Develop Reference

How to send image from matlab to android over bluetooth?

Problem:
I want to send an image from matlab to android over bluetooth.
Matlab and android are connected to each other and I can send strings without a problem.
fprintf(tabletObj, 'sleep');
I have a really huge byteArray containing the image I want to send to android. Here you can see just the first bytes:
planString = [-119,80,78,71,13,10,26,10,0,0,0,13,73,72,68,82,0,0,0,72,0,0,0,72,8,6,0,0,0,85,-19,-77,71,0,0,0,4,115,66,73,84,8,8,8,8,124,8,100,-120,0,0,29,2,73,68,65,84,120,-100,-75,-100,121,-68,37, ... ]
After that, I set in matlab the OutputBufferSize to the size of the image and send it to the tablet.
s = whos('planString');
obj1.OutputBufferSize = s.bytes;
% Send it to tablet
fwrite(tabletObj, planString, 'int8');
In android you can see following incoming bytes.
Why are there just the first 6 bytes and not more?
The next incoming bytes are more then just 6 bytes, why?
I set the buffersize in android to the same size like matlab.
private void listen() {
byte[] buffer = new byte[picSize]; // buffer store for the stream
Log.i(TAG, "buffer length" + buffer.length);
while (true) {
try {
inputStream.read(buffer);
newMessageReceived(new String(buffer, "UTF-8")); // Send the obtained bytes to the UI activity
} catch (IOException e) {
break;
}
}
}
Edit #1:
I used following code to get only the "right" bytes and put that into an ArrayList with bytes. Now, it seems like that I have just the needed bytes. But it's too slow! You need to wait for more than 1 min. to get all bytes from matlab. Is there a better solution? Why are the incoming bytes split sometimes in 3, sometimes in 15, ...? (see picture below code)
ArrayList<byte[]> bytes = new ArrayList<byte[]>();
...
int nread = inputStream.read(buffer);
byte[] newOne = new byte[nread];
System.arraycopy(buffer, 0, newOne, 0, nread);
bytes.add(newOne);

private void listen() {
byte[] buffer = new byte[10000];
int nbytes = 0;
while (true) {
try {
int nread = inputStream.read(buffer, nbytes, buffer.length - nbytes);
nbytes += nread;
... // after getting all bytes
newMessageReceived(buffer, nbytes); // Send bytes to the UI activity
} catch (IOException e) {
break;
}
}
}
#greenapps thanks for the solution.

Increase volume output of recorded audio

I am trying to make a call recording app in Android. I am using loudspeaker to record both uplink and downlink audio. The only problem I am facing is the volume is too low. I've increased the volume of device using AudioManager to max and it can't go beyond that.
I've first used MediaRecorder, but since it had limited functions and provides compressed audio, I've tried with AudioRecorder. Still I havn't figured out how to increase the audio. I've checked on projects on Github too, but it's of no use. I've searched on stackoverflow for last two weeks, but couldn't find anything at all.
I am quite sure that it's possible, since many other apps are doing it. For instance Automatic Call recorder does that.
I understand that I have to do something with the audio buffer, but I am not quite sure what needs to be done on that. Can you guide me on that.
Update:-
I am sorry that I forgot to mention that I am already using Gain. My code is almost similar to RehearsalAssistant (in fact I derived it from there). The gain doesn't work for more than 10dB and that doesn't increase the audio volume too much. What I wanted is I should be able to listen to the audio without putting my ear on the speaker which is what lacking in my code.
I've asked a similar question on functioning of the volume/loudness at SoundDesign SE here. It mentions that the Gain and loudness is related but it doesn't set the actual loudness level. I am not sure how things work, but I am determined to get the loud volume output.

