Below is the code for my play() method which simply generates an arbitrary set of frequencies and blends them into one tone.
The problem is that it only plays for a split second - I need is to play it continuously. I would appreciate suggestions on how to constantly generate the sound using the AudioTrack class in Android. I believe it has something to do with the MODE_STREAM constant, but I can't quite work out how.
Here is the link to AudioTrack class documentation:
http://developer.android.com/reference/android/media/AudioTrack.html
EDIT: I forgot to mention one important aspect, it can't loop. Due to the mixing of sometimes up to 50+ frequencies, it will sound choppy because there is no least common denominator for all frequency peaks - or it's too far down the waveform to store as one sound.
/**
* play - begins playing the sound
*/
public void play() {
// Get array of frequencies with their relative strengths
double[][] soundData = getData();
// Track samples array
final double samples[] = new double[1024];
// Calculate the average sum in the array and write it to sample
for (int i = 0; i < samples.length; ++i) {
double valueSum = 0;
for (int j = 0; j < soundData.length; j++) {
valueSum += Math.sin(2 * Math.PI * i / (SAMPLE_RATE / soundData[j][0]));
}
samples[i] = valueSum / soundData.length;
}
// 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
mTrack = new AudioTrack(AudioManager.STREAM_MUSIC, SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, minBuffer, AudioTrack.MODE_STREAM);
// Begin playing track
mTrack.play();
// Fill the buffer
if (mBuffer.length < samples.length) {
mBuffer = new short[samples.length];
}
for (int k = 0; k < samples.length; k++) {
mBuffer[k] = (short) (samples[k] * Short.MAX_VALUE);
}
// Write audio data to track for real-time audio sythesis
mTrack.write(mBuffer, 0, samples.length);
}
// Once everything has successfully begun, indicate such.
isPlaying = true;
}
It looks like the code is almost there. It just needs a loop to keep generating the samples, putting them in the buffer, and writing them to the AudioTrack. Right now just one buffer full gets written before it exits which is why it stops so quickly.
void getSamples(double[] samples) {
// Get array of frequencies with their relative strengths
double[][] soundData = getData();
// Calculate the average sum in the array and write it to sample
for (int i = 0; i < samples.length; ++i) {
double valueSum = 0;
for (int j = 0; j < soundData.length; j++) {
valueSum += Math.sin(2 * Math.PI * i / (SAMPLE_RATE / soundData[j][0]));
}
samples[i] = valueSum / soundData.length;
}
}
public void endPlay() {
done = true;
}
/**
* play - begins playing the sound
*/
public void play() {
// 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
mTrack = new AudioTrack(AudioManager.STREAM_MUSIC, SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, minBuffer, AudioTrack.MODE_STREAM);
// Begin playing track
mTrack.play();
// Track samples array
final double samples[] = new double[1024];
while (!done) {
// Fill the buffer
if (mBuffer.length < samples.length) {
mBuffer = new short[samples.length];
}
getSamples(samples);
for (int k = 0; k < samples.length; k++) {
mBuffer[k] = (short) (samples[k] * Short.MAX_VALUE);
}
// Write audio data to track for real-time audio sythesis
mTrack.write(mBuffer, 0, samples.length);
// Once everything has successfully begun, indicate such.
isPlaying = true;
}
}
// Once everything is done, indicate such.
isPlaying = false;
}
Related
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.
i'm trying to draw audio amplitudes by time, i'm using to achieve this, the AudioRecord class, which gives me a raw audio array.
new Thread(new Runnable() {
#Override
public void run() {
while (mIsRecording) {
int readSize = mRecorder.read(mBuffer, 0, mBuffer.length);
for (int i = 0; i < readSize; i++) {
long time = mChronometer.getTimeElapsed();
ampArray.add((mBuffer[i]));
timeArray.add(time);
}
}
}
}).start();
}
The parameters i use for AudioRecored are:
public static final int SAMPLE_RATE = 8000;
private void initRecorder() {
int bufferSize = AudioRecord.getMinBufferSize(SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT);
mBuffer = new short[bufferSize];
mRecorder = new AudioRecord(MediaRecorder.AudioSource.MIC, SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT, bufferSize);
}
The result i get is this one:
The result i get -----
What i'm looking for
Am I missing something here?
Thanks in advance.
