Develop Reference

Why Can't I Play Raw Audio Bytes Using AudioTrack's Static Mode?

I have an Android app where there is some raw audio bytes stored in a variable.
If I use an AudioTrack to play this audio data, it only works if I use AudioTrack.MODE_STREAM:
byte[] recordedAudioAsBytes;
public void playButtonPressed(View v) {
// this verifies that audio data exists as expected
for (int i=0; i<recordedAudioAsBytes.length; i++) {
Log.i("ABC", "byte[" + i + "] = " + recordedAudioAsBytes[i]);
}
// STREAM MODE ACTUALLY WORKS!!
/*
AudioTrack player = new AudioTrack(AudioManager.STREAM_MUSIC, SAMPLERATE, CHANNELS,
ENCODING, MY_CHOSEN_BUFFER_SIZE, AudioTrack.MODE_STREAM);
player.play();
player.write(recordedAudioAsBytes, 0, recordedAudioAsBytes.length);
*/
// STATIC MODE DOES NOT WORK
AudioTrack player = new AudioTrack(AudioManager.STREAM_MUSIC, SAMPLERATE, PLAYBACK_CHANNELS,
ENCODING, MY_CHOSEN_BUFFER_SIZE, AudioTrack.MODE_STATIC);
player.write(recordedAudioAsBytes, 0, recordedAudioAsBytes.length);
player.play();
}
If I use AudioTrack.MODE_STATIC, the output is glitchy -- it just makes a nasty pop and sounds very short with hardly anything audible.
So why is that? Does STATIC_MODE require that the audio data have a header?
That's all I can think of.
If you'd like to see all the code, check this question.

It seems to me that you are using the same MY_CHOSEN_BUFFER_SIZE for 'streaming' and 'static' mode!? This might explain why it sounds short...
In order to use Audiotracks 'static-mode' you have to use the size of your Byte-Array (bigger will also work) as buffersize. The Audio will be treated as one big chunk of data.
See: AudioTrack.Builder
setBufferSizeInBytes()... "If using the AudioTrack in static mode (see AudioTrack#MODE_STATIC), this is the maximum size of the sound that will be played by this instance."

Android DSP - Trouble with AudioTrack and Sub-Audible Signals

I'm having some trouble getting the AudioTrack class to do what I want it to do. I'm trying to make an app that first lets the user draw a single-cycle waveform on the screen, and then outputs an arbitrary number of cycles of that waveform at an arbitrary frequency through the headphone out. For the primary use, the frequency will be below the audible range, something like 1/60Hz - 20Hz (If anyone's familiar with Eurorack/modular synths, I'm hoping to use the headphone out as a CV source).
The problem I'm having is that the AudioTrack seems to output a highly inaccurate reproduction of the waveform at the low end of this frequency range, even though it does output an accurate reproduction at the high end. The lower the frequency gets, the more the waveform gets 'squished' to the left on my oscilloscope. The pictures below show this phenomenon on a waveform that is supposed to start each cycle low and increase linearly to a max value, but this phenomenon happens with other waveshapes too.
So far, I have the app set up (1) to create an ArrayList to hold the data points from the user's input, (2) to convert the ArrayList into a float[] to feed to the AudioTrack, and (3) to setup the AudioTrack and write the float[] to it when needed. Since the ArrayList and float[] both maintain an accurate reproduction of the waveform, I'm pretty sure my problem is at (3), or else it's a hardware limitation of my phone/scope.
Here's the relevant code for the AudioTrack.
Method that (re)initializes the AudioTrack:
public void updateTrackForWave(Waveform wave, Context context) {
if (wave.getInterpolatedWaveData() == null) return;
int sampleRate = Integer.parseInt(
((AudioManager) context.getSystemService(Context.AUDIO_SERVICE))
.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE));
if (wave.getOutputChannel() == Waveform.OutputChannelEnum.left) {
if (mTrackL != null) mTrackL.release();
mTrackL = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_FLOAT,
Float.BYTES * wave.getInterpolatedWaveData().length,
AudioTrack.MODE_STATIC);
mTrackL.setVolume(AudioTrack.getMaxVolume());
} else if (wave.getOutputChannel() == Waveform.OutputChannelEnum.right) {
if (mTrackR != null) mTrackR.release();
mTrackR = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_FLOAT,
Float.BYTES * wave.getInterpolatedWaveData().length,
AudioTrack.MODE_STATIC);
mTrackR.setVolume(AudioTrack.getMaxVolume());
}
}
Method that plays AudioTrack:
private void playWaveOnTrack(Waveform wave, AudioTrack track) {
if (track == null) return;
if (track.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
return;
} else {
if (wave.isCycleMode()) track.setLoopPoints(0,
wave.getInterpolatedWaveData().length, -1);
else if (wave.isOneShotMode()) track.setLoopPoints(0,
wave.getInterpolatedWaveData().length, 0);
if (track.getPlayState() == AudioTrack.PLAYSTATE_PAUSED) {
track.play();
} else if (track.getPlayState() == AudioTrack.PLAYSTATE_STOPPED) {
track.write(wave.getInterpolatedWaveData(), 0,
wave.getInterpolatedWaveData().length, AudioTrack.WRITE_BLOCKING);
track.play();
}
}
}
It's worth noting that wave.getInterpolatedWaveData() returns the float[] from (3) above.
Anyways, any help you could spare on this would be greatly appreciated! Not sure if I've made a mistake in the code I have, or if there's some code I should add (maybe an AudioEffect of some kind?), or if I'm asking too much of my phone, or what.
PS I'm new here and to Android programming in general, so please do point out any forum norms I should be following but am not, or any alternative coding approaches I might not know about.
Pictures:
Waveform at 100Hz
Waveform at 10Hz
Waveform at 2Hz

