Develop Reference

Android: confused about how to get amplitude of a frequency generated and played by AudioTrack

In my app I generate list of sounds with different frequencies, for example 1000Hz, 2000Hz, 4000Hz ... for left and right channel:
private AudioTrack generateTone(Ear ear) {
// fill out the array
int numSamples = SAMPLE_RATE * getDurationInSeconds();
double[] sample = new double[numSamples];
byte[] generatedSnd = new byte[2 * numSamples];
for (int i = 0; i < numSamples; ++i) {
sample[i] = Math.sin(2 * Math.PI * i / (SAMPLE_RATE / getLatestFreqInHz()));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
for (final double dVal : sample) {
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
SAMPLE_RATE, AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, numSamples,
AudioTrack.MODE_STATIC);
audioTrack.write(generatedSnd, 0, generatedSnd.length);
int channel;
if (ear == Ear.LEFT) {
audioTrack.setStereoVolume(1.0f, 0.0f);
} else if (ear == Ear.RIGHT) {
audioTrack.setStereoVolume(0.0f, 1.0f);
}
return audioTrack;
}
I use this code to control volume:
audioManager = (AudioManager) context.getApplicationContext().getSystemService(Context.AUDIO_SERVICE);
MAX_VOLUME = audioManager.getStreamMaxVolume(STREAM_MUSIC);
MIN_VOLUME = audioManager.getStreamMinVolume(STREAM_MUSIC);
APP_DEFAULT_VOLUME = (MAX_VOLUME + MIN_VOLUME) /2 ;
#Override
public void setToAppDefaultVolume() {
Timber.tag(TAG).d("Resetting to app default sound volume.");
audioManager.setStreamVolume(STREAM_MUSIC, APP_DEFAULT_VOLUME, 0);
}
#Override
public void increaseVolume() {
if (!isReachedMaxVolume()) {
Timber.tag(TAG).d("Increasing device sound volume from %d to %d", currentVolumeLevel(), currentVolumeLevel() + STEP);
audioManager.setStreamVolume(STREAM_MUSIC, currentVolumeLevel() + STEP, 0);
} else {
Timber.tag(TAG).d("Reached the maximum device volume.");
}
}
#Override
public void decreaseVolume() {
if (!isReachedMinVolume()) {
Timber.tag(TAG).d("Decreasing device sound volume from %d to %d", currentVolumeLevel(), currentVolumeLevel() - STEP);
audioManager.setStreamVolume(STREAM_MUSIC, currentVolumeLevel() - STEP, 0);
} else {
Timber.tag(TAG).d("Reached the preferred minimum volume");
}
}
I start playing each tone (frequency) with APP_DEFAULT_VOLUME and gradually increase tone. When user confirm he/she heard specific tone with specific volume, I want to calculate it's amplitude and log it for latter so I can review user hearing ...
But i have no clue how to do so!
All solutions I found was about reading data from microphone and calculate the amplitude to visualize it on screen ...
My scenario is much simpler. I have device volume recorded, frequency is fixed and recorded, and Audio is generated dynamically and is not a file.
Will anyone help me with this scenario ?
Thanks in advance.

Android Creating Video from Screen Scraping: Why is output Image wonky?

