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How do I modulate a signal for radio transmission (SDR) in Java?

Off Topic: Let me start by saying Java is completely new to me. I've been programming for over 15 years and never have had a need for it beyond modifying others' codebases, so please forgive my ignorance and possibly improper terminology. I'm also not very familiar with RF, so if I'm way left field here, please let me know!
I'm building an SDR (Software Defined Radio) radio transmitter, and while I can successfully transmit on a frequency, when I send the stream (either from the device's microphone or bytes from a tone generator), what is coming through my handheld receiver sounds like static.
I believe this to be due to my receiver being set up to receive NFM (Narrowband Frequency Modulation) and WFM (Wideband Frequency Modulation) while the transmission coming from my SDR is sending raw, unmodulated data.
My question is: how do I modulate audio bytes (i.e. an InputStream) so that the resulting bytes are modulated in FM (Frequency Modulation) or AM (Amplitude Modulation), which I can then transmit through the SDR?
I can't seem to find a class or package that handles modulation (eventually I'm going to have to modulate WFM, FM, AM, SB, LSB, USB, DSB, etc.) despite there being quite a few open-source SDR codebases, but if you know where I can find this, that basically answers this question. Everything I've found so far has been for demodulation.
This is a class I've built around Xarph's Answer here on StackOverflow, it simply returns a byte array containing a simple, unmodulated audio signal, which can then be used to play sound through speakers (or transmit over an SDR, but due to the result not being properly modulated, it doesn't come through correctly on the receiver's end, which is what I'm having trouble figuring out)
public class ToneGenerator {
public static byte[] generateTone() {
return generateTone(60, 1000, 8000);
}
public static byte[] generateTone(double duration) {
return generateTone(duration, 1000, 8000);
}
public static byte[] generateTone(double duration, double freqOfTone) {
return generateTone(duration, freqOfTone, 8000);
}
public static byte[] generateTone(double duration, double freqOfTone, int sampleRate) {
double dnumSamples = duration * sampleRate;
dnumSamples = Math.ceil(dnumSamples);
int numSamples = (int) dnumSamples;
double sample[] = new double[numSamples];
byte generatedSnd[] = new byte[2 * numSamples];
for (int i = 0; i < numSamples; ++i) { // Fill the sample array
sample[i] = Math.sin(freqOfTone * 2 * Math.PI * i / (sampleRate));
}
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalized.
// convert to 16 bit pcm sound array
// assumes the sample buffer is normalised.
int idx = 0;
int i = 0 ;
int ramp = numSamples / 20 ; // Amplitude ramp as a percent of sample count
for (i = 0; i< ramp; ++i) { // Ramp amplitude up (to avoid clicks)
double dVal = sample[i];
// Ramp up to maximum
final short val = (short) ((dVal * 32767 * i/ramp));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
for (i = i; i< numSamples - ramp; ++i) { // Max amplitude for most of the samples
double dVal = sample[i];
// 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);
}
for (i = i; i< numSamples; ++i) { // Ramp amplitude down
double dVal = sample[i];
// Ramp down to zero
final short val = (short) ((dVal * 32767 * (numSamples-i)/ramp ));
// in 16 bit wav PCM, first byte is the low order byte
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
}
return generatedSnd;
}
}
An answer to this doesn't necessarily need to be code, actually theory and an understanding of how FM or AM modulation works when it comes to processing a byte array and converting it to the proper format would probably be more valuable since I'll have to implement more modes in the future.