EDIT: Drawing method:
When the recording is stopped, i send all the values saved in the amplitude array and the one in the time array to the LineGraphSeries of the GraphView Api
series = new LineGraphSeries<DataPoint>(generateData(ampArray, timeArray));
graph.addSeries(series);
generateData method:
double x = 0; int i = 0; short y = 0;
private DataPoint[] generateData(ArrayList<Short> ampArray, ArrayList<Double> timeArray) {
DataPoint[] values = new DataPoint[ampArray.size()];
for (int i=0; i< ampArray.size(); i++) {
x = timeArray.get(i);
y = ampArray.get(i);
DataPoint v = new DataPoint(x, y);
values[i] = v;
}
return values;
}
I'm going to take an educated guess here and suggest that it has something to do with these two lines:
long time = mChronometer.getTimeElapsed();
timeArray.add(time);
It looks to me like you are trying to plot the samples which have occurred in a different time regime and you are processing in batch against the current CPU clock which would explain your results - for example you might process a big block of samples - which you can do much faster than they occurred in the first place - and they might all get a similar time axis value.
The proper approach is to reconstruct the time axis for the samples themselves. Assume the first sample you process is time 0. If your sample rate is 48000 then each sample is 1/48000 of a second. The approach would be something like this:
int sampleNumber = 0;
while (mIsRecording) {
int readSize = mRecorder.read(mBuffer, 0, mBuffer.length);
for (int i = 0; i < readSize; i++) {
ampArray.add((mBuffer[i]));
double time = sampleNumber / SAMPLE_RATE;
timeArray.add(time);
sampleNumber++;
}
}
Note, I changed timeArray from int to double as it is now in seconds rather than milliseconds. If you prefer milliseconds then multiply time by 1000 and cast to a long.
Also, you don't need to create an array for the time axis as you can determine the time of any sample based upon its absolute index in the ampArray.
I am making an app where I need to read continues stream of sound which is sent in the form of an byte array. The server side records sound like this (based on an example here on SO):
// Get the minimum buffer size required for the successful creation of an AudioRecord object.
int bufferSizeInBytes = AudioRecord.getMinBufferSize(RECORDER_SAMPLERATE, RECORDER_CHANNELS,
RECORDER_AUDIO_ENCODING);
bufferSizeInBytes = 30000;
// Initialize Audio Recorder.
_audio_recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, RECORDER_SAMPLERATE,
RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING, bufferSizeInBytes);
// Start Recording.
_audio_recorder.startRecording();
int numberOfReadBytes = 0;
byte audioBuffer[] = new byte[bufferSizeInBytes];
boolean recording = false;
float tempFloatBuffer[] = new float[3];
int tempIndex = 0;
byte totalByteBuffer[] = new byte[60 * 44100 * 2];
while (true)
{
float totalAbsValue = 0.0f;
short sample = 0;
numberOfReadBytes = _audio_recorder.read(audioBuffer, 0, bufferSizeInBytes);
for (int i = 0; i < bufferSizeInBytes; i += 2)
{
sample = (short) ((audioBuffer[i]) | audioBuffer[i + 1] << 8);
totalAbsValue += Math.abs(sample) / (numberOfReadBytes / 2);
}
tempFloatBuffer[tempIndex % 3] = totalAbsValue;
float temp = 0.0f;
for (int i = 0; i < 3; ++i)
temp += tempFloatBuffer[i];
if ((temp >= 0 && temp <= _sensitivity) && recording == false)
{
Log.i("TAG", "1");
tempIndex++;
continue;
}
if (temp > _sensitivity && recording == false)
{
Log.i("TAG", "2");
recording = true;
}
if(temp < _sensitivity && recording == true)
{
recording = false;
continue;
}
for (int i = 0; i < numberOfReadBytes; i++)
totalByteBuffer[i] = audioBuffer[i];
if (prepare_sound(totalByteBuffer, numberOfReadBytes))
{
totalByteBuffer = new byte[60 * 44100 * 2];
tempIndex++;
}
}
The example this is taken from is recording sound and saves it to a file when there is no more sound to record. My goal on the other hand is to record sound when there is sound and send this sound on the fly when still recording. Hence, I want to send sounds right a way and not store it to a file when there is no more sound to record. So far I am taking the byte[] with data and stores it in an object an sends it to a client using ObjectOutputStream. The client will then create a temp sound file and play it using MediaPlayer. But I feel that this is not the most effect way to achieve my goal. So, is there any more efficient way to do this with respect to send an continues stream of data as media player does not support playing pure byte[] of data?