It's a hardware limitation of your phone. In order to avoid a constant DC current flowing through each headphone speaker, they are capacitively coupled -- each channel is in series with one or more capacitors. This gives your output waveform a time constant; the average voltage level of the waveform will always tend towards 0 (GND), which makes it impossible to output DC, or even low-frequency signals.

How To Record Sound in Android with Better Quality and Reduce Noise

I’m trying to build a music analytics app for android platform.
the app is using MediaRecorder.AudioSource.MIC
to record the music form the MIC and them encode it PCM 16BIT with 11025 freq, but the recorded audio sample are very low quality is there any way to make it better, decrease the noise?
mRecordInstance = new AudioRecord(MediaRecorder.AudioSource.MIC,FREQUENCY, CHANNEL,ENCODING, minBufferSize);
mRecordInstance.startRecording();
do
{
samplesIn += mRecordInstance.read(audioData, samplesIn, bufferSize - samplesIn);
if(mRecordInstance.getRecordingState() == AudioRecord.RECORDSTATE_STOPPED)
break;
}
while (samplesIn < bufferSize);
Thanks in Advance

The solution above didnt work for me.
So, i searched around and found this article.
Long story short, I used MediaRecorder.AudioSource.VOICE_RECOGNITION instead of AudioSource.MIC, which gave me really good results and noise in the background did reduce very much.
The great thing about this solution is, it can be used with both AudioRecord and MediaRecorder class.

The best combination of SR and buffer size is very device dependant, so your results will vary depending on the hardware. I use this utility to figure out what the best combination is for devices running Android 4.2 and above;
public static DeviceValues getDeviceValues(Context context) {
try {
AudioManager am = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
try {
Method getProperty = AudioManager.class.getMethod("getProperty", String.class);
Field bufferSizeField = AudioManager.class.getField("PROPERTY_OUTPUT_FRAMES_PER_BUFFER");
Field sampleRateField = AudioManager.class.getField("PROPERTY_OUTPUT_SAMPLE_RATE");
int bufferSize = Integer.valueOf((String)getProperty.invoke(am, (String)bufferSizeField.get(am)));
int sampleRate = Integer.valueOf((String)getProperty.invoke(am, (String)sampleRateField.get(am)));
return new DeviceValues(sampleRate, bufferSize);
} catch(NoSuchMethodException e) {
return selectBestValue(getValidSampleRates(context));
}
} catch(Exception e) {
return new DeviceValues(DEFAULT_SAMPLE_RATE, DEFAULT_BUFFER_SIZE);
}
}
This uses reflection to check if the getProperty method is available, because this method was introduced in API level 17. If you are developing for a specific device type, you might want to experiment with various buffer sizes and sample rates. The defaults that I use as a fallback are;
private static final int DEFAULT_SAMPLE_RATE = 22050;
private static final int DEFAULT_BUFFER_SIZE = 1024;
Additionally I check the various SR by seeing if getMinBufferSize returns a reasonable value for use;
private static List<DeviceValues> getValidSampleRates(Context context) {
List<DeviceValues> available = new ArrayList<DeviceValues>();
for (int rate : new int[] {8000, 11025, 16000, 22050, 32000, 44100, 48000, 96000}) { // add the rates you wish to check against
int bufferSize = AudioRecord.getMinBufferSize(rate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT);
if (bufferSize > 0 && bufferSize < 2048) {
available.add(new DeviceValues(rate, bufferSize * 2));
}
}
return available;
}
This depends on the logic that if getMinBufferSize returns 0, the sample rate is not available in the device. You should experiment with these values for your particular use case.

Though it is an old question following solution will be helpful.
We can use MediaRecorder to record audio with ease.
private void startRecording() {
MediaRecorder recorder = new MediaRecorder();
recorder.setAudioSource(MediaRecorder.AudioSource.MIC);
recorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AAC);
recorder.setAudioEncodingBitRate(96000)
recorder.setAudioSamplingRate(44100)
recorder.setOutputFile(".../audioName.m4a");
try {
recorder.prepare();
} catch (IOException e) {
Log.e(LOG_TAG, "prepare() failed");
}
recorder.start();
}
Note:
MediaRecorder.AudioEncoder.AAC is used as MediaRecorder.AudioEncoder.AMR_NB encoding is no longer supported in iOS. Reference
AudioEncodingBitRate should be used either 96000 or 128000 as required for clarity of sound.

Record audio loops through sample rates?

I am trying to read and understand audio in Android. In my search I came along this article. Here he has wrote a code to record audio in wav format. But there is one thing I don't fully understand, and that is the first loop of his code:
public class ExtAudioRecorder
{
private final static int[] sampleRates = {44100, 22050, 11025, 8000};
public static ExtAudioRecorder getInstanse(Boolean recordingCompressed)
{
ExtAudioRecorder result = null;
if(recordingCompressed)
{
result = new ExtAudioRecorder( false,
AudioSource.MIC,
sampleRates[3],
AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT);
}
else
{
int i=0;
do
{
result = new ExtAudioRecorder( true,
AudioSource.MIC,
sampleRates[i],
AudioFormat.CHANNEL_CONFIGURATION_MONO,
AudioFormat.ENCODING_PCM_16BIT);
} while((++i<sampleRates.length) & !(result.getState() == ExtAudioRecorder.State.INITIALIZING));
}
return result;
He gives like a basic information about it, but I don't get this completely. Does this has anything to do with the performance of different types of Android devices? Anyway, hope somebody can brighten this up for me :)

He is trying to initialize the audio recorder with different sample rates, from these {44100, 22050, 11025, 8000}.
Depending on the underlying hardware, not all sample rates may be supported by the device.
Although the documentation says:
"44100Hz is currently the only rate that is guaranteed to work on all devices, but other rates such as 22050, 16000, and 11025 may work on some devices."
I think the author has written code to make sure that if initialization at a sample rate fails, an attempt is made to initialize at some other sample rate, unless the initialization is successful, which is given by the check he is making in the loop condition.

android java audio dsp sites or android sound library?

anyone know of any usefull links for learning audio dsp for android?
or a sound library?
im trying to make a basic mixer for playing wav files but realised i dont know enough about dsp, and i cant find anything at all for android.
i have a wav file loaded into a byte array and an AudioTrack on a short loop.
how can i feed the data in?
i expect this post will be ignored but its worth a try.
FileInputStream is = new FileInputStream(filePath);
BufferedInputStream bis = new BufferedInputStream(is);
DataInputStream dis = new DataInputStream(bis);
int i = 0;
while (dis.available() > 0) {
byteData[i] = dis.readByte(); //byteData
i++;
}
final int minSize = AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT );
track = new AudioTrack( AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT,
minSize, AudioTrack.MODE_STREAM);
track.play();
bRun=true;
new Thread(new Runnable() {
public void run() {
track.write(byteData, 0, minSize);
}
}).start();

I'll give this a shot just because I was in your position a few months ago...
If you already have the wav file audio samples in a byte array, you simple need to pass the samples to the audio track object (lookup the write() methods).
To mix audio together you simply add the sames from each track. For example, add the first sample from track 1 to track 2, add the second sample from track 1 to track 2 and so on. The end result would ideally be a third array containing the added samplws which you pass to the 'write' method of your audio track instance.
You must be mindful of clipping here. If your data type 'short' then the maximum value allowed is 32768. A simple way to ensure that your added samples do not exceed this limit is to peform the addition and store the result in a variable whose data type is larger than a short (eg. int) and evaluate the result. If it's greater than 32768 then make it equal to 32768 and cast it back to a short.
int result = track1[i] + track2[i];
if(result > 32768) {
result = 32768;
}
else if(result < -32768) {
result = -32768;
}
mixedAudio[i] = (short)result;
Notice how the snippet above also tests for the minimum range of a short.
Appologies for the lack of formatting here, I'm on my mobile phone on a train :-)
Good luck.