Develop Reference

Android force AudioRecord to use headphone mic

I use AudioRecord to record music but when I record it uses the phone mic.
how can I force him to use the channel of the Headphone?
I use this code:
int minSize = AudioRecord.getMinBufferSize(8000, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT);
AudioRecord ar = new AudioRecord(MediaRecorder.AudioSource.DEFAULT, 8000, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, minSize);
new Handler().postDelayed(new Runnable() {
#Override
public void run() {
recorder[0] = false;
}
}, timeInSecondsToRecord * 1000);
short[] buffer = new short[minSize];
ar.startRecording();
Log.d("Started","Reording");
while (recorder[0]) {
ar.read(buffer, 0, minSize);
for (short s : buffer) {
if (s>1000)
System.out.println("signalVal=" + s);
}
}
Log.d("Finished","Reording");
ar.stop();
Thank you

You can use AudioManager.isWiredHeadsetOn() for checking if the headset are plugged in or not. If the above value is false dont perform any action or whatever you want to do. And also you need permission first: MODIFY_AUDIO_SETTINGS
Hope this helps. :)

I had this problem in reverse (trying to force use of the phone's mic instead of the headphone mic). You can choose a mic by scanning the AudioDeviceInfo array in AudioManager. Here's what I think it would look like for you (if TYPE_LINE_ANALOG doesn't work, check out the AudioDeviceInfo page for more possibilities or debug what devices come up):
AudioRecord audioRecord =
new AudioRecord(...);
// Force use of the line in mic
AudioManager audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
for ( AudioDeviceInfo device : audioManager.getDevices(AudioManager.GET_DEVICES_INPUTS)) {
if ( device.getType() == AudioDeviceInfo.TYPE_LINE_ANALOG) {
audioRecord.setPreferredDevice(device);
break;
}
}

AudioTrack not playing samples at button clicked sometimes

My android OS is Android M. Nexus 6.
I implemented a AndroidSpeakerWriter as
public class AndroidSpeakerWriter {
private final static String TAG= "AndroidSpeakerWriter";
private AudioTrack audioTrack;
short[] buffer;
public AndroidSpeakerWriter() {
buffer = new short[1024];
}
public void init(int sampleRateInHZ){
int minBufferSize = AudioTrack.getMinBufferSize(sampleRateInHZ,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHZ,
AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT, minBufferSize,
AudioTrack.MODE_STREAM); // 0-static 1-stream
}
public void fillBuffer(short[] samples) {
if (buffer.length<samples.length) {
buffer = new short[samples.length];
}
System.arraycopy(samples, 0, buffer, 0, samples.length);
}
public void writeSamples(short[] samples) {
fillBuffer(samples);
audioTrack.write(buffer, 0, samples.length);
}
public void stop() {
audioTrack.stop();
}
public void play() {
audioTrack.play();
}
}
Then I just send samples when I click a button
public void play(final short[] signal) {
if (signal == null){
Log.d(TAG, "play: a null signal");
return;
}
Thread t = new Thread(new Runnable() {
#Override
public void run() {
android.os.Process
.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
androidSpeakerWriter.play();
androidSpeakerWriter.writeSamples(signal);
androidSpeakerWriter.stop();
}
});
t.start();
}
The problem is the device does not beep every time I click the button.
Sometimes it works, sometimes it doesn't.
There is no such a problem when I run this on an old nexus galaxy phone android 4.3. Anybody has encountered a similar problem? Thanks in advance for any help.
One thing is that currently my beep is pretty short (256 samples), not even close to the minBufferSize.
The bufferSizeInBytes in the constructor of AudioTrack for static mode should be the audio sample length you wanna play according to the vague document.
So is it still has a minimal size constraint on the buffer even for static mode? Why a nexus galaxy can play a 256 sample audio in static mode and a nexus 6 can not.
I use AudioManager to get the native buffer size/ sampling rate
nexus galaxy: 144/44100 nexus 6: 192/48000
I found those related:
AudioRecord and AudioTrack latency
Does AudioTrack buffer need to be full always in streaming mode?
https://github.com/igorski/MWEngine/wiki/Understanding-Android-audio-towards-achieving-low-latency-response

I believe it is caused by improper synchronization between thread. Your androidSpeakerWriter instance is running continously in different thread calling play(), writeSamples(), stop() respectively. Click of button will trigger creation of new thread with same androidSpeakerWriter instance.
So while Thread A is executing androidSpeakerWriter.play(), Thread B might be executing androidSpeakerWriter.writeSamples() which might overwrite current audio data being played.
Try
synchronized(androidSpeakerWriter) {
androidSpeakerWriter.play();
androidSpeakerWriter.writeSamples(signal);
androidSpeakerWriter.stop();
}
MODE_STREAM is used if you must play long audio data that will not fit into memory. If you need to play short audio file such beep sound, you can use MODE_STATIC when creating AudioTrack. then change your playback code such following:
synchronized(androidSpeakerWriter) {
androidSpeakerWriter.writeSamples(signal);
androidSpeakerWriter.play();
}

Can we check the state of microphone of an android phone, if we can then how can I do this

I want to write a program to check if the internal microphone of android phone is on, off or in use by some other application.
If this is possible then how can I do this?
I read related questions at stack overflow but did not find a solution.

Here's what I'm using to check if the microphone is busy (based on Odaym answer and my own tests):
(Updated with Android 6.0 Marshmallow compatibility, as suggested in comments)
public static boolean checkIfMicrophoneIsBusy(Context ctx){
AudioRecord audio = null;
boolean ready = true;
try{
int baseSampleRate = 44100;
int channel = AudioFormat.CHANNEL_IN_MONO;
int format = AudioFormat.ENCODING_PCM_16BIT;
int buffSize = AudioRecord.getMinBufferSize(baseSampleRate, channel, format );
audio = new AudioRecord(MediaRecorder.AudioSource.MIC, baseSampleRate, channel, format, buffSize );
audio.startRecording();
short buffer[] = new short[buffSize];
int audioStatus = audio.read(buffer, 0, buffSize);
if(audioStatus == AudioRecord.ERROR_INVALID_OPERATION || audioStatus == AudioRecord.STATE_UNINITIALIZED /* For Android 6.0 */)
ready = false;
}
catch(Exception e){
ready = false;
}
finally {
try{
audio.release();
}
catch(Exception e){}
}
return ready;
}

If you are using an AudioRecord object to record audio, like:
AudioRecord audio = new AudioRecord(MediaRecorder.AudioSource.MIC,
Constants.SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO,
AudioFormat.ENCODING_PCM_16BIT,Constants.BUFFER_SIZE_BYTES);
audio.startRecording();
Then right after audio.startRecording(), you're going to have to provide a buffer for reading the audio data into, and begin reading. You do that with:
int audioStatus = audio.read(bufferObject, 0, bufferSize);
The Android documentation for read() mentions the return value ERROR_INVALID_OPERATION (Constant Value: -3), this is only returned when the Mic is busy so you can check for that in your code and show a message that the Audio source is busy with another app.

As far as I know, there is no way to know the microphone's state (Busy, Available,..). Sorry

Playing music with AudioTrack buffer by buffer on Eclipse - no sound

i'm programming for Android 2.1.Could you help me with the following problem?
I have three files, and the general purpose is to play a sound with audiotrack buffer by buffer. I'm getting pretty desperate here because I tried about everything, and there still is no sound coming out of my speakers (while android's integrated mediaplayer has no problem playing sounds via the emulator).
Source code:
An audioplayer class, which implements the audio track. It will receive a buffer, in which the sound is contained.
public AudioPlayer(int sampleRate, int channelConfiguration, int audioFormat) throws ProjectException {
minBufferSize = AudioTrack.getMinBufferSize(sampleRate, channelConfiguration, audioFormat);
audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRate, channelConfiguration,
audioFormat, minBufferSize, AudioTrack.MODE_STREAM);
if(audioTrack == null)
throw new ProjectException("Erreur lors de l'instantiation de AudioTrack");
audioTrack.setStereoVolume((float)1.0, (float)1.0);
}
#Override
public void addToQueue(short[] buffer) {
audioTrack.write(buffer, 0, buffer.length*Short.SIZE);
if(!isPlaying ) {
audioTrack.play();
isPlaying = true;
}
}
A model class, which I use to fill the buffer. Normally, it would load sound from a file, but here it just uses a simulator (440Hz), for debugging.
Buffer sizes are chosen very loosely; normally first buffer size should be 6615 and then 4410. That's, again, only for debug.
public void onTimeChange() {
if(begin) {
//First fill about 300ms
begin = false;
short[][] buffer = new short[channels][numFramesBegin];
//numFramesBegin is for example 10000
//For debugging only buffer[0] is useful
fillSimulatedBuffer(buffer, framesRead);
framesRead += numFramesBegin;
audioPlayer.addToQueue(buffer[0]);
}
else {
try {
short[][] buffer = new short[channels][numFrames];
//Afterwards fill like 200ms
fillSimulatedBuffer(buffer, framesRead);
framesRead += numFrames;
audioPlayer.addToQueue(buffer[0]);
} catch (Exception e) {
e.printStackTrace();
}
}
}
private short simulator(int time, short amplitude) {
//a pure A (frequency=440)
//this is probably wrong due to sampling rate, but 44 and 4400 won't work either
return (short)(amplitude*((short)(Math.sin((double)(simulatorFrequency*time)))));
}
private void fillSimulatedBuffer(short[][] buffer, int offset) {
for(int i = 0; i < buffer[0].length; i++)
buffer[0][i] = simulator(offset + i, amplitude);
}
A timeTask class that calls model.ontimechange() every 200 ms.
public class ReadMusic extends TimerTask {
private final Model model;
public ReadMusic(Model model) {
this.model = model;
}
#Override
public void run() {
System.out.println("Task run");
model.onTimeChange();
}
}
What debugging showed me:
timeTask works fine, it does its job;
Buffer values seem coherent, and buffer size is bigger than minBufSize;
Audiotrack's playing state is "playing"
no exceptions are caught in model functions.
Any ideas would be greatly appreciated!

OK I found the problem.
There is an error in the current AudioTrack documentation regarding AudioTrack and short buffer input: the specified buffer size should be the size of the buffer itself (buffer.length) and not the size in bytes.

Android - Getting audio to play through earpiece

I currently have code that reads a recording in from the devices mic using the AudioRecord class and then playing it back out using the AudioTrack class.
My problem is that when I play it out it plays via the speaker phone.
I want it to play out via the ear piece on the device.
Here is my code:
public class LoopProg extends Activity {
boolean isRecording; //currently not used
AudioManager am;
int count = 0;
/** Called when the activity is first created. */
#Override
public void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
am = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
am.setMicrophoneMute(true);
while(count <= 1000000){
Record record = new Record();
record.run();
count ++;
Log.d("COUNT", "Count is : " + count);
}
}
public class Record extends Thread{
static final int bufferSize = 200000;
final short[] buffer = new short[bufferSize];
short[] readBuffer = new short[bufferSize];
public void run() {
isRecording = true;
android.os.Process.setThreadPriority(android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
int buffersize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT);
AudioRecord arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize);
AudioTrack atrack = new AudioTrack(AudioManager.STREAM_MUSIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize, AudioTrack.MODE_STREAM);
am.setRouting(AudioManager.MODE_NORMAL,1, AudioManager.STREAM_MUSIC);
int ok = am.getRouting(AudioManager.ROUTE_EARPIECE);
Log.d("ROUTING", "getRouting = " + ok);
setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
//am.setSpeakerphoneOn(true);
Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn());
am.setSpeakerphoneOn(false);
Log.d("SPEAKERPHONE", "Is speakerphone on? : " + am.isSpeakerphoneOn());
atrack.setPlaybackRate(11025);
byte[] buffer = new byte[buffersize];
arec.startRecording();
atrack.play();
while(isRecording) {
arec.read(buffer, 0, buffersize);
atrack.write(buffer, 0, buffer.length);
}
arec.stop();
atrack.stop();
isRecording = false;
}
}
}
As you can see if the code I have tried using the AudioManager class and its methods including the deprecated setRouting method and nothing works, the setSpeakerphoneOn method seems to have no effect at all, neither does the routing method.
Has anyone got any ideas on how to get it to play via the earpiece instead of the spaker phone?

Just got it to work on 2.2. I still needed the In_Call setup which I don't really like but I'll deal with it for now. I was able to ditch the call routing stuff which is deprecated now. I found you definitely need the Modify_Audio_Settings permission, no error without it but it the setSpeakerPhone method just does nothing. Here is the mock up of the code I used.
private AudioManager m_amAudioManager;
m_amAudioManager = (AudioManager)getSystemService(Context.AUDIO_SERVICE);
m_amAudioManager.setMode(AudioManager.MODE_IN_CALL);
m_amAudioManager.setSpeakerphoneOn(false);

Please use this code, works well:
//PLAY ON EARPIECE
mPlayer.setAudioStreamType(AudioManager.STREAM_VOICE_CALL);
audioManager.setMode(AudioManager.MODE_IN_CALL);
audioManager.setSpeakerphoneOn(false);
//PLAY ON SPEAKER
mPlayer.setAudioStreamType(AudioManager.STREAM_MUSIC);
audioManager.setMode(AudioManager.MODE_IN_CALL);
audioManager.setSpeakerphoneOn(true);

There was some related discussion in this recent question:
Android - can I mute currently playing audio applications?
Based on the AudioManager source code, it seems that you must be in "call mode" before the setSpeakerphoneOn method has any effect.
However, I recently saw an application that could seamlessly switch between earpiece and speakerphone while still showing the current stream as the "media" stream, so I would be interested in any further answers.

Misleaded by some answers here for quite a lot of time. I'm using Android 2.2. "audioManager.setSpeakerphoneOn(false);" is working.
audioManager.setSpeakerphoneOn(false);
...
mediaPlayer.setDataSource(..);
mediaPlayer.setAudioStreamType(AudioManager.STREAM_VOICE_CALL);
mediaPlayer.prepare();

public MediaPlayer m_mpSpeakerPlayer;
private AudioManager m_amAudioManager;
m_amAudioManager = (AudioManager)getSystemService(Context.AUDIO_SERVICE);
// 從Receiver Earpiece發音
m_amAudioManager.setSpeakerphoneOn(false);
m_amAudioManager.setRouting(AudioManager.MODE_NORMAL, AudioManager.ROUTE_EARPIECE, AudioManager.ROUTE_ALL);
Log.i(TAG, String.valueOf(m_amAudioManager.getRouting(AudioManager.ROUTE_EARPIECE)));
setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
// 把聲音設定成從Earpiece出來
// 重點在這行，要把它設定為正在通話中
m_amAudioManager.setMode(AudioManager.MODE_IN_CALL);
// 開始放音樂
m_mpSpeakerPlayer.reset();
m_mpSpeakerPlayer.setDataSource("sdcard/receiver.mp3");
m_mpSpeakerPlayer.prepare();
m_mpSpeakerPlayer.start();
//最後再把它設定從Speaker放音，達成！
m_amAudioManager.setMode(AudioManager.MODE_NORMAL);

I appear to have got it working on 1.6.
So I said I'd post here how I done it.
To get it working in 1.6 I:
Used the AudioManager class to set setSpeakerphoneOn(false), I then used the Voice_Call_Stream and add volume control to the Voice_Call_Stream.
The setSpeakerphoneOn(false) method is used in onCreate() of the activity and this appears to route to the headset, I then used a button and used the setSpeakerphoneOn(true) method and the audio gets routed to the speaker.
The method only appears to work when it is used in onCreate() for me and I haven't tested it extensively but for the moment it allows me to switch between headset and speaker on a 1.6 device

now in 2022 in Android S (12, API31) setSpeakerphoneOn(false) isn't working reliablie, it is in fact deprecated! instead we can use new API and setCommunicationDevice(AudioDeviceInfo) method. (works with AudioManager.STREAM_VOICE_CALL)
some sample
Boolean result = null;
ArrayList<Integer> targetTypes = new ArrayList<>();
if (earpieceMode) {
targetTypes.add(AudioDeviceInfo.TYPE_BUILTIN_EARPIECE);
} else { // play out loud
targetTypes.add(AudioDeviceInfo.TYPE_BUILTIN_SPEAKER);
}
AudioDeviceInfo currentDevice = audioManager.getCommunicationDevice();
if (currentDevice!=null) {
for (Integer targetType : targetTypes) {
if (targetType == currentDevice.getType()) {
Log.i("AUDIO_MANAