private static List<Note> midiEventsToNotes(List<MidiEvent> midiEvents) {
List<Note> output = new ArrayList<>();
Predicate<MidiEvent> noteEvent = me -> me.getMessage().getStatus() >> REST_OF_STATUS_BYTE == NOTE_ON
|| me.getMessage().getStatus() >> REST_OF_STATUS_BYTE == NOTE_OFF;
List<MidiEvent> noteEvents = midiEvents.stream().filter(noteEvent).collect(Collectors.toList());
for (int i = 0; i < noteEvents.size(); i++) {
int eventType = noteEvents.get(i).getMessage().getStatus() >> REST_OF_STATUS_BYTE;
if (eventType == NOTE_ON) {
byte pitch = noteEvents.get(i).getMessage().getMessage()[1];
int startBeat = (int) (noteEvents.get(i).getTick() / ticksPerBeat);
for (MidiEvent ne: noteEvents) {
int pairType = ne.getMessage().getStatus() >> REST_OF_STATUS_BYTE;
byte pitch2 = ne.getMessage().getMessage()[1];
if (pairType == NOTE_OFF && pitch == pitch2) {
int value = (int) (ne.getTick() / ticksPerBeat - startBeat);
int pianoKey = pitch - MIDI_A0_VALUE;
output.add(new Note(startBeat, value, pianoKey));
break;
}
}
}
}
return output;
}
I'm trying to read a MIDI file (convert data in the file to data that is handled by my application's model). Here's the method in question. It takes a list of MidiEvent, which should just be a list of all MidiEvent in the file's sequence. The method should output a list of Note, Note being a class in the model. First the method filters the list down to only note-on and note-off events. Then, for each event, if the event is a note-on, it tries to pair it with the corresponding note-off and instantiate a Note.
I've been testing it with midi files containing only one note. As expected, the debugger tells me that there are two elements in the filtered list noteEvents, but they are somehow both note-on events (they have the same status byte), and obviously the method doesn't work because of that. Is there something wrong with the method, or is it how Java is converting the file to events, or are the midi files just bad?
The MIDI Specification says:
MIDI provides two roughly equivalent means of turning off a note (voice). A note may be turned off either by sending a Note-Off message for the same note number and channel, or by sending a Note-On message for that note and channel with a velocity value of zero. The advantage to using "Note-On at zero velocity" is that it can avoid sending additional status bytes when Running Status is employed.
Due to this efficiency, sending Note-On messages with velocity values of zero is the most commonly used method.
Related
I am managing audio capturing and playing using java sound API (targetDataLine and sourceDataLine). Now suppose in a conference environment, one participant's audio queue size got greater than jitter size (due to processing or network) and I want to fast forward the audio bytes I have of that participant to make it shorter than jitter size.
How can I fast forward the audio byte array of that participant?
I can't do it during playing as normally Player thread just deque 1 frame from every participant's queue and mix it for playing. The only way I can get that is if I deque more than 1 frame of that participant and mix(?) it for fast-forwarding before mixing it with other participants 1 dequeued frame for playing?
Thanks in advance for any kind of help or advice.
There are two ways to speed up the playback that I know of. In one case, the faster pace creates a rise in pitch. The coding for this is relatively easy. In the other case, pitch is kept constant, but it involves a technique of working with sound granules (granular synthesis), and is harder to explain.
For the situation where maintaining the same pitch is not a concern, the basic plan is as follows: instead of advancing by single frames, advance by a frame + a small increment. For example, let's say that advancing 1.1 frames over a course of 44000 frames is sufficient to catch you up. (That would also mean that the pitch increase would be about 1/10 of an octave.)
To advance a "fractional" frame, you first have to convert the bytes of the two bracketing frames to PCM. Then, use linear interpolation to get the intermediate value. Then convert that intermediate value back to bytes for the output line.
For example, if you are advancing from frame[0] to frame["1.1"] you will need to know the PCM for frame[1] and frame[2]. The intermediate value can be calculated using a weighted average:
value = PCM[1] * 9/10 + PCM[2] * 1/10
I think it might be good to make the amount by which you advance change gradually. Take a few dozen frames to ramp up the increment and allow time to ramp down again when returning to normal dequeuing. If you suddenly change the rate at which you are reading the audio data, it is possible to introduce a discontinuity that will be heard as a click.