You obviously have the AudioRecord stuff running, so I skip the decision for sampleRate and inputSource. The main point is that you need to appropriately manipulate each sample of your recorded data in your recording loop to increase the volume. Like so:
int minRecBufBytes = AudioRecord.getMinBufferSize( sampleRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT );
// ...
audioRecord = new AudioRecord( inputSource, sampleRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, minRecBufBytes );
// Setup the recording buffer, size, and pointer (in this case quadruple buffering)
int recBufferByteSize = minRecBufBytes*2;
byte[] recBuffer = new byte[recBufferByteSize];
int frameByteSize = minRecBufBytes/2;
int sampleBytes = frameByteSize;
int recBufferBytePtr = 0;
audioRecord.startRecording();
// Do the following in the loop you prefer, e.g.
while ( continueRecording ) {
int reallySampledBytes = audioRecord.read( recBuffer, recBufferBytePtr, sampleBytes );
int i = 0;
while ( i < reallySampledBytes ) {
float sample = (float)( recBuffer[recBufferBytePtr+i ] & 0xFF
| recBuffer[recBufferBytePtr+i+1] << 8 );
// THIS is the point were the work is done:
// Increase level by about 6dB:
sample *= 2;
// Or increase level by 20dB:
// sample *= 10;
// Or if you prefer any dB value, then calculate the gain factor outside the loop
// float gainFactor = (float)Math.pow( 10., dB / 20. ); // dB to gain factor
// sample *= gainFactor;
// Avoid 16-bit-integer overflow when writing back the manipulated data:
if ( sample >= 32767f ) {
recBuffer[recBufferBytePtr+i ] = (byte)0xFF;
recBuffer[recBufferBytePtr+i+1] = 0x7F;
} else if ( sample <= -32768f ) {
recBuffer[recBufferBytePtr+i ] = 0x00;
recBuffer[recBufferBytePtr+i+1] = (byte)0x80;
} else {
int s = (int)( 0.5f + sample ); // Here, dithering would be more appropriate
recBuffer[recBufferBytePtr+i ] = (byte)(s & 0xFF);
recBuffer[recBufferBytePtr+i+1] = (byte)(s >> 8 & 0xFF);
}
i += 2;
}
// Do other stuff like saving the part of buffer to a file
// if ( reallySampledBytes > 0 ) { ... save recBuffer+recBufferBytePtr, length: reallySampledBytes
// Then move the recording pointer to the next position in the recording buffer
recBufferBytePtr += reallySampledBytes;
// Wrap around at the end of the recording buffer, e.g. like so:
if ( recBufferBytePtr >= recBufferByteSize ) {
recBufferBytePtr = 0;
sampleBytes = frameByteSize;
} else {
sampleBytes = recBufferByteSize - recBufferBytePtr;
if ( sampleBytes > frameByteSize )
sampleBytes = frameByteSize;
}
}

Thanks to Hartmut and beworker for the solution. Hartmut's code did worked at near 12-14 dB. I did merged the code from the sonic library too to increase volume, but that increase too much noise and distortion, so I kept the volume at 1.5-2.0 and instead tried to increase gain. I got decent sound volume which doesn't sound too loud in phone, but when listened on a PC sounds loud enough. Looks like that's the farthest I could go.
I am posting my final code to increase the loudness. Be aware that using increasing mVolume increases too much noise. Try to increase gain instead.
private AudioRecord.OnRecordPositionUpdateListener updateListener = new AudioRecord.OnRecordPositionUpdateListener() {
#Override
public void onPeriodicNotification(AudioRecord recorder) {
aRecorder.read(bBuffer, bBuffer.capacity()); // Fill buffer
if (getState() != State.RECORDING)
return;
try {
if (bSamples == 16) {
shBuffer.rewind();
int bLength = shBuffer.capacity(); // Faster than accessing buffer.capacity each time
for (int i = 0; i < bLength; i++) { // 16bit sample size
short curSample = (short) (shBuffer.get(i) * gain);
if (curSample > cAmplitude) { // Check amplitude
cAmplitude = curSample;
}
if(mVolume != 1.0f) {
// Adjust output volume.
int fixedPointVolume = (int)(mVolume*4096.0f);
int value = (curSample*fixedPointVolume) >> 12;
if(value > 32767) {
value = 32767;
} else if(value < -32767) {
value = -32767;
}
curSample = (short)value;
/*scaleSamples(outputBuffer, originalNumOutputSamples, numOutputSamples - originalNumOutputSamples,
mVolume, nChannels);*/
}
shBuffer.put(curSample);
}
} else { // 8bit sample size
int bLength = bBuffer.capacity(); // Faster than accessing buffer.capacity each time
bBuffer.rewind();
for (int i = 0; i < bLength; i++) {
byte curSample = (byte) (bBuffer.get(i) * gain);
if (curSample > cAmplitude) { // Check amplitude
cAmplitude = curSample;
}
bBuffer.put(curSample);
}
}
bBuffer.rewind();
fChannel.write(bBuffer); // Write buffer to file
payloadSize += bBuffer.capacity();
} catch (IOException e) {
e.printStackTrace();
Log.e(NoobAudioRecorder.class.getName(), "Error occured in updateListener, recording is aborted");
stop();
}
}
#Override
public void onMarkerReached(AudioRecord recorder) {
// NOT USED
}
};

simple use MPEG_4 format
To increase the call recording volume use AudioManager as follows:
int deviceCallVol;
AudioManager audioManager;
Start Recording:
audioManager = (AudioManager)context.getSystemService(Context.AUDIO_SERVICE);
//get the current volume set
deviceCallVol = audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
//set volume to maximum
audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL), 0);
recorder.setAudioSource(MediaRecorder.AudioSource.VOICE_CALL);
recorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AAC);
recorder.setAudioEncodingBitRate(32);
recorder.setAudioSamplingRate(44100);
Stop Recording:
//revert volume to initial state
audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, deviceCallVol, 0);

