) and place it on a http server and you can load the video data over http from your JavaFX client. Ensure that your client is a certified system configuration for media playback using JavaFX.
That is probably sufficient for what you need.
If you need something a bit more fancy, JavaFX also supports http live streaming, which you can read up on and see if you need (which you probably don't). I don't have instructions on setting up a http live streaming server, nor a link to somewhere on the internet on how to do that (you would have to do your own research on that if you want to go that route).
Also, note, I converted the mjpeg player lab assignment you reference in your question to JavaFX to answer the question: Display RTP MJPEG. It is useful if you want to understand at a low level how such video playback is done. However, I would not recommend using this method for your video playback for a production project - instead just use the built-in JavaFX MediaPlayer.

Here are the basic code: http://xuggle.googlecode.com/svn/trunk/java/xuggle-xuggler/src/com/xuggle/xuggler/demos/DecodeAndPlayAudioAndVideo.java
but I changed it to:
package Pasban;
/**
*
* #modified by Pasban
*/
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import com.xuggle.xuggler.Global;
import com.xuggle.xuggler.IAudioSamples;
import com.xuggle.xuggler.IContainer;
import com.xuggle.xuggler.IPacket;
import com.xuggle.xuggler.IPixelFormat;
import com.xuggle.xuggler.IStream;
import com.xuggle.xuggler.IStreamCoder;
import com.xuggle.xuggler.ICodec;
import com.xuggle.xuggler.IVideoPicture;
import com.xuggle.xuggler.IVideoResampler;
import com.xuggle.xuggler.Utils;
import com.xuggle.xuggler.demos.VideoImage;
import java.awt.Dimension;
/**
* Takes a media container, finds the first video stream,
* decodes that stream, and then plays the audio and video.
*
* This code does a VERY coarse job of matching time-stamps, and thus
* the audio and video will float in and out of slight sync. Getting
* time-stamps syncing-up with audio is very system dependent and left
* as an exercise for the reader.
*
* #author aclarke
*
*/
public class DecodeAndPlayAudioAndVideo {
/**
* The audio line we'll output sound to; it'll be the default audio device on your system if available
*/
private static SourceDataLine mLine;
/**
* The window we'll draw the video on.
*
*/
private static VideoImage mScreen = null;
private static long mSystemVideoClockStartTime;
private static long mFirstVideoTimestampInStream;
/**
* Takes a media container (file) as the first argument, opens it,
* plays audio as quickly as it can, and opens up a Swing window and displays
* video frames with <i>roughly</i> the right timing.
*
* #param args Must contain one string which represents a filename
*/
#SuppressWarnings("deprecation")
public static void main(String[] args) {
String filename = "http://techslides.com/demos/sample-videos/small.mp4";
// Let's make sure that we can actually convert video pixel formats.
if (!IVideoResampler.isSupported(IVideoResampler.Feature.FEATURE_COLORSPACECONVERSION)) {
throw new RuntimeException("you must install the GPL version of Xuggler (with IVideoResampler support) for this demo to work");
}
// Create a Xuggler container object
IContainer container = IContainer.make();
// Open up the container
if (container.open("http://techslides.com/demos/sample-videos/small.mp4", IContainer.Type.READ, null) < 0) {
throw new IllegalArgumentException("could not open file: " + filename);
}
// query how many streams the call to open found
int numStreams = container.getNumStreams();
// and iterate through the streams to find the first audio stream
int videoStreamId = -1;
IStreamCoder videoCoder = null;
int audioStreamId = -1;
IStreamCoder audioCoder = null;
for (int i = 0; i < numStreams; i++) {
// Find the stream object
IStream stream = container.getStream(i);
// Get the pre-configured decoder that can decode this stream;
IStreamCoder coder = stream.getStreamCoder();
if (videoStreamId == -1 && coder.getCodecType() == ICodec.Type.CODEC_TYPE_VIDEO) {
videoStreamId = i;
videoCoder = coder;
} else if (audioStreamId == -1 && coder.getCodecType() == ICodec.Type.CODEC_TYPE_AUDIO) {
audioStreamId = i;
audioCoder = coder;
}
}
if (videoStreamId == -1 && audioStreamId == -1) {
throw new RuntimeException("could not find audio or video stream in container: " + filename);
}
/*
* Check if we have a video stream in this file. If so let's open up our decoder so it can
* do work.