Update #6 Discovered I was accessing RGB values improperly. I assumed I was accessing data from an Int[], but was instead accessing byte information from a Byte[]. Changed to accessing from Int[] and get the following image:
Update #5 Adding code used to get RGBA ByteBuffer for reference
private void screenScrape() {
Log.d(TAG, "In screenScrape");
//read pixels from frame buffer into PBO (GL_PIXEL_PACK_BUFFER)
mSurface.queueEvent(new Runnable() {
#Override
public void run() {
Log.d(TAG, "In Screen Scrape 1");
//generate and bind buffer ID
GLES30.glGenBuffers(1, pboIds);
checkGlError("Gen Buffers");
GLES30.glBindBuffer(GLES30.GL_PIXEL_PACK_BUFFER, pboIds.get(0));
checkGlError("Bind Buffers");
//creates and initializes data store for PBO. Any pre-existing data store is deleted
GLES30.glBufferData(GLES30.GL_PIXEL_PACK_BUFFER, (mWidth * mHeight * 4), null, GLES30.GL_STATIC_READ);
checkGlError("Buffer Data");
//glReadPixelsPBO(0,0,w,h,GLES30.GL_RGB,GLES30.GL_UNSIGNED_SHORT_5_6_5,0);
glReadPixelsPBO(0, 0, mWidth, mHeight, GLES30.GL_RGBA, GLES30.GL_UNSIGNED_BYTE, 0);
checkGlError("Read Pixels");
//GLES30.glReadPixels(0,0,w,h,GLES30.GL_RGBA,GLES30.GL_UNSIGNED_BYTE,intBuffer);
}
});
//map PBO data into client address space
mSurface.queueEvent(new Runnable() {
#Override
public void run() {
Log.d(TAG, "In Screen Scrape 2");
//read pixels from PBO into a byte buffer for processing. Unmap buffer for use in next pass
mapBuffer = ((ByteBuffer) GLES30.glMapBufferRange(GLES30.GL_PIXEL_PACK_BUFFER, 0, 4 * mWidth * mHeight, GLES30.GL_MAP_READ_BIT)).order(ByteOrder.nativeOrder());
checkGlError("Map Buffer");
GLES30.glUnmapBuffer(GLES30.GL_PIXEL_PACK_BUFFER);
checkGlError("Unmap Buffer");
isByteBufferEmpty(mapBuffer, "MAP BUFFER");
convertColorSpaceByteArray(mapBuffer);
mapBuffer.clear();
}
});
}
Update #4 For reference, here is the original image to compare against.
Update #3 This is the output image after interleaving all U/V data into a single array and passing it to the Image object at inputImagePlanes[1]; inputImagePlanes[2]; is unused;
The next image is the same interleaved UV data, but we load this into inputImagePlanes[2]; instead of inputImagePlanes[1];
Update #2 This is the output image after padding the U/V buffers with a zero in between each byte of 'real' data. uArray[uvByteIndex] = (byte) 0;
Update #1 As suggested by a comment, here are the row and pixel strides I get from calling getPixelStride and getRowStride
Y Plane Pixel Stride = 1, Row Stride = 960
U Plane Pixel Stride = 2, Row Stride = 960
V Plane Pixel Stride = 2, Row Stride = 960
The goal of my application is to read pixels out from the screen, compress them, and then send that h264 stream over WiFi to be played be a receiver.
Currently I'm using the MediaMuxer class to convert the raw h264 stream to an MP4, and then save it to file. However the end result video is messed up and I can't figure out why. Lets walk through some of processing and see if we can find anything that jumps out.
Step 1 Set up the encoder. I'm currently taking screen images once every 2 seconds, and using "video/avc" for MIME_TYPE
//create codec for compression
try {
mCodec = MediaCodec.createEncoderByType(MIME_TYPE);
} catch (IOException e) {
Log.d(TAG, "FAILED: Initializing Media Codec");
}
//set up format for codec
MediaFormat mFormat = MediaFormat.createVideoFormat(MIME_TYPE, mWidth, mHeight);
mFormat.setInteger(MediaFormat.KEY_COLOR_FORMAT, MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420Flexible);
mFormat.setInteger(MediaFormat.KEY_BIT_RATE, 16000000);
mFormat.setInteger(MediaFormat.KEY_FRAME_RATE, 1/2);
mFormat.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, 5);
Step 2 Read pixels out from screen. This is done using openGL ES, and the pixels are read out in RGBA format. (I've confirmed this part to be working)
Step 3 Convert the RGBA pixels to YUV420 (IYUV) format. This is done using the following method. Note that I have 2 methods for encoding called at the end of this method.
private void convertColorSpaceByteArray(ByteBuffer rgbBuffer) {
long startTime = System.currentTimeMillis();
Log.d(TAG, "In convertColorspace");
final int frameSize = mWidth * mHeight;
final int chromaSize = frameSize / 4;
byte[] rgbByteArray = new byte[rgbBuffer.remaining()];
rgbBuffer.get(rgbByteArray);
byte[] yuvByteArray = new byte[inputBufferSize];
Log.d(TAG, "Input Buffer size = " + inputBufferSize);
byte[] yArray = new byte[frameSize];
byte[] uArray = new byte[(frameSize / 4)];
byte[] vArray = new byte[(frameSize / 4)];
isByteBufferEmpty(rgbBuffer, "RGB BUFFER");
int yIndex = 0;
int uIndex = frameSize;
int vIndex = frameSize + chromaSize;
int yByteIndex = 0;
int uvByteIndex = 0;
int R, G, B, Y, U, V;
int index = 0;
//this loop controls the rows
for (int i = 0; i < mHeight; i++) {