In my app I use an open source sonic library. Its main purpose is to speed up / slow down speech, but besides this it allows to increase loudness too. I apply it to playback, but it must work for recording similarly. Just pass your samples through it before compressing them. It has a Java interface too. Hope this helps.

Android: AudioTrack makes a click sound at beginning

I'm working on an Android app and I would like to play some short sounds(~ 2s). I tried Soundpool but it doesn't really suit for me since it can't check if a sounds is already playing. So I decided to use AudioTrack.
It works quite good BUT most of the time, when it begins to play a sound there is a "click" sound.
I checked my audiofiles and they are clean.
I use audiotrack on stream mode. I saw that static mode is better for short sounds but after many searchs I still don't understand how to make it work.
I also read that the clicking noise can be caused by the header of the wav file, so maybe the sound would disappear if I skip this header with setPlaybackHeadPosition(int positionInFrames) function (that is supposed to work only in static mode)
Here is my code (so the problem is the ticking noise at the beginning)
int minBufferSize = AudioTrack.getMinBufferSize(44100, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, minBufferSize, AudioTrack.MODE_STREAM);
audioTrack.play();
int i = 0;
int bufferSize = 2048; //don't really know which value to put
audioTrack.setPlaybackRate(88200);
byte [] buffer = new byte[bufferSize];
//there we open the wav file >
InputStream inputStream = getResources().openRawResource(R.raw.abordage);
try {
while((i = inputStream.read(buffer)) != -1)
audioTrack.write(buffer, 0, i);
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
try {
inputStream.close();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
Does anyone has a solution to avoid that noise? I tried this, that works sometimes but not everytime. Could someone show me how to implement audiotrack in MODE_STATIC ?
Thank you

I found that Scott Stensland's reasoning was fitting my issue (thanks!).
I eliminated the pop by running a dead simple linear fade-in filter over the beginning of the sample array. The filter makes sample values start from 0 and slowly increase in amplitude to their original value. By always starting at a value of 0 at the zero cross over point the pop never occurs.
A similar fade-out filter was applied at the end of the sample array. The filter duration can easily be adjusted.
import android.util.Log;
public class FadeInFadeOutFilter
{
private static final String TAG = FadeInFadeOutFilter.class.getSimpleName();
private final int filterDurationInSamples;
public FadeInFadeOutFilter ( int filterDurationInSamples )
{
this.filterDurationInSamples = filterDurationInSamples;
}
public void filter ( short[] audioShortArray )
{
filter(audioShortArray, audioShortArray.length);
}
public void filter ( short[] audioShortArray, int audioShortArraySize )
{
if ( audioShortArraySize/2 <= filterDurationInSamples ) {
Log.w(TAG, "filtering audioShortArray with less samples filterDurationInSamples; untested, pops or even crashes may occur. audioShortArraySize="+audioShortArraySize+", filterDurationInSamples="+filterDurationInSamples);
}
final int I = Math.min(filterDurationInSamples, audioShortArraySize/2);
// Perform fade-in and fade-out simultaneously in one loop.
final int fadeOutOffset = audioShortArraySize - filterDurationInSamples;
for ( int i = 0 ; i < I ; i++ ) {
// Fade-in beginning.
final double fadeInAmplification = (double)i/I; // Linear ramp-up 0..1.
audioShortArray[i] = (short)(fadeInAmplification * audioShortArray[i]);
// Fade-out end.
final double fadeOutAmplification = 1 - fadeInAmplification; // Linear ramp-down 1..0.
final int j = i + fadeOutOffset;
audioShortArray[j] = (short)(fadeOutAmplification * audioShortArray[j]);
}
}
}

In my case. It was WAV-header.
And...
byte[] buf44 = new byte[44];
int read = inputStream.read(buf44, 0, 44);
...solved it.

A common cause of audio "pop" is due to the rendering process not starting/stopping sound at the zero cross over point (assuming min/max of -1 to +1 cross over would be 0). Transducers like speakers or ear-buds are at rest (no sound input) which maps to this zero cross level. If an audio rendering process fails to start/stop from/to this zero, the transducer is being asked to do the impossible, namely instantaneously go from its resting state to some non-zero position in its min/max movement range, (or visa versa if you get a "pop" at the end).

Finally, after a lot of experimentation, I made it work without the click noise. Here is my code (unfortunaly, I can't read the size of the inputStream since the getChannel().size() method only works with FileInputStream type)
try{
long totalAudioLen = 0;
InputStream inputStream = getResources().openRawResource(R.raw.abordage); // open the file
totalAudioLen = inputStream.available();
byte[] rawBytes = new byte[(int)totalAudioLen];
AudioTrack track = new AudioTrack(AudioManager.STREAM_MUSIC,
44100,
AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT,
(int)totalAudioLen,
AudioTrack.MODE_STATIC);
int offset = 0;
int numRead = 0;
track.setPlaybackHeadPosition(100); // IMPORTANT to skip the click
while (offset < rawBytes.length
&& (numRead=inputStream.read(rawBytes, offset, rawBytes.length-offset)) >= 0) {
offset += numRead;
} //don't really know why it works, it reads the file
track.write(rawBytes, 0, (int)totalAudioLen); //write it in the buffer?
track.play(); // launch the play
track.setPlaybackRate(88200);
inputStream.close();
}
catch (FileNotFoundException e) {
Log.e(TAG, "Error loading audio to bytes", e);
} catch (IOException e) {
Log.e(TAG, "Error loading audio to bytes", e);
} catch (IllegalArgumentException e) {
Log.e(TAG, "Error loading audio to bytes", e);
}
So the solution to skip the clicking noise is to use MODE_STATIC and setPlaybackHeadPosition function to skip the beginning of the audio file (that is probably the header or I don't know what).
I hope that this part of code will help someone, I spent too many time trying to find a static mode code sample without finding a way to load a raw ressource.
Edit: After testing this solution on various devices, it appears that they have the clicking noise anyway.

For "setPlaybackHeadPosition" to work, you have to play and pause first. It doesn't work if your track is stopped or not started. Trust me. This is dumb. But it works:
track.play();
track.pause();
track.setPlaybackHeadPosition(100);
// then continue with track.write, track.play, etc.

Difference between Android and Matlab (reading a wav file into array )

I am trying to read a wav file into an array using Android. In order to validate the results I read the same wav file using Matlab. The problem is that the values are different. Your help is highly appreciated in solving this problem.
Kindly, find below the Matlab and Android code with the associated results:
Matlab Code:
fName = 'C:\Users\me\Desktop\audioText.txt';
fid = fopen(fName,'w');
dlmwrite(fName,y_sub,'-append','delimiter','\t','newline','pc');
Matlab Results:
0.00097656
0.00045776
0.0010681
0.00073242
0.00054932
-0.00064087
0.0010376
-0.00027466
-0.00036621
-9.1553e-05
0.00015259
0.0021362
-0.00024414
-3.0518e-05
-0.00021362
Android Code:
String filePath;
private static DataOutputStream fout;
ByteArrayOutputStream out;
BufferedInputStream in;
filePath = "mnt/sdcard/audio.wav";
out = new ByteArrayOutputStream();
try {
in = new BufferedInputStream(new FileInputStream(filePath));
} catch (FileNotFoundException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
int read;
byte[] buff = new byte[2000000];
try {
while ((read = in.read(buff)) > 0)
{
out.write(buff, 0, read);
}
} catch (IOException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
try {
out.flush();
} catch (IOException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
byte[] audioBytes = out.toByteArray();
}
Android Results:
82, 73, 70, 70, 92, 108, 40, 0, 87, 65, 86, 69, 102, 109
Thanks,

In Android you're reading the file header, not the actual values of the sound samples. Your values in Android are ASCII for
RIFF\l( WAVEfm
In Matlab I'm not sure what you're doing... looks like you're writing, not reading a file.

The dir command is quite helpful here. It either displays the whole content of a directory but you can also specify a glob to just return a sub-set of files, e.g. dir('*.wav'). This returns an struct-array containing file information such as name, date, bytes, isdir and so on.
To get started, try the following:
filelist = dir('*.wav');
for file = filelist
fprintf('Processing %s\n', file.name);
fid = fopen(file.name);
% Do something here with your file.
fclose(fid);
end
If a processing result has to be stored per file,
I often use the following pattern. I usually pre-allocate an array, a struct array or
a cell array of the same size as the filelist. Then I use an integer index to iterate
over the file list, which I can also use to write the output. If the information to be
stored is homogeneous (e.g. one scalar per file), use an array or a struct array.
However, if the information differs from file to file (e.g. vectors or matrices of different size) use a cell array instead.
An example using an ordinary array:
filelist = dir('*.wav');
% Pre-allocate an array to store some per-file information.
result = zeros(size(filelist));
for index = 1 : length(filelist)
fprintf('Processing %s\n', filelist(index).name);
% Read the sample rate Fs and store it.
[y, Fs] = wavread(filelist(index).name);
result(index) = Fs;
end
% result(1) .. result(N) contain the sample rates of each file.
An example using a cell array:
filelist = dir('*.wav');
% Pre-allocate a cell array to store some per-file information.
result = cell(size(filelist));
for index = 1 : length(filelist)
fprintf('Processing %s\n', filelist(index).name);
% Read the data of the WAV file and store it.
y = wavread(filelist(index).name);
result{index} = y;
end
% result{1} .. result{N} contain the data of the WAV files.

I am not sure what is the problem exactly, but I got the correct readings when I used the following code:
File filein = new File(filePath, "audio.wav");
try
{
// Open the wav file specified as the first argument
WavFile wavFile = WavFile.openWavFile(filein);
// Display information about the wav file
wavFile.display();
// Get the number of audio channels in the wav file
int numChannels = wavFile.getNumChannels();
// Create a buffer of 100 frames
double[] buffer = new double[20000 * numChannels];
int framesRead;
double min = Double.MAX_VALUE;
double max = Double.MIN_VALUE;
do
{
// Read frames into buffer
framesRead = wavFile.readFrames(buffer, 20000);
// Loop through frames and look for minimum and maximum value
for (int s=0 ; s<framesRead * numChannels ; s++)
{
if (buffer[s] > max) max = buffer[s];
if (buffer[s] < min) min = buffer[s];
}
}
while (framesRead != 0);
// Close the wavFile
wavFile.close();
// Output the minimum and maximum value
System.out.printf("Min: %f, Max: %f\n", min, max);
}
catch (Exception e)
{
System.err.println(e);
}

FSK Modulation and Playing sine Tone in Android

I want to do some FSK Modulation over the audio port. So the problem is that my sinus wave isn't very good. It is disturb by even parts. I used the code original from http://marblemice.blogspot.com/2010/04/generate-and-play-tone-in-android.html with the further modification from Playing an arbitrary tone with Android and https://market.android.com/details?id=re.serialout&feature=search_result .
So where is the failure? What do I wrong?
private static int bitRate=300;
private static int sampleRate=48000;
private static int freq1=600;
public static void loopOnes(){
playque.add(UARTHigh);
athread.interrupt();
}
private static byte[] UARTHigh() {
int numSamples=sampleRate/bitRate;
double sample[]=new double[numSamples];
byte[] buffer=new byte[numSamples*2];
for(int i=0; i<numSamples;++i){
sample[i]=Math.sin(2*Math.PI*i*freq1/sampleRate);
}
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
buffer[idx++] = (byte) (val & 0x00ff);
buffer[idx++] = (byte) ((val & 0xff00) >>> 8);
}
return buffer;
}
private static void playSound(){
active = true;
while(active)
{
try {Thread.sleep(Long.MAX_VALUE);} catch (InterruptedException e) {
while (playque.isEmpty() == false)
{
if (atrk != null)
{
if (generatedSnd != null)
{
// Das letzte Sample erst fertig abspielen lassen
// systemClock.sleep(xx) xx könnte angepasst werden
while (atrk.getPlaybackHeadPosition() < (generatedSnd.length))
SystemClock.sleep(50); // let existing sample finish first: this can probably be set to a smarter number using the information above
}
atrk.release();
}
UpdateParameters(); // might as well do it at every iteration, it's cheap
generatedSnd = playque.poll();
length = generatedSnd.length;
if (minbufsize<length)
minbufsize=length;
atrk = new AudioTrack(AudioManager.STREAM_MUSIC,
sampleRate, AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT, minbufsize,
AudioTrack.MODE_STATIC);
atrk.setStereoVolume(1,1);
atrk.write(generatedSnd, 0, length);
atrk.play();
}
// Playque is Empty =>send StopBit!
// Set Loop Points
int setLoopError=atrk.setLoopPoints(0, length, -1);
atrk.play();
}
}
}
}

So the answer is to change from MODE_STATIC to MODE_STREAM and don't use Looping Points. In a new thread with low priority a busy loop writes the tracks.