I have used this basic plan for dynamic control of playback speed, but I haven't had the experience of employing it for the situation that you are describing. Regulating the variable speed could be tricky if you also are trying to enforce keeping the transitions smooth.
The basic idea for using granules involves obtaining contiguous PCM (I'm not clear what the optimum number of frames would be for voice, 1 to 50 millis is cited as commonly being used with this technique in synthesis), and giving it a volume envelope that allows you to mix sequential granules end-to-end (they must overlap).
I think the envelopes for the granules make use of a Hann function or Hamming window--but I'm not clear on the details, such as the overlapping placement of the granules so that they mix/transition smoothly. I've only dabbled, and I'm going to assume folks at Signal Processing will be the best bet for advice on how to code this.
I found a fantastic git repo (sonic library, mainly for audio player) which actually does exactly what I wanted with so much controls. I can input a whole .wav file or even chunks of audio byte arrays and after processing, we can get speed up play experience and so more. For real time processing I actually called this on every chunk of audio byte array.
I found another way/algo to detect whether a audio chunk/byte array is voice or not and after depending on it's result, I can simply ignore playing non voice packets which gives us around 1.5x speedup with less processing.
public class DTHVAD {
public static final int INITIAL_EMIN = 100;
public static final double INITIAL_DELTAJ = 1.0001;
private static boolean isFirstFrame;
private static double Emax;
private static double Emin;
private static int inactiveFrameCounter;
private static double Lamda; //
private static double DeltaJ;
static {
initDTH();
}
private static void initDTH() {
Emax = 0;
Emin = 0;
isFirstFrame = true;
Lamda = 0.950; // range is 0.950---0.999
DeltaJ = 1.0001;
}
public static boolean isAllSilence(short[] samples, int length) {
boolean r = true;
for (int l = 0; l < length; l += 80) {
if (!isSilence(samples, l, l+80)) {
r = false;
break;
}
}
return r;
}
public static boolean isSilence(short[] samples, int offset, int length) {
boolean isSilenceR = false;
long energy = energyRMSE(samples, offset, length);
// printf("en=%ld\n",energy);
if (isFirstFrame) {
Emax = energy;
Emin = INITIAL_EMIN;
isFirstFrame = false;
}
if (energy > Emax) {
Emax = energy;
}
if (energy < Emin) {
if ((int) energy == 0) {
Emin = INITIAL_EMIN;
} else {
Emin = energy;
}
DeltaJ = INITIAL_DELTAJ; // Resetting DeltaJ with initial value
} else {
DeltaJ = DeltaJ * 1.0001;
}
long thresshold = (long) ((1 - Lamda) * Emax + Lamda * Emin);
// printf("e=%ld,Emin=%f, Emax=%f, thres=%ld\n",energy,Emin,Emax,thresshold);
Lamda = (Emax - Emin) / Emax;
if (energy > thresshold) {
isSilenceR = false; // voice marking
} else {
isSilenceR = true; // noise marking
}
Emin = Emin * DeltaJ;
return isSilenceR;
}
private static long energyRMSE(short[] samples, int offset, int length) {
double cEnergy = 0;
float reversOfN = (float) 1 / length;
long step = 0;
for (int i = offset; i < length; i++) {
step = samples[i] * samples[i]; // x*x/N=
// printf("step=%ld cEng=%ld\n",step,cEnergy);
cEnergy += (long) ((float) step * reversOfN);// for length =80
// reverseOfN=0.0125
}
cEnergy = Math.pow(cEnergy, 0.5);
return (long) cEnergy;
}
}
Here I can convert my byte array to short array and detect whether it is voice or non voice by
frame.silence = DTHVAD.isSilence(encodeShortBuffer, 0, shortLen);
I am writing some code that intends to take a Wave file, and write it out to and AudioTrack in mode stream. This is a minimum viable test to get AudioTrack stream mode working.
But once I write some buffer of audio to the AudioTrack, and subsequently call play(), the method getPlaybackHeadPosition() continually returns 0.
EDIT: If I ignore my available frames check, and just continually write buffers to the AudioTrack, the write method returns 0 (after the the first buffer write), indicating that it simply did not write any more audio. So it seems that the AudioTrack just doesn't want to start playing.
My code is properly priming the audiotrack. The play method is not throwing any exceptions, so I am not sure what is going wrong.
When stepping through the code, everything on my end is exactly how I anticipate it, so I am thinking somehow I have the AudioTrack configured wrong.
I am running on an emulator, but I don't think that should be an issue.
The WavFile class I am using is a vetted class that I have up and running reliably in lots of Java projects, it is tested to work well.
Observe the following log write, which is a snippet from the larger chunk of code. This log write is never hitting...
if (headPosition > 0)
Log.e("headPosition is greater than zero!!");
..
public static void writeToAudioTrackStream(final WavFile wave)
{
Log.e("writeToAudioTrackStream");
Thread thread = new Thread()
{
public void run()
{
try {
final float[] data = wave.getData();
int format = -1;
if (wave.getChannel() == 1)
format = AudioFormat.CHANNEL_OUT_MONO;
else if (wave.getChannel() == 2)
format = AudioFormat.CHANNEL_OUT_STEREO;
else
throw new RuntimeException("writeToAudioTrackStatic() - unsupported number of channels value = "+wave.getChannel());
final int bufferSizeInFrames = 2048;
final int bytesPerSmp = wave.getBytesPerSmp();
final int bufferSizeInBytes = bufferSizeInFrames * bytesPerSmp * wave.getChannel();
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, wave.getSmpRate(),
format,
AudioFormat.ENCODING_PCM_FLOAT,
bufferSizeInBytes,
AudioTrack.MODE_STREAM);
int index = 0;
float[] buffer = new float[bufferSizeInFrames * wave.getChannel()];
boolean started = false;
int framesWritten = 0;
while (index < data.length) {
// calculate the available space in the buffer
int headPosition = audioTrack.getPlaybackHeadPosition();
if (headPosition > 0)
Log.e("headPosition is greater than zero!!");
int framesInBuffer = framesWritten - headPosition;
int availableFrames = bufferSizeInFrames - framesInBuffer;
// once the buffer has no space, the prime is done, so start playing
if (availableFrames == 0) {
if (!started) {
audioTrack.play();
started = true;
}
continue;
}
int endOffset = availableFrames * wave.getChannel();
for (int i = 0; i < endOffset; i++)
buffer[i] = data[index + i];
int samplesWritten = audioTrack.write(buffer , 0 , endOffset , AudioTrack.WRITE_BLOCKING);
// could return error values
if (samplesWritten < 0)
throw new RuntimeException("AudioTrack write error.");
framesWritten += samplesWritten / wave.getChannel();
index = endOffset;
}
}
catch (Exception e) {
Log.e(e.toString());
}
}
};
thread.start();
}
Per the documentation,
For portability, an application should prime the data path to the maximum allowed by writing data until the write() method returns a short transfer count. This allows play() to start immediately, and reduces the chance of underrun.
With a strict reading, this might be seen to contradict the earlier statement:
...you can optionally prime the data path prior to calling play(), by writing up to bufferSizeInBytes...
(emphasis mine), but the intent is clear enough: You're supposed to get a short write first.
This is just to get play started. Once that takes place, you can, in fact, use
getPlaybackHeadPosition() to determine when more space is available. I've used that technique successfully in my own code, on many different devices/API levels.
As an aside: You should be prepared for getPlaybackHeadPosition() to change only in large increments (if I remember correctly, it's getMinBufferSize()/2). This is the max resolution available from the system; onMarkerReached() cannot be used to do any better.
I'm trying to get the number of Tracks of a MIDI sequence:
File file = new File(strSource);
Sequence sequence = MidiSystem.getSequence(file);
int numTracks = sequence.getTracks().length;
... where strSource is the full path+file name of my .mid file.
numTracks is 1, but the .mid file has 16 tracks (as i can see when i open it in another MIDI editor). The file type is 0.
I read somewhere that type-0 files can't have multiple tracks for the same channel. In this case all tracks are forced into a single track. Is that correct? How can I avoid that?
It seems you're right, type-0 files hold multiple tracks in just one.
Here you have some info.
Isn't possible to extract each separate track from a type 0 file.
Check MIDI file type, if an external MIDI editor can detect multiple tracks, it can be a type 1 or type 2 file, even if extension doesn't match.
I looked at the file with a hex tool ...
It actually has only one track.
The other editor creates the multiple Tracks by itself. It seems to search for program change messages and then put the events into new tracks.
I have programmed a small function that converts a Type # 0 sequence to a Type # 1 sequence
/**
* Make multiple tracks from file0 track
* #param in : Sequence with single track
* #return Multiple track sequence
*/
private Sequence extractFile0Tracks (Sequence in) throws InvalidMidiDataException
{
Track inTrack = in.getTracks()[0];
HashMap<Integer, ArrayList<MidiEvent>> msgMap = new HashMap<>();
// Distribute events per channel to ArrayList map
for (int i = 0; i < inTrack.size(); i++)
{
MidiEvent event = inTrack.get(i);
MidiMessage message = event.getMessage();
if (message instanceof ShortMessage)
{
ShortMessage sm = (ShortMessage) message;
int channel = sm.getChannel() + 1;
ArrayList<MidiEvent> msgList = msgMap.computeIfAbsent(channel, k -> new ArrayList<>());
msgList.add(event);
}
}
// Create sequence with multiple tracks
Sequence newSeq = new Sequence(in.getDivisionType(), in.getResolution());
for (ArrayList<MidiEvent> msgList : msgMap.values())
{
Track tr = newSeq.createTrack();
for (MidiEvent m1 : msgList)
tr.add(m1);
}
return newSeq;
}
I have a method which takes a parameter which is Partition enum. This method will be called by multiple background threads (15 max) around same time period by passing different value of partition. Here dataHoldersByPartition is a map of Partition and ConcurrentLinkedQueue<DataHolder>.
private final ImmutableMap<Partition, ConcurrentLinkedQueue<DataHolder>> dataHoldersByPartition;
//... some code to populate entry in `dataHoldersByPartition`
private void validateAndSend(final Partition partition) {
ConcurrentLinkedQueue<DataHolder> dataHolders = dataHoldersByPartition.get(partition);
Map<byte[], byte[]> clientKeyBytesAndProcessBytesHolder = new HashMap<>();
int totalSize = 0;
DataHolder dataHolder;
while ((dataHolder = dataHolders.poll()) != null) {
byte[] clientKeyBytes = dataHolder.getClientKey().getBytes(StandardCharsets.UTF_8);
if (clientKeyBytes.length > 255)
continue;
byte[] processBytes = dataHolder.getProcessBytes();
int clientKeyLength = clientKeyBytes.length;
int processBytesLength = processBytes.length;
int additionalLength = clientKeyLength + processBytesLength;
if (totalSize + additionalLength > 50000) {
Message message = new Message(clientKeyBytesAndProcessBytesHolder, partition);
// here size of `message.serialize()` byte array should always be less than 50k at all cost
sendToDatabase(message.getAddress(), message.serialize());
clientKeyBytesAndProcessBytesHolder = new HashMap<>();
totalSize = 0;
}
clientKeyBytesAndProcessBytesHolder.put(clientKeyBytes, processBytes);
totalSize += additionalLength;
}
// calling again with remaining values only if clientKeyBytesAndProcessBytesHolder is not empty
if(!clientKeyBytesAndProcessBytesHolder.isEmpty()) {
Message message = new Message(partition, clientKeyBytesAndProcessBytesHolder);
// here size of `message.serialize()` byte array should always be less than 50k at all cost
sendToDatabase(message.getAddress(), message.serialize());
}
}
And below is my Message class:
public final class Message {
private final byte dataCenter;
private final byte recordVersion;
private final Map<byte[], byte[]> clientKeyBytesAndProcessBytesHolder;
private final long address;
private final long addressFrom;
private final long addressOrigin;
private final byte recordsPartition;
private final byte replicated;
public Message(Map<byte[], byte[]> clientKeyBytesAndProcessBytesHolder, Partition recordPartition) {
this.clientKeyBytesAndProcessBytesHolder = clientKeyBytesAndProcessBytesHolder;
this.recordsPartition = (byte) recordPartition.getPartition();
this.dataCenter = Utils.CURRENT_LOCATION.get().datacenter();
this.recordVersion = 1;
this.replicated = 0;
long packedAddress = new Data().packAddress();
this.address = packedAddress;
this.addressFrom = 0L;
this.addressOrigin = packedAddress;
}
// Output of this method should always be less than 50k always
public byte[] serialize() {