*/
IVideoResampler resampler = null;
if (videoCoder != null) {
if (videoCoder.open() < 0) {
throw new RuntimeException("could not open audio decoder for container: " + filename);
}
if (videoCoder.getPixelType() != IPixelFormat.Type.BGR24) {
// if this stream is not in BGR24, we're going to need to
// convert it. The VideoResampler does that for us.
resampler = IVideoResampler.make(videoCoder.getWidth(), videoCoder.getHeight(), IPixelFormat.Type.BGR24,
videoCoder.getWidth(), videoCoder.getHeight(), videoCoder.getPixelType());
openJavaVideo(videoCoder);
if (resampler == null) {
throw new RuntimeException("could not create color space resampler for: " + filename);
}
}
/*
* And once we have that, we draw a window on screen
*/
}
if (audioCoder != null) {
if (audioCoder.open() < 0) {
throw new RuntimeException("could not open audio decoder for container: " + filename);
}
/*
* And once we have that, we ask the Java Sound System to get itself ready.
*/
try {
openJavaSound(audioCoder);
} catch (LineUnavailableException ex) {
throw new RuntimeException("unable to open sound device on your system when playing back container: " + filename);
}
}
/*
* Now, we start walking through the container looking at each packet.
*/
IPacket packet = IPacket.make();
mFirstVideoTimestampInStream = Global.NO_PTS;
mSystemVideoClockStartTime = 0;
while (container.readNextPacket(packet) >= 0) {
/*
* Now we have a packet, let's see if it belongs to our video stream
*/
if (packet.getStreamIndex() == videoStreamId) {
/*
* We allocate a new picture to get the data out of Xuggler
*/
IVideoPicture picture = IVideoPicture.make(videoCoder.getPixelType(),
videoCoder.getWidth(), videoCoder.getHeight());
/*
* Now, we decode the video, checking for any errors.
*
*/
int bytesDecoded = videoCoder.decodeVideo(picture, packet, 0);
if (bytesDecoded < 0) {
throw new RuntimeException("got error decoding audio in: " + filename);
}
/*
* Some decoders will consume data in a packet, but will not be able to construct
* a full video picture yet. Therefore you should always check if you
* got a complete picture from the decoder
*/
if (picture.isComplete()) {
IVideoPicture newPic = picture;
/*
* If the resampler is not null, that means we didn't get the video in BGR24 format and
* need to convert it into BGR24 format.
*/
if (resampler != null) {
// we must resample
newPic = IVideoPicture.make(resampler.getOutputPixelFormat(), picture.getWidth(), picture.getHeight());
if (resampler.resample(newPic, picture) < 0) {
throw new RuntimeException("could not resample video from: " + filename);
}
}
if (newPic.getPixelType() != IPixelFormat.Type.BGR24) {
throw new RuntimeException("could not decode video as BGR 24 bit data in: " + filename);
}
long delay = millisecondsUntilTimeToDisplay(newPic);
// if there is no audio stream; go ahead and hold up the main thread. We'll end
// up caching fewer video pictures in memory that way.
try {
if (delay > 0) {
Thread.sleep(delay);
}
} catch (InterruptedException e) {
return;
}
// And finally, convert the picture to an image and display it
mScreen.setImage(Utils.videoPictureToImage(newPic));
}
} else if (packet.getStreamIndex() == audioStreamId) {
/*
* We allocate a set of samples with the same number of channels as the
* coder tells us is in this buffer.
*
* We also pass in a buffer size (1024 in our example), although Xuggler
* will probably allocate more space than just the 1024 (it's not important why).
*/
IAudioSamples samples = IAudioSamples.make(1024, audioCoder.getChannels());
/*
* A packet can actually contain multiple sets of samples (or frames of samples
* in audio-decoding speak). So, we may need to call decode audio multiple
* times at different offsets in the packet's data. We capture that here.
*/
int offset = 0;
/*
* Keep going until we've processed all data
*/
while (offset < packet.getSize()) {
int bytesDecoded = audioCoder.decodeAudio(samples, packet, offset);
if (bytesDecoded < 0) {
throw new RuntimeException("got error decoding audio in: " + filename);
}
offset += bytesDecoded;
/*
* Some decoder will consume data in a packet, but will not be able to construct
* a full set of samples yet. Therefore you should always check if you
* got a complete set of samples from the decoder
*/
if (samples.isComplete()) {
// note: this call will block if Java's sound buffers fill up, and we're
// okay with that. That's why we have the video "sleeping" occur
// on another thread.
playJavaSound(samples);
}
}
} else {
/*
* This packet isn't part of our video stream, so we just silently drop it.
*/
do {
} while (false);
}
}
/*
* Technically since we're exiting anyway, these will be cleaned up by
* the garbage collector... but because we're nice people and want
* to be invited places for Christmas, we're going to show how to clean up.
*/
if (videoCoder != null) {
videoCoder.close();
videoCoder = null;
}
if (audioCoder != null) {
audioCoder.close();
audioCoder = null;
}
if (container != null) {
container.close();
container = null;
}
closeJavaSound();
closeJavaVideo();
}
private static long millisecondsUntilTimeToDisplay(IVideoPicture picture) {
/**
* We could just display the images as quickly as we decode them, but it turns
* out we can decode a lot faster than you think.
*
* So instead, the following code does a poor-man's version of trying to
* match up the frame-rate requested for each IVideoPicture with the system
* clock time on your computer.
*
* Remember that all Xuggler IAudioSamples and IVideoPicture objects always
* give timestamps in Microseconds, relative to the first decoded item. If
* instead you used the packet timestamps, they can be in different units depending
* on your IContainer, and IStream and things can get hairy quickly.
*/
long millisecondsToSleep = 0;
if (mFirstVideoTimestampInStream == Global.NO_PTS) {
// This is our first time through
mFirstVideoTimestampInStream = picture.getTimeStamp();
// get the starting clock time so we can hold up frames
// until the right time.
mSystemVideoClockStartTime = System.currentTimeMillis();
millisecondsToSleep = 0;
} else {
long systemClockCurrentTime = System.currentTimeMillis();
long millisecondsClockTimeSinceStartofVideo = systemClockCurrentTime - mSystemVideoClockStartTime;
// compute how long for this frame since the first frame in the stream.
// remember that IVideoPicture and IAudioSamples timestamps are always in MICROSECONDS,
// so we divide by 1000 to get milliseconds.
long millisecondsStreamTimeSinceStartOfVideo = (picture.getTimeStamp() - mFirstVideoTimestampInStream) / 1000;
final long millisecondsTolerance = 50; // and we give ourselfs 50 ms of tolerance
millisecondsToSleep = (millisecondsStreamTimeSinceStartOfVideo
- (millisecondsClockTimeSinceStartofVideo + millisecondsTolerance));
}
return millisecondsToSleep;
}
/**
* Opens a Swing window on screen.
*/
/**
* Forces the swing thread to terminate; I'm sure there is a right
* way to do this in swing, but this works too.
*/
private static void closeJavaVideo() {
System.exit(0);
}
private static void openJavaSound(IStreamCoder aAudioCoder) throws LineUnavailableException {
AudioFormat audioFormat = new AudioFormat(aAudioCoder.getSampleRate(),
(int) IAudioSamples.findSampleBitDepth(aAudioCoder.getSampleFormat()),
aAudioCoder.getChannels(),
true, /* xuggler defaults to signed 16 bit samples */
false);
DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
mLine = (SourceDataLine) AudioSystem.getLine(info);
/**
* if that succeeded, try opening the line.
*/
mLine.open(audioFormat);
/**
* And if that succeed, start the line.
*/
mLine.start();
}
private static void playJavaSound(IAudioSamples aSamples) {
/**
* We're just going to dump all the samples into the line.
*/
byte[] rawBytes = aSamples.getData().getByteArray(0, aSamples.getSize());
mLine.write(rawBytes, 0, aSamples.getSize());
}
private static void closeJavaSound() {
if (mLine != null) {
/*
* Wait for the line to finish playing
*/
mLine.drain();
/*
* Close the line.