There is a lot that I don't know about radio. But I think I can say a couple things about the basics of modulation and the problem at hand given the modicum of physics that I have and the experience of coding an FM synthesizer.
First off, I think you might find it easier to work with the source signal's PCM data points if you convert them to normalized floats (ranging from -1f to 1f), rather than working with shorts.
The target frequency of the receiver, 510-1700 kHz (AM radio) is significantly faster than the sample rate of the source sound (presumably 44.1kHz). Assuming you have a way to output the resulting data, the math would involve taking a PCM value from your signal, scaling it appropriately (IDK how much) and multiplying the value against the PCM data points generated by your carrier signal that corresponds to the time interval.
For example, if the carrier signal were 882 kHz, you would multiply a sequence of 20 carrier signal values with the source signal value before moving on to the next source signal value. Again, my ignorance: the tech may have some sort of smoothing algorithm for the transition between the source signal data points. I really don't know about that or not, or at what stage it occurs.
For FM, we have carrier signals in the MHz range, so we are talking orders of magnitude more data being generated per each source signal value than with AM. I don't know the exact algorithm used but here is a simple conceptual way to implement frequency modulation of a sine that I used with my FM synthesizer.
Let's say you have a table with 1000 data points that represents a single sine wave that ranges between -1f to 1f. Let's say you have a cursor that repeatedly traverses the table. If the cursor advanced exactly 1 data point at 44100 fps and delivered the values at that rate, the resulting tone would be 44.1 Hz, yes? But you can also traverse the table via intervals larger than 1, for example 1.5. When the cursor lands in between two table values, one can use linear interpolation to determine the value to output. The cursor increment of 1.5 would result in the sine wave being pitched at 66.2 Hz.
What I think is happening with FM is that this cursor increment is continuously varied, and the amount it is varied depends on some sort of scaling from the source signal translated into a range of increments.
The specifics of the scaling are unknown to me. But suppose a signal is being transmitted with a carrier of 10MHz and ranges ~1% (roughly from 9.9 MHz to 10.1 MHz), the normalized source signal would have some sort of algorithm where a PCM value of -1 match an increment that traverses the carrier wave causing it to produce the slower frequency and +1 match an increment that traverses the carrier wave causing it to produce the higher frequency. So, if an increment of +1 delivers 10 MHz, maybe a source wave PCM signal of -1 elicits a cursor increment of +0.99, a PCM value of -0.5 elicits an increment of +0.995, a value of +0.5 elicits an increment of +1.005, a value of +1 elicits a cursor increment of 1.01.
This is pure speculation on my part as to the relationship between the source PCM values and how that are used to modulate the carrier frequency. But maybe it helps give a concrete image of the basic mechanism?
(I use something similar, employing a cursor to iterate over wav PCM data points at arbitrary increments, in AudioCue (a class for playing back audio data based on the Java Clip), for real time frequency shifting. Code line 1183 holds the cursor that iterates over the PCM data that was imported from the wav file, with the variable idx holding the cursor increment amount. Line 1317 is where we fetch the audio value after incrementing the cursor. Code lines 1372 has the method readFractionalFrame() which performs the linear interpolation. Real time volume changes are also implemented, and I use smoothing on the values that are provided from the public input hooks.)
Again, IDK if any sort of smoothing is used between source signal values or not. In my experience a lot of the tech involves filtering and other tricks of various sorts that improve fidelity or processing calculations.

How can sound be generated from known points?

I have 10 lists of points (each point is time-amplitude pair), where each list belongs to known frequency
So i have a class InputValue with two fields sampleDate (long) and sampleValue (double), and 10 lists - List samples800Hz, samples400Hz and so on.
800Hz list contains about 1600 points (not fixed value because data sampler can have un-predictable delays) for each second, 400Hz list contains about 800 points for each second and so on.
How can i:
Generate sound from list of points
Mix several or all lists in one sound?
If i got it right, i need to resample each list to one sample rate (can java soundformat take custom sample rates like 1600, or i should use standart ones, where lowest is 8000?) and then fill samples buffer like
AudioFormat af = new AudioFormat( (float )1600, 8, 1, true, false );
SourceDataLine sdl = AudioSystem.getSourceDataLine( af );
sdl.open();
sdl.start();
for( int i = 0; i < 1600; i++ ) {
buf[ 0 ] = ???
sdl.write( buf, 0, 1 );
}
sdl.drain();
sdl.stop();
But how can i tell sdl that my aplitude value belongs to some frequency? and how can i mix different frequencies?
BTW, can i, instead of resampling each list, create 10 audioformats with different sample rates (1600 for 800Hz, 800 for 400Hz and so on) and later mix 10 sdls in one?

It sounds like you're trying to use a wavetable for your sound generation. If you're simply just recreating an 800 Hz tone, this is easy:
static int sample = 0;
for (int i = 0; i < 1600; i++) {
buf[i] = samples800Hz[sample];
sample = (sample + 1) % SAMPLES_800HZ_SIZE;
}
Lets say you want to combine an 800 Hz and 1600 Hz tone... just add it together (you might have to mix the values so they don't clip):
static int sample1 = 0, sample2 = 0;
for (int i = 0; i < 1600; i++) {
// Multiply each sample by 0.5; this gives us a 50% mix between the two
buf[i] = (samples800Hz[sample1] * 0.5) + (samples1600Hz[sample2] * 0.5);
sample1 = (sample1 + 1) % SAMPLES_800HZ_SIZE;
sample2 = (sample2 + 1) % SAMPLES_1600HZ_SIZE;
}
Now my answer doesn't consider how many times/number of frames your system is running its callback. You'll have to figure that out on your own. Also, if you wanted to have multiple tone generation instead of endlessly making lists, I would urge you to look up wavetable oscillators. A wavetable is basically creating one array of a tone and then adjusting the speed/phase you read the table to generate a desired frequency.
Good luck!