int bufferCapacity = getBufferCapacity(clientKeyBytesAndProcessBytesHolder); // 36 + dataSize + 1 + 1 + keyLength + 8 + 2;
ByteBuffer byteBuffer = ByteBuffer.allocate(bufferCapacity).order(ByteOrder.BIG_ENDIAN);
// header layout
byteBuffer.put(dataCenter).put(recordVersion).putInt(clientKeyBytesAndProcessBytesHolder.size())
.putInt(bufferCapacity).putLong(address).putLong(addressFrom).putLong(addressOrigin)
.put(recordsPartition).put(replicated);
// now the data layout
for (Map.Entry<byte[], byte[]> entry : clientKeyBytesAndProcessBytesHolder.entrySet()) {
byte keyType = 0;
byte[] key = entry.getKey();
byte[] value = entry.getValue();
byte keyLength = (byte) key.length;
short valueLength = (short) value.length;
ByteBuffer dataBuffer = ByteBuffer.wrap(value);
long timestamp = valueLength > 10 ? dataBuffer.getLong(2) : System.currentTimeMillis();
byteBuffer.put(keyType).put(keyLength).put(key).putLong(timestamp).putShort(valueLength)
.put(value);
}
return byteBuffer.array();
}
private int getBufferCapacity(Map<byte[], byte[]> clientKeyBytesAndProcessBytesHolder) {
int size = 36;
for (Entry<byte[], byte[]> entry : clientKeyBytesAndProcessBytesHolder.entrySet()) {
size += 1 + 1 + 8 + 2;
size += entry.getKey().length;
size += entry.getValue().length;
}
return size;
}
// getters and to string method here
}
Basically, what I have to make sure is whenever the sendToDatabase method is called, size of message.serialize() byte array should always be less than 50k at all cost. My sendToDatabase method sends byte array coming out from serialize method. And because of that condition I am doing below validation plus few other stuff. In the method, I will iterate dataHolders CLQ and I will extract clientKeyBytes and processBytes from it. Here is the validation I am doing:
If the clientKeyBytes length is greater than 255 then I will skip it and continue iterating.
I will keep incrementing the totalSize variable which will be the sum of clientKeyLength and processBytesLength, and this totalSize length should always be less than 50000 bytes.
As soon as it reaches the 50000 limit, I will send the clientKeyBytesAndProcessBytesHolder map to the sendToDatabase method and clear out the map, reset totalSize to 0 and start populating again.
If it doesn't reaches that limit and dataHolders got empty, then it will send whatever it has.
I believe there is some bug in my current code because of which maybe some records are not being sent properly or dropped somewhere because of my condition and I am not able to figure this out. Looks like to properly achieve this 50k condition I may have to use getBufferCapacity method to correctly figure out the size before calling sendToDatabase method?
I checked your code, its look good as per your logic. As you said it will always store the information which is less than 50K but it will actually store information till 50K. To make it less than 50K you have to change the if condition to if (totalSize + additionalLength >= 50000).
If your codes still not fulfilling your requirement i.e. storing information when totalSize + additionalLength is greater than 50k I can advise you few thinks.
As more than 50 threads call this method you need to consider two section in your codes to be synchronize.
One is global variable which is a container dataHoldersByPartition object. If multiple concurrent and parallel searches happened in this container object, outcome might not be perfect. Just check whether container type is synchronized or not. If not make this block like below:-
synchronized(this){
ConcurrentLinkedQueue<DataHolder> dataHolders = dataHoldersByPartition.get(partition);
}
Now, I can give only two suggestion to fix this issue. One is instead of if (totalSize + additionalLength > 50000) this you can check the size of the object clientKeyBytesAndProcessBytesHolder if(sizeof(clientKeyBytesAndProcessBytesHolder) >= 50000) (check appropriate method for sizeof in java). And second one is narrow down the area to check whether it is a side effect of multithreading or not. All these suggestion are to find out the area where exactly problem is and fix should be from your end only.
First check whether you method validateAndSend is exactly satisfying your requirement or not. For that synchronize whole validateAndSend method first and check whether everything fine or still have the same result. If still have the same result that means it is not because of multithreading but your coding is not as per requirement. If its work fine that means it is a problem of multithreading. If method synchronization is fixing your issue but degrade the performance you just remove the synchronization from it and concentrate every small block of your code which might cause the issue and make it synchronize block and remove if still not fixing your issue. Like that finally you locate the block of code which is actually creating the issue and leave it as synchronize to fix it finally.
For example first attempt:-
`private synchronize void validateAndSend`
Second attempts: Remove synchronize key words from the method and do the below step:-
synchronize(this){
Message message = new Message(clientKeyBytesAndProcessBytesHolder, partition);
sendToDatabase(message.getAddress(), message.serialize());
}
If you think that I did not correctly understand you please let me know.
In your validateAndSend I would put whole data to the queue, and do whole processing in separate thread. Please consider command model. That way all threads are going to put their load on queue. Consumer thread has all the data, all the information in place, and can process it quite effectively. The only complicated part is sending response / result back to calling thread. Since in your case that is not a problem - the better. There are some more benefits of this pattern - please look at netflix/hystrix.
I need to write a client that communicates with server using socket. the message protocol is in json format. the server will push multiple json blocks to client on demand.
The message is like this:
{"a": 1, "b": { "c": 1}}{"a": 1, "b": { "c": 1}}...
You can see that there is no separator or identifier between the json blocks.
The json parsers I can find (like fastjson, jackson) are all only able to deal the stream as a whole json block, even the stream api they provide. when I use these api to parse the stream, they will throw an exception at the end of first json block, said the next token "{" is invalid.
Is there a json parser in java can deal with my problem? or is there other way to solve this problem?
Finally it seams no JSON parser in java suit for my situation.
I am using netty to build a network application. for nio, when there is data coming from network, the decode method in ByteToMessageDecoder is called.
In this method I need to find out JSON block from ByteBuf.
Since there is no available JSON parser, I wrote a method to split the JSON Block from ByteBuf.
public static void extractJsonBlocks(ByteBuf buf, List<Object> out) throws UnsupportedEncodingException {
// the total bytes that can read from ByteBuf
int readable = buf.readableBytes();
int bracketDepth = 0;
// when found a json block, this value will be set
int offset = 0;
// whether current character is in a string value
boolean inStr = false;
// a temporary bytes buf for store json block
byte[] data = new byte[readable];
// loop all the coming data
for (int i = 0; i < readable; i++) {
// read from ByteBuf
byte b = buf.readByte();
// put it in the buffer, be care of the offset
data[i - offset] = b;
if (b == SYM_L_BRACKET && !inStr) {
// if it a left bracket and not in a string value
bracketDepth++;
} else if (b == SYM_R_BRACKET && !inStr) {
// if it a right bracket and not in a string value
if (bracketDepth == 1) {
// if current bracket depth is 1, means found a whole json block
out.add(new String(data, "utf-8").trim());
// create a new buffer
data = new byte[readable - offset];
// update the offset
offset = i;
// reset the bracket depth
bracketDepth = 0;
} else {
bracketDepth--;
}
} else if (b == SYM_QUOTE) {
// when find a quote, we need see whether preview character is escape.
byte prev = i == 0 ? 0 : data[i - 1 - offset];
if (prev != SYM_ESCAPE) {
inStr = !inStr;
}
}
}
// finally there may still be some data left in the ByteBuf, that can not form a json block, they should be used to combine with the following datas
// so we need to reset the reader index to the first byte of the left data
// and discard the data used for json blocks
buf.readerIndex(offset == 0 ? offset : offset + 1);
buf.discardReadBytes();
}
maybe this is a not perfect parser, but it works well for my application now.
You can do it with Genson.
First configure it to allow "permissive" parsing and then deserialize the values in an iterator. Genson will parse the objects one by one as you call next on the iterator. Thus you can parse this way very large inputs.
Genson genson = new GensonBuilder().usePermissiveParsing(true).create();
ObjectReader reader = genson.createReader(inputStream);
Iterator<SomeObject> iterator = genson.deserializeValues(reader, GenericType.of(SomeObject.class));
This part of the API is a bit verbose as the use case is not so common.
UPDATE In Genson 1.4 usePermissiveParsing has been removed in favor of accepting root values not wrapped in an array by default. See https://github.com/owlike/genson/issues/78