*/
mLine.close();
mLine = null;
}
}
private static void openJavaVideo(IStreamCoder videoCoder) {
mScreen = new VideoImage();
mScreen.setPreferredSize(new Dimension(videoCoder.getWidth(), videoCoder.getHeight()));
mScreen.setLocationRelativeTo(null);
}
}
Things I changed:
private static void openJavaVideo(IStreamCoder videoCoder) {
mScreen = new VideoImage();
mScreen.setPreferredSize(new Dimension(videoCoder.getWidth(), videoCoder.getHeight()));
mScreen.setLocationRelativeTo(null);
}
Moved openJavaVideo method into videoStream detector:
openJavaVideo(videoCoder);
Changed the first part of the main:
public static void main(String[] args) {
String filename = "http://techslides.com/demos/sample-videos/small.mp4";
// Let's make sure that we can actually convert video pixel formats.
if (!IVideoResampler.isSupported(IVideoResampler.Feature.FEATURE_COLORSPACECONVERSION)) {
throw new RuntimeException("you must install the GPL version of Xuggler (with IVideoResampler support) for this demo to work");
}
// Create a Xuggler container object
IContainer container = IContainer.make();
// Open up the container
if (container.open("http://techslides.com/demos/sample-videos/small.mp4", IContainer.Type.READ, null) < 0) {
throw new IllegalArgumentException("could not open file: " + filename);
}
Actually, the important part was:
if (container.open("http://techslides.com/demos/sample-videos/small.mp4", IContainer.Type.READ, null) < 0) {
throw new IllegalArgumentException("could not open file: " + filename);
}

Xuggle is/was one of the best:
Streaming video with Xuggler
Right now I don;t have a complete project, but I believe it had an example with its demo files.
Search google for xuggle video streaming demo or similar keywords with Xuggler. It is easy to use and support most of the known formats as it wraps FFMPEG with itself.

I came to another idea it may worth trying.
Create a JavaFX application.
Add a web browser into it, check webengine
create a template webpage which it contains a html player, or load the page in your server where it accept file id, then create a page that create a player for that file, similar to youtube, then auto play it.
This will be much better idea it you could do it.
There are sample codes for webengine and javaFX. Once you loaded a page, say, youtube or vimeo and played a video there, then sky is the limit :)

Cant use AudioFileFormat fileFormat = AudioSystem.getAudioFileFormat(file); in android?

I've been trying to figure out for the life of me how to cut up a certain amount of milliseconds out of a sample. My first problem is when I insert this code into eclipse. It tells me AudioInputStream, AudioFileFormat and FileFormat are not available. If I can't use FileFormat then i can't pass methods into other methods to get the correct calculations to save the correct amount of music. I can't even import them. But more importantly, how can I change the parameter "secondstocopy" into "millisecondstocopy" without ruining the integrity of the algorithm? Still new to java, thank you for your help!
//edit// FileFormat fileFormat = AudioSystem.getAudioFileFormat(file); It seems I can't use this bit of code for android. The getAudioFileFormat doesn't work. Nor can I calculate the fram rate because of.. int bytesPerSecond = format. * (int)format.getFrameRate(); How do I properly get the format and frame rate with media player so that I can calculate the bytesPerSecond and secondsToCopy?
AudioSystem is part of JavaSound, and JavaSound is part of the desktop JVM/SDK. JavaSound is NOT present in the Android JVM/SDK, so your old code will not compile on any current Android SDK. I get an exception using java on android (java.lang.NoClassDefFoundError), why?