//this loop controls the columns
for (int j = 0; j < mWidth; j++) {
R = (rgbByteArray[index] & 0xff0000) >> 16;
G = (rgbByteArray[index] & 0xff00) >> 8;
B = (rgbByteArray[index] & 0xff);
Y = ((66 * R + 129 * G + 25 * B + 128) >> 8) + 16;
U = ((-38 * R - 74 * G + 112 * B + 128) >> 8) + 128;
V = ((112 * R - 94 * G - 18 * B + 128) >> 8) + 128;
//clamp and load in the Y data
yuvByteArray[yIndex++] = (byte) ((Y < 16) ? 16 : ((Y > 235) ? 235 : Y));
yArray[yByteIndex] = (byte) ((Y < 16) ? 16 : ((Y > 235) ? 235 : Y));
yByteIndex++;
if (i % 2 == 0 && index % 2 == 0) {
//clamp and load in the U & V data
yuvByteArray[uIndex++] = (byte) ((U < 16) ? 16 : ((U > 239) ? 239 : U));
yuvByteArray[vIndex++] = (byte) ((V < 16) ? 16 : ((V > 239) ? 239 : V));
uArray[uvByteIndex] = (byte) ((U < 16) ? 16 : ((U > 239) ? 239 : U));
vArray[uvByteIndex] = (byte) ((V < 16) ? 16 : ((V > 239) ? 239 : V));
uvByteIndex++;
}
index++;
}
}
encodeVideoFromImage(yArray, uArray, vArray);
encodeVideoFromBuffer(yuvByteArray);
}
Step 4 Encode the data! I currently have two different ways of doing this, and each has a different output. One uses a ByteBuffer returned from MediaCodec.getInputBuffer();, the other uses an Image returned from MediaCodec.getInputImage();
Encoding using ByteBuffer
private void encodeVideoFromBuffer(byte[] yuvData) {
Log.d(TAG, "In encodeVideo");
int inputSize = 0;
//create index for input buffer
inputBufferIndex = mCodec.dequeueInputBuffer(0);
//create the input buffer for submission to encoder
ByteBuffer inputBuffer = mCodec.getInputBuffer(inputBufferIndex);
//clear, then copy yuv buffer into the input buffer
inputBuffer.clear();
inputBuffer.put(yuvData);
//flip buffer before reading data out of it
inputBuffer.flip();
mCodec.queueInputBuffer(inputBufferIndex, 0, inputBuffer.remaining(), presentationTime, 0);
presentationTime += MICROSECONDS_BETWEEN_FRAMES;
sendToWifi();
}
And the associated output image (note: I took a screenshot of the MP4)
Encoding using Image
private void encodeVideoFromImage(byte[] yToEncode, byte[] uToEncode, byte[]vToEncode) {
Log.d(TAG, "In encodeVideo");
int inputSize = 0;
//create index for input buffer
inputBufferIndex = mCodec.dequeueInputBuffer(0);
//create the input buffer for submission to encoder
Image inputImage = mCodec.getInputImage(inputBufferIndex);
Image.Plane[] inputImagePlanes = inputImage.getPlanes();
ByteBuffer yPlaneBuffer = inputImagePlanes[0].getBuffer();
ByteBuffer uPlaneBuffer = inputImagePlanes[1].getBuffer();
ByteBuffer vPlaneBuffer = inputImagePlanes[2].getBuffer();
yPlaneBuffer.put(yToEncode);
uPlaneBuffer.put(uToEncode);
vPlaneBuffer.put(vToEncode);
yPlaneBuffer.flip();
uPlaneBuffer.flip();
vPlaneBuffer.flip();
mCodec.queueInputBuffer(inputBufferIndex, 0, inputBufferSize, presentationTime, 0);
presentationTime += MICROSECONDS_BETWEEN_FRAMES;
sendToWifi();
}
And the associated output image (note: I took a screenshot of the MP4)
Step 5 Convert H264 Stream to MP4. Finally I grab the output buffer from the codec, and use MediaMuxer to convert the raw h264 stream to an MP4 that I can play and test for correctness
private void sendToWifi() {
Log.d(TAG, "In sendToWifi");
MediaCodec.BufferInfo mBufferInfo = new MediaCodec.BufferInfo();
//Check to see if encoder has output before proceeding
boolean waitingForOutput = true;
boolean outputHasChanged = false;
int outputBufferIndex = 0;
while (waitingForOutput) {
//access the output buffer from the codec
outputBufferIndex = mCodec.dequeueOutputBuffer(mBufferInfo, -1);
if (outputBufferIndex == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
outputFormat = mCodec.getOutputFormat();
outputHasChanged = true;
Log.d(TAG, "OUTPUT FORMAT HAS CHANGED");
}
if (outputBufferIndex >= 0) {
waitingForOutput = false;
}
}
//this buffer now contains the compressed YUV data, ready to be sent over WiFi
ByteBuffer outputBuffer = mCodec.getOutputBuffer(outputBufferIndex);
//adjust output buffer position and limit. As of API 19, this is not automatic
if(mBufferInfo.size != 0) {
outputBuffer.position(mBufferInfo.offset);
outputBuffer.limit(mBufferInfo.offset + mBufferInfo.size);
}
////////////////////////////////FOR DEGBUG/////////////////////////////
if (muxerNotStarted && outputHasChanged) {
//set up track
mTrackIndex = mMuxer.addTrack(outputFormat);
mMuxer.start();
muxerNotStarted = false;
}
if (!muxerNotStarted) {
mMuxer.writeSampleData(mTrackIndex, outputBuffer, mBufferInfo);
}
////////////////////////////END DEBUG//////////////////////////////////
//release the buffer
mCodec.releaseOutputBuffer(outputBufferIndex, false);
muxerPasses++;
}
If you've made it this far you're a gentleman (or lady!) and a scholar! Basically I'm stumped as to why my image is not coming out properly. I'm relatively new to video processing so I'm sure I'm just missing something.

If you're API 19+, might as well stick with encoding method #2, getImage()/encodeVideoFromImage(), since that is more modern.
Focusing on that method: One problem was, you had an unexpected image format. With COLOR_FormatYUV420Flexible, you know you're going to have 8-bit U and V components, but you won't know in advance where they go. That's why you have to query the Image.Plane formats. Could be different on every device.
In this case, the UV format turned out to be interleaved (very common on Android devices). If you're using Java, and you supply each array (U/V) separately, with the "stride" requested ("spacer" byte in-between each sample), I believe one array ends up clobbering the other, because these are actually "direct" ByteBuffers, and they were intended to be used from native code, like in this answer. The solution I explained was to copy an interleaved array into the third (V) plane, and ignore the U plane. On the native side, these two planes actually overlap each other in memory (except for the first and last byte), so filling one causes the implementation to fill both.
If you use the second (U) plane instead, you'll find things work, but the colors look funny. That's also because of the overlapping arrangement of these two planes; what that does, effectively, is shift every array element by one byte (which puts U's where V's should be, and vice versa.)
...In other words, this solution is actually a bit of a hack. Probably the only way to do this correctly, and have it work on all devices, is to use native code (as in the answer I linked above).
Once the color plane problem is fixed, that leaves all the funny overlapping text and vertical striations. These were actually caused by your interpretation of the RGB data, which had the wrong stride.
And, once that is fixed, you have a decent-looking picture. It's been mirrored vertically; I don't know the root cause of that, but I suspect it's an OpenGL issue.

measure running time of a method in Android

i have a method that generates an audio signal from a smartphone's speaker, i set the duration time for 1 second. i want to know if the played sound is really played for one second or not.
for that i created a playSound() method and used getTimeInMillis() method to measure the time.
the result was varying from 11 ms to 679 ms. it seemed to me that it just measures the excution time of the method not the whole running time (the time till the playSound() stops).
is there a way to measure that time in java?
void genTone() throws IOException{
double instfreq=0, numerator;
for (int i=0;i<numSample; i++ )
{
numerator=(double)(i)/(double)numSample;
instfreq =freq1+(numerator*(freq2-freq1));
if ((i % 1000) == 0) {
Log.e("Current Freq:", String.format("Freq is: %f at loop %d of %d", instfreq, i, numSample));
}
sample[i]=Math.sin(2*Math.PI*i/(sampleRate/instfreq));
}
int idx = 0;
for (final double dVal : sample) {
// scale to maximum amplitude
final short val = (short) ((dVal * 32767)); // max positive sample for signed 16 bit integers is 32767
// in 16 bit wave PCM, first byte is the low order byte (pcm: pulse control modulation)
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
DataOutputStream dd=new DataOutputStream( new FileOutputStream(Environment.getExternalStorageDirectory().getAbsolutePath()+"/tessst.wav" ));
dd.writeBytes("RIFF");
dd.writeInt(48000); // Final file size not known yet, write 0
dd.writeBytes("WAVE");
dd.writeBytes("fmt ");
dd.writeInt(Integer.reverseBytes(16)); // Sub-chunk size, 16 for PCM
dd.writeShort(Short.reverseBytes((short) 1)); // AudioFormat, 1 for PCM
dd.writeShort(Short.reverseBytes( (short) myChannels));// Number of channels, 1 for mono, 2 for stereo
dd.writeInt(Integer.reverseBytes(sampleRate)); // Sample rate
dd.writeInt(Integer.reverseBytes( (sampleRate*myBitsPerSample*myChannels/8))); // Byte rate, SampleRate*NumberOfChannels*BitsPerSample/8
dd.writeShort(Short.reverseBytes((short) (myChannels*myBitsPerSample/8))); // Block align, NumberOfChannels*BitsPerSample/8
dd.writeShort(Short.reverseBytes( myBitsPerSample)); // Bits per sample
dd.writeBytes("data");
dd.write(generatedSnd,0,numSample);
dd.close();
void playSound(){
AudioTrack audioTrack= null;
try{
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,sampleRate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, generatedSnd.length, AudioTrack.MODE_STATIC);
audioTrack.write(generatedSnd, 0, generatedSnd.length);
audioTrack.play();
}
catch(Exception e)
{
System.out.print(e);
}
}
void excute() throws IOException
{
long time=Calendar.getInstance().getTimeInMillis();
playSound();
time=Calendar.getInstance().getTimeInMillis()-time;
Log.e("time", String.format(" "+time+" ms"));
}

Android - Mixing multiple static waveforms into a single AudioTrack

I am making a class that takes an array of frequencies values (i.e. 440Hz, 880Hz, 1760Hz) and plays how they would sound combined into a single AudioTrack. I am not a sound programmer, so this is difficult for me to write myself, where I believe that it is a relatively easy problem to an experienced sound programmer. Here is some of the code below in the play method:
public void play() {
// Get array of frequencies with their relative strengths
double[][] soundData = getData();
// TODO
// Perform a calculation to fill an array with the mixed sound - then play it in an infinite loop
// Need an AudioTrack that will play calculated loop
// Track sample info
int numOfSamples = DURATION * SAMPLE_RATE;
double sample[] = new double[numOfSamples];
byte sound[] = new byte[2 * numOfSamples];
// fill out the array
for (int i = 0; i < numOfSamples; ++i) {
sample[i] = Math.sin(2 * Math.PI * i / (SAMPLE_RATE / 440));
}
int i = 0;
for (double dVal : sample) {
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
sound[i++] = (byte) (val & 0x00ff);
sound[i++] = (byte) ((val & 0xff00) >>> 8);
}
// Obtain a minimum buffer size
int minBuffer = AudioTrack.getMinBufferSize(SAMPLE_RATE, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
if (minBuffer > 0) {
// Create an AudioTrack
AudioTrack track = new AudioTrack(AudioManager.STREAM_MUSIC, SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, numOfSamples, AudioTrack.MODE_STATIC);
// Write audio data to track
track.write(sound, 0, sound.length);
// Begin playing track
track.play();
}
// Once everything has successfully begun, indicate such.
isPlaying = true;
}
Right now, this code simply plays a concert A (440Hz). It was to test whether this code works. Now, I need to take a bunch a frequencies, perform some kind of calculation, and write the sample data.

Ok, so the answer did turn out to be a simple summation loop. Here it is, just replace this for loop with the original one:
// fill out the array
for (int i = 0; i < numOfSamples; ++i) {
double valueSum = 0;
for (int j = 0; j < soundData.length; j++) {
valueSum += Math.sin(2 * Math.PI * i / (SAMPLE_RATE / soundData[j][0]));
}
sample[i] = valueSum / soundData.length;
}
Now, what this does is simply take all possible frequencies, add them together into the variable, valueSum, and then divide that by the length of the frequency array, soundData, which is a simple average. This produces a nice sine wave mixture of an arbitrarily long array of frequencies.
I haven't tested performance, but I do have this running in a thread, otherwise it could crash the UI. So, hope this helps - I am marking this as the answer.

If you intend to mix multiple waveforms into one, you might prevent clipping in several ways.
Assuming sample[i] is a float representing the sum of all sounds.
HARD CLIPPING:
if (sample[i]> 1.0f)
{
sample[i]= 1.0f;
}
if (sample[i]< -1.0f)
{
sample[i]= -1.0f;
}
HEADROOM (y= 1.1x - 0.2x^3 for the curve, min and max cap slighty under 1.0f)
if (sample[i] <= -1.25f)
{
sample[i] = -0.987654f;
}
else if (sample[i] >= 1.25f)
{
sample[i] = 0.987654f;
}
else
{
sample[i] = 1.1f * sample[i] - 0.2f * sample[i] * sample[i] * sample[i];
}
For a 3rd polynomial waveshapper (less smooth), replace the last line above with:
sample[i]= 1.1f * sample[i]- 0.2f * sample[i] * sample[i] * sample[i];

AudioTrack: start called from a thread

I using AudioTrack in static mode to reproduce the same signal over and over again.
I have followed the example in here and sometimes it works perfectly, but sometimes it throws this error and it produces no sound:
AudioTrack: start called from a thread
01-23 15:26:16.902: W/libutils.threads(1133): Thread (this=0x3973b8): don't call waitForExit() from this Thread object's thread. It's a guaranteed deadlock!
This is the source code. I'm trying to ensure that I call stop and reload the data for the next "play" execution.
public class SoundPlayer {
// originally from http://marblemice.blogspot.com/2010/04/generate-and-play-tone-in-android.html
private int numSamples;
private double sample[];
private byte generatedSnd[];
private AudioTrack audioTrack;
public SoundPlayer(float duration, int sampleRate, double freqOfTone) {
super();
this.numSamples = (int) (duration * sampleRate);
this.sample = new double[numSamples];
this.generatedSnd = new byte[2 * numSamples];
// fill out the array
for (int i = 0; i < numSamples; ++i) {
sample[i] = Math.sin(2 * Math.PI * i / (sampleRate / freqOfTone));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
for (final double dVal : sample) {
// scale to maximum amplitude
final short val = (short) ((dVal * 32767));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, numSamples,
AudioTrack.MODE_STATIC);
audioTrack.write(generatedSnd, 0, generatedSnd.length);
}
public void playSound() {
if ( audioTrack.getPlayState() == (AudioTrack.PLAYSTATE_PLAYING | AudioTrack.PLAYSTATE_PAUSED )) {
audioTrack.stop();
audioTrack.reloadStaticData();
}
Log.i("Audio", "playState: " + audioTrack.getPlayState());
audioTrack.play();
audioTrack.stop();
audioTrack.reloadStaticData();
}
}
If we open the android source code, it does not explains a lot:
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
LOGV("start");
if (t != 0) {
if (t->exitPending()) {
if (t->requestExitAndWait() == WOULD_BLOCK) {
LOGE("AudioTrack::start called from thread");
return;
}
}
t->mLock.lock();
}
Does anyone know how to handle this?

I had a similar problem once.
Simply put, if you happen to have stuff running in more than one thread you have to make sure that creating the AudioTrack and calling play() and stop() on it must happen in the same thread.
However, it doesn't mean that you have to create the audio samples in that thread, too. In case you use static audio data (AudioTrack.MODE_STATIC) you can preload or generate them elsewhere and then use it even multiple times throughout the lifetime of your app.