Thanks for any help and tips!
Found out that the best solutions for me is to record the sound and when the buffer is full it is sent to the client side. The client then uses an AudioTrack instance to play the byte[] that contains the data like this:
public void onSoundReceived(byte[] sound)
{
_audio_input_stream.write(sound, 0, sound.length);
}
This also makes the sound more "non-lagging" as this is not a MediaPlayer instance which will stop the sound after each time the data was done playing.
I have a problem to generate smooth sinus wave.
I've done it few years ago on C++ and everything worked perfect. Now I am trying to do this using AudioTrack and I do not know what is wrong.
This is my test case:
I want to produce for five second a sinus wave which is smooth (no crack etc.). For one second I generate 44100 samples and divided it on couple of buffer with size 8192 (probably this is the reason of cracks, but how can I fix it, without giving bigger size of buffer).
Unfortunatelly using my code the sound is not smooth and instead of 5 second it takes about 1 second. I would be gratefull for any help.
Please let me now if this piece of code is not enough.
class Constants:
//<---
public final static int SAMPLING = 44100;
public final static int DEFAULT_GEN_DURATION = 1000;
public final static int DEFAULT_NUM_SAMPLES = DEFAULT_GEN_DURATION * SAMPLING / 1000; //44100 per second
public final static int DEFAULT_BUFFER_SIZE = 8192;
//--->
//preparing buffers to play;
Buffer buffer = new Buffer();
short[] buffer_values = new short[Constants.DEFAULT_BUFFER_SIZE];
float[] samples = new float[Constants.DEFAULT_BUFFER_SIZE];
float d = (float) (( Constants.FREQUENCIES[index] * 2 * Math.PI ) / Constants.SAMPLING);
int numSamples = Constants.DEFAULT_NUM_SAMPLES; //44100 per second - for test
float x = 0;
int index_in_buffer = 0;
for(int i = 0; i < numSamples; i++){
if(index_in_buffer >= Constants.DEFAULT_BUFFER_SIZE - 1){
buffer.setBufferShort(buffer_values);
buffer.setBufferSizeShort(index_in_buffer);
queue_with_data_AL.add(buffer); //add buffer to queue
buffer_values = new short[Constants.DEFAULT_BUFFER_SIZE];
samples = new float[Constants.DEFAULT_BUFFER_SIZE];
index_in_buffer = 0;
}
samples[index_in_buffer] = (float) Math.sin(x);
buffer_values[index_in_buffer] = (short) (samples[index_in_buffer] * Short.MAX_VALUE);
x += d;
index_in_buffer++;
}
buffer.setBufferShort(buffer_values);
buffer.setBufferSizeShort(index_in_buffer+1);
queue_with_data_AL.add(buffer);
index_in_buffer = 0;
}
//class AudioPlayer
public AudioPlayer(int sampleRate) { //44100
int minSize = AudioTrack.getMinBufferSize(sampleRate,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
audiotrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT,
minSize, AudioTrack.MODE_STREAM);
}
public void play(byte[] audioData, int sizeOfBuffer) {
audiotrack.write(audioData, 0, sizeOfBuffer);
}
public void start() {
if (state == Constants.STOP_STATE) {
state = Constants.START_STATE;
int startLength = 0;
while (state == Constants.START_STATE) {
Buffer buffer = getBufferFromQueueAL(); //getting buffer from prepared list
if (buffer != null) {
short[] data = buffer.getBufferShort();
int size_of_data = buffer.getBufferSizeShort();
if (data != null) {
int len = audiotrack.write(data, 0, size_of_data);
if (startLength == 0) {
audiotrack.play();
}
startLength += len;
} else {
break;
}
} else {
MessagesLog.e(TAG, "get null data");
break;
}
}
if (audiotrack != null) {
audiotrack.pause();
audiotrack.flush();
audiotrack.stop();
}
}
}
You are playing only 1 second because 44100 samples at a samplerate of 44100Hz result in exactly 1 second of sound.
You have to generate 5 times more samples if you want to play 5 seconds of sound (e.g. multiply DEFAULT_NUM_SAMPLES by 5) in your code.
I've found the solution by myself. After adding Buffer to queue_with_data_AL I've forgotten create new instance of Buffer object. So in queue was couple of buffer with the same instance, hence sinus wave were not continuous.
Thanks if someone was trying to solve my problem. Unfortunatelly it was my programming mistake.
Best regards.
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];