How can I calculate the file format and fram rate and stuff without the javasound jvm/sdk. I can tell you now that the sounds are 44100 and 16 bit something. In the get info pane of the recording file, it says total bit rate: 128,000 with two audio channels. I'm not good with encoding stuff but I'm recording the audio with a program on mac called wiretap pro. It tells me the the parameters.
import java.io.*;
import javax.sound.sampled.*;
class AudioFileProcessor {
public static void main(String[] args) {
copyAudio("/tmp/uke.wav", "/tmp/uke-shortened.wav", ƒ, 1);
}
public static void copyAudio(String sourceFileName, String destinationFileName, int startSecond, int secondsToCopy) {
AudioInputStream inputStream = null;
AudioInputStream shortenedStream = null;
try {
File file = new File(sourceFileName);
AudioFileFormat fileFormat = AudioSystem.getAudioFileFormat(file);
AudioFormat format = fileFormat.getFormat();
inputStream = AudioSystem.getAudioInputStream(file);
int bytesPerSecond = format.getFrameSize() * (int)format.getFrameRate();
inputStream.skip(startSecond * bytesPerSecond);
long framesOfAudioToCopy = secondsToCopy * (int)format.getFrameRate();
shortenedStream = new AudioInputStream(inputStream, format, framesOfAudioToCopy);
File destinationFile = new File(destinationFileName);
AudioSystem.write(shortenedStream, fileFormat.getType(), destinationFile);
} catch (Exception e) {
println(e);
} finally {
if (inputStream != null) try { inputStream.close(); } catch (Exception e) { println(e); }
if (shortenedStream != null) try { shortenedStream.close(); } catch (Exception e) { println(e); }
}
}
}

Converting seconds into milliseconds - as a method parameter?

I've been trying to figure out for the life of me how to cut up a certain amount of milliseconds out of a sample. My first problem is when I insert this code into eclipse. It tells me AudioInputStream, AudioFileFormat and FileFormat are not available. If I can't use FileFormat then i can't pass methods into other methods to get the correct calculations to save the correct amount of music. I can't even import them. But more importantly, how can I change the parameter "secondstocopy" into "millisecondstocopy" without ruining the integrity of the algorithm? Still new to java, thank you for your help!
//edit// FileFormat fileFormat = AudioSystem.getAudioFileFormat(file); It seems I can't use this bit of code for android. The getAudioFileFormat doesn't work. Nor can I calculate the fram rate because of.. int bytesPerSecond = format. * (int)format.getFrameRate(); How do I properly get the format and frame rate with media player so that I can calculate the bytesPerSecond and secondsToCopy?
AudioSystem is part of JavaSound, and JavaSound is part of the desktop JVM/SDK. JavaSound is NOT present in the Android JVM/SDK, so your old code will not compile on any current Android SDK. I get an exception using java on android (java.lang.NoClassDefFoundError), why?
How can I calculate the file format and fram rate and stuff without the javasound jvm/sdk. I can tell you now that the sounds are 44100 and 16 bit something. In the get info pane of the recording file, it says total bit rate: 128,000 with two audio channels. I'm not good with encoding stuff but I'm recording the audio with a program on mac called wiretap pro. It tells me the the parameters.
import java.io.*;
import javax.sound.sampled.*;
class AudioFileProcessor {
public static void main(String[] args) {
copyAudio("/tmp/uke.wav", "/tmp/uke-shortened.wav", ƒ, 1);
}
public static void copyAudio(String sourceFileName, String destinationFileName, int startSecond, int secondsToCopy) {
AudioInputStream inputStream = null;
AudioInputStream shortenedStream = null;
try {
File file = new File(sourceFileName);
AudioFileFormat fileFormat = AudioSystem.getAudioFileFormat(file);
AudioFormat format = fileFormat.getFormat();
inputStream = AudioSystem.getAudioInputStream(file);
int bytesPerSecond = format.getFrameSize() * (int)format.getFrameRate();
inputStream.skip(startSecond * bytesPerSecond);
long framesOfAudioToCopy = secondsToCopy * (int)format.getFrameRate();
shortenedStream = new AudioInputStream(inputStream, format, framesOfAudioToCopy);
File destinationFile = new File(destinationFileName);
AudioSystem.write(shortenedStream, fileFormat.getType(), destinationFile);
} catch (Exception e) {
println(e);
} finally {
if (inputStream != null) try { inputStream.close(); } catch (Exception e) { println(e); }
if (shortenedStream != null) try { shortenedStream.close(); } catch (Exception e) { println(e); }
}
}
}
this code was taken from: cutting a wave file

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