Fourier transforming a byte array

I am not so proficient in Java, so please keep it quite simple. I will, though, try to understand everything you post. Here's my problem.
I have written code to record audio from an external microphone and store that in a .wav. Storing this file is relevant for archiving purposes. What I need to do is a FFT of the stored audio.
My approach to this was loading the wav file as a byte array and transforming that, with the problem that 1. There's a header in the way I need to get rid of, but I should be able to do that and 2. I got a byte array, but most if not all FFT algorithms I found online and tried to patch into my project work with complex / two double arrays.
I tried to work around both these problems and finally was able to plot my FFT array as a graph, when I found out it was just giving me back "0"s. The .wav file is fine though, I can play it back without problems. I thought maybe converting the bytes into doubles was the problem for me, so here's my approach to that (I know it's not pretty)
byte ByteArray[] = Files.readAllBytes(wav_path);
String s = new String(ByteArray);
double[] DoubleArray = toDouble(ByteArray);
// build 2^n array, fill up with zeroes
boolean exp = false;
int i = 0;
int pow = 0;
while (!exp) {
pow = (int) Math.pow(2, i);
if (pow > ByteArray.length) {
exp = true;
} else {
i++;
}
}
System.out.println(pow);
double[] Filledup = new double[pow];
for (int j = 0; j < DoubleArray.length; j++) {
Filledup[j] = DoubleArray[j];
System.out.println(DoubleArray[j]);
}
for (int k = DoubleArray.length; k < Filledup.length; k++) {
Filledup[k] = 0;
}
This is the function I'm using to convert the byte array into a double array:
public static double[] toDouble(byte[] byteArray) {
ByteBuffer byteBuffer = ByteBuffer.wrap(byteArray);
double[] doubles = new double[byteArray.length / 8];
for (int i = 0; i < doubles.length; i++) {
doubles[i] = byteBuffer.getDouble(i * 8);
}
return doubles;
}
The header still is in there, I know that, but that should be the smallest problem right now. I transformed my byte array to a double array, then filled up that array to the next power of 2 with zeroes, so that the FFT can actually work (it needs an array of 2^n values). The FFT algorithm I'm using gets two double arrays as input, one being the real, the other being the imaginary part. I read, that for this to work, I'd have to keep the imaginary array empty (but its length being the same as the real array).
Worth to mention: I'm recording with 44100 kHz, 16 bit and mono.
If necessary, I'll post the FFT I'm using.
If I try to print the values of the double array, I get kind of weird results:
...
-2.0311904060823147E236
-1.3309975624948503E241
1.630738286366793E-260
1.0682002560745842E-255
-5.961832069690704E197
-1.1476447092561027E164
-1.1008407401197794E217
-8.109566204271759E298
-1.6104556241572942E265
-2.2081172620352248E130
NaN
3.643749694745671E-217
-3.9085815506127892E202
-4.0747557114875874E149
...
I know that somewhere the problem lies with me overlooking something very simple I should be aware of, but I can't seem to find the problem. My question finally is: How can I get this to work?

There's a header in the way I need to get rid of […]
You need to use javax.sound.sampled.AudioInputStream to read the file if you want to "skip" the header. This is useful to learn anyway, because you would need the data in the header to interpret the bytes if you did not know the exact format ahead of time.
I'm recording with 44100 kHz, 16 bit and mono.
So, this almost certainly means the data in the file is encoded as 16-bit integers (short in Java nomenclature).
Right now, your ByteBuffer code makes the assumption that it's already 64-bit floating point and that's why you get strange results. In other words, you are reinterpreting the binary short data as if it were double.
What you need to do is read in the short data and then convert it to double.
For example, here's a rudimentary routine to do such as you're trying to do (supporting 8-, 16-, 32- and 64-bit signed integer PCM):
import javax.sound.sampled.*;
import javax.sound.sampled.AudioFormat.Encoding;
import java.io.*;
import java.nio.*;
static double[] readFully(File file)
throws UnsupportedAudioFileException, IOException {
AudioInputStream in = AudioSystem.getAudioInputStream(file);
AudioFormat fmt = in.getFormat();
byte[] bytes;
try {
if(fmt.getEncoding() != Encoding.PCM_SIGNED) {
throw new UnsupportedAudioFileException();
}
// read the data fully
bytes = new byte[in.available()];
in.read(bytes);
} finally {
in.close();
}
int bits = fmt.getSampleSizeInBits();
double max = Math.pow(2, bits - 1);
ByteBuffer bb = ByteBuffer.wrap(bytes);
bb.order(fmt.isBigEndian() ?
ByteOrder.BIG_ENDIAN : ByteOrder.LITTLE_ENDIAN);
double[] samples = new double[bytes.length * 8 / bits];
// convert sample-by-sample to a scale of
// -1.0 <= samples[i] < 1.0
for(int i = 0; i < samples.length; ++i) {
switch(bits) {
case 8: samples[i] = ( bb.get() / max );
break;
case 16: samples[i] = ( bb.getShort() / max );
break;
case 32: samples[i] = ( bb.getInt() / max );
break;
case 64: samples[i] = ( bb.getLong() / max );
break;
default: throw new UnsupportedAudioFileException();
}
}
return samples;
}
The FFT algorithm I'm using gets two double arrays as input, one being the real, the other being the imaginary part. I read, that for this to work, I'd have to keep the imaginary array empty (but its length being the same as the real array).
That's right. The real part is the audio sample array from the file, the imaginary part is an array of equal length, filled with 0's e.g.:
double[] realPart = mySamples;
double[] imagPart = new double[realPart.length];
myFft(realPart, imagPart);
More info... "How do I use audio sample data from Java Sound?"

The samples in a wave file are not going to be already 8-byte doubles that can be directly copied as per your posted code.
You need to look up (partially from the WAVE header format and from the RIFF specification) the data type, format, length and endianess of the samples before converting them to doubles.
Try 2 byte little-endian signed integers as a likely possibility.

Function to generate tone skipping cretain frequencies

The following was taken from an Android app:
public void genTone(int freq){
for(int i = 0; i<numSamples; i++){
samples[i] = Math.pow(-1, (float)(i / (sampleRate/freq)));
}
int idx = 0;
int volume = 32767 * cx/wide;
for (final double dVal : samples) {
final short val = (short) ((dVal+1) * volume);
generatedSnd[idx++] = (byte) (val & 0x00ff);
generatedSnd[idx++] = (byte) ((val & 0xff00) >>> 8);
if(isRecording){
toRec.add((byte)(val & 0x00ff));
toRec.add((byte)((val & 0xff00) >>> 8));
}
}
}
The above code is a Java function in an Android app the generate a square wave with a specified frequency. The frequency is determined by an integer 'note' which is the last recorded position of a MotionEvent divided by the height of the screen. The frequency is 440 * 2^(note/12). I've made the program output in text the note and frequency, and it outputs what I want it to, but at certain notes, even though it outputs a different frequency in text, it sounds exactly the same. Is 8000 too low a sampleRate(per second)? Is this a well-known bug? Anything you can help me with?

The primary issue is that you need to leave the period as a float, because 8000/789 and 8000/739 otherwise round to the same integer:
float period = (float)sampleRate / (float)freq;
for(int i = 0; i<numSamples; i++){
samples[i] = ((int)(i/period) % 2)==0 ? 0 : 1;
}
As a secondary stylistic issue, I would also suggest removing the "+1" from the following line and instead requiring that cx/wide (which you should really call "gain") be exclusively in the range [0,1]:
final short val = (short) ((dVal) * volume);
Your original code was effectively generating samples as (0, +2, 0, +2,...) because of the +1, and then multiplying by cx/wide, but this would cause an overflow in your short val if cx/wide were ever greater than 1/2 (because 2*32767*1/2 is the maximum value of a short).
A more common practice would be to specify cx/wide as a gain in the range of (0,1) (where 0=min volume and 1=max volume) and also to enforce that constraint in the function.
A tertiary issue is that you are probably only using half of the dynamic range of the output. If the output is supposed to be 16-bit signed samples (look it up), then you can generate the original samples as ? -1 : 1 instead of 0/1. If the output stream accepts 16-bit unsigned samples, then you'd want to leave the samples as 0/1, but change volume to 65535, and use an int instead of a short for val.

AudioTrack - short array to byte array distortion using jlayer(java mp3 decoder)

I'm using jLayer to decode MP3 data, with this call:
SampleBuffer output = (SampleBuffer) decoder.decodeFrame(frameHeader, bitstream);
This call which returns the decoded data, returns an array of short[].
output.getBuffer();
When I call AudioTrack write() with that method, it plays fine as I loop through the file:
at.write(output.getBuffer(), 0, output.getBuffer().length);
However, when I convert the short[] array to byte[] array using any of the methods in this answer: https://stackoverflow.com/a/12347176/1176436 the sound gets distorted and jittery:
at.write(output.getBuffer(), 0, output.getBuffer().length);
becomes:
byte[] array = ShortToByte_Twiddle_Method(output.getBuffer());
at.write(array, 0, array.length);
Am I doing anything wrong and what can I do to fix it? Unfortunately I need the pcm data to be in a byte array for another 3rd party library I'm using. The file is 22kHz if that matters and this is how at is being instantiated:
at = new AudioTrack(AudioManager.STREAM_MUSIC, 22050, AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_16BIT, 10000 /* 10 second buffer */,
AudioTrack.MODE_STREAM);
Thank you so much in advance.
Edit: This is how I'm instantiating the AudioTrack variable now. So for 44kHz files, the value that is getting sent is 44100, while for 22kHz files, the value is 22050.
at = new AudioTrack(AudioManager.STREAM_MUSIC, decoder.getOutputFrequency(),
decoder.getOutputChannels() > 1 ? AudioFormat.CHANNEL_OUT_STEREO : AudioFormat.CHANNEL_OUT_MONO,
AudioFormat.ENCODING_PCM_16BIT, 10000 /* 10 second buffer */,
AudioTrack.MODE_STREAM);
This is decode method:
public byte[] decode(InputStream inputStream, int startMs, int maxMs) throws IOException {
ByteArrayOutputStream outStream = new ByteArrayOutputStream(1024);
float totalMs = 0;
boolean seeking = true;
try {
Bitstream bitstream = new Bitstream(inputStream);
Decoder decoder = new Decoder();
boolean done = false;
while (!done) {
Header frameHeader = bitstream.readFrame();
if (frameHeader == null) {
done = true;
} else {
totalMs += frameHeader.ms_per_frame();
if (totalMs >= startMs) {
seeking = false;
}
if (!seeking) {
// logger.debug("Handling header: " + frameHeader.layer_string());
SampleBuffer output = (SampleBuffer) decoder.decodeFrame(frameHeader, bitstream);
short[] pcm = output.getBuffer();
for (short s : pcm) {
outStream.write(s & 0xff);
outStream.write((s >> 8) & 0xff);
}
}
if (totalMs >= (startMs + maxMs)) {
done = true;
}
}
bitstream.closeFrame();
}
return outStream.toByteArray();
} catch (BitstreamException e) {
throw new IOException("Bitstream error: " + e);
} catch (DecoderException e) {
throw new IOException("Decoder error: " + e);
}
}
This is how it sounds (wait a few seconds): https://vimeo.com/60951237 (and this is the actual file: http://www.tonycuffe.com/mp3/tail%20toddle.mp3)
Edit: I would have loved to have split the bounty, but instead I have given the bounty to Bill and the accepted answer to Neil. Both were a tremendous help. For those wondering, I ended up rewriting the Sonic native code which helped me move along the process.

As #Bill Pringlemeir says, the problem is that your conversion method doesn't actually convert. A short is a 16 bit number; a byte is an 8 bit number. The method you have chosen doesn't convert the contents of the shorts (ie go from 16 bits to 8 bits for the contents), it changes the way in which the same collection of bits is stored. As you say, you need something like this:
SampleBuffer output = (SampleBuffer) decoder.decodeFrame(frameHeader, bitstream);
byte[] array = MyShortToByte(output.getBuffer());
at.write(array, 0, array.length);
#Bill Pringlemeir's approach is equivalent to dividing all the shorts by 256 to ensure they fit in the byte range:
byte[] MyShortToByte(short[] buffer) {
int N = buffer.length;
ByteBuffer byteBuf = ByteBuffer.allocate(N);
while (N >= i) {
byte b = (byte)(buffer[i]/256); /*convert to byte. */
byteBuf.put(b);
i++;
}
return byteBuf.array();
}
This will work, but will probably give you very quiet, edgy tones. If you can afford the processing time, a two pass approach will probably give better results:
byte[] MyShortToByte(short[] buffer) {
int N = buffer.length;
short min = 0;
short max = 0;
for (int i=0; i<N; i++) {
if (buffer[i] > max) max = buffer[i];
if (buffer[i] < min) min = buffer[i];
}
short scaling = 1+(max-min)/256; // 1+ ensures we stay within range and guarantee no divide by zero if sequence is pure silence ...
ByteBuffer byteBuf = ByteBuffer.allocate(N);
for (int i=0; i<N; i++) {
byte b = (byte)(buffer[i]/scaling); /*convert to byte. */
byteBuf.put(b);
}
return byteBuf.array();
}
Again, beware signed / unsigned issue. The above works signed-> signed and unsigned->unsigned; but not between the two. It may be that you are reading signed shorts (-32768-32767), but need to output unsigned bytes (0-255), ...
If you can afford the processing time, a more precise (smoother) approach would be to go via floats (this also gets round the signed/unsigned issue):
byte[] MyShortToByte(short[] buffer) {
int N = buffer.length;
float f[] = new float[N];
float min = 0.0f;
float max = 0.0f;
for (int i=0; i<N; i++) {
f[i] = (float)(buffer[i]);
if (f[i] > max) max = f[i];
if (f[i] < min) min = f[i];
}
float scaling = 1.0f+(max-min)/256.0f; // +1 ensures we stay within range and guarantee no divide by zero if sequence is pure silence ...
ByteBuffer byteBuf = ByteBuffer.allocate(N);
for (int i=0; i<N; i++) {
byte b = (byte)(f[i]/scaling); /*convert to byte. */
byteBuf.put(b);
}
return byteBuf.array();
}

The issue is with your short to byte conversion. The byte conversion link preserves all information including the high and low byte portions. When you are converting from 16bit to 8bit PCM samples, you must discard the lower byte. My Java skills are weak, so the following may not work verbatim. See also: short to byte conversion.
ByteBuffer byteBuf = ByteBuffer.allocate(N);
while (N >= i) {
/* byte b = (byte)((buffer[i]>>8)&0xff); convert to byte. native endian */
byte b = (byte)(buffer[i]&0xff); /*convert to byte; swapped endian. */
byteBuf.put(b);
i++;
}
That is the following conversion,
AAAA AAAA SBBB BBBB -> AAAA AAAA, +1 if S==1 and positive else -1 if S==1
A is a bit that is kept. B is a discarded bit and S is a bit that you may wish to use for rounding. The rounding is not needed, but it may sound a little better. Basically, 16 bit PCM is higher resolution than 8 bit PCM. You lose those bits when the conversion is done. The short to byte routine tries to preserve all information.
Of course, you must tell the sound library that you are using 8-bit PCM. My guess,
at = new AudioTrack(AudioManager.STREAM_MUSIC, 22050, AudioFormat.CHANNEL_OUT_STEREO,
AudioFormat.ENCODING_PCM_8BIT, 10000 /* 10 second buffer */,
AudioTrack.MODE_STREAM);
If you can only use 16bit PCM to play audio, then you have to do the inverse and convert the 8bit PCM from the library to 16bit PCM for playback. Also note, that typically, 8bit samples are often NOT straight PCM but u-law or a-law encoded. If the 3rd party library uses these formats, the conversion is different but you should be able to code it from the wikipedia links.
NOTE: I have not included the rounding code as overflow and sign handling will complicate the answer. You must check for overflow (Ie, 0x8f + 1 gives 0xff or 255 + 1 giving -1). However, I suspect the library is not straight 8bit PCM.
See Also: Alsa PCM overview, Multi-media wiki entry on PCM - Ultimately Android uses ALSA for sound.
Other factors that must be correct for a PCM raw buffer are sample rate, number of channels (stereo/mono), PCM format including bits, companding, little/big endian and sample interleaving.
EDIT: After some investigation, the JLayer decoder typically returns big endian 16bit values. The Sonic filter, takes a byte but threats them as 16bit little endian underneath. Finally, the AudioTrack class expects 16 bit little endian underneath. I believe that for some reason the JLayer mp3 decoder will return 16bit little endian values. The decode() method in the question does a byte swap of the 16 bit values. Also, the posted audio sounds as if the bytes are swapped.
public byte[] decode(InputStream inputStream, int startMs, int maxMs, bool swap) throws IOException {
...
short[] pcm = output.getBuffer();
for (short s : pcm) {
if(swap) {
outStream.write(s & 0xff);
outStream.write((s >> 8) & 0xff);
} else {
outStream.write((s >> 8) & 0xff);
outStream.write(s & 0xff);
}
}
...
For 44k mp3s, you call the routine with swap = true;. For the 22k mp3 swap = false. This explains all the reported phenomena. I don't know why the JLayer mp3 decoder would sometimes output big endian and other times little endian. I imagine it depends on the source mp3 and not the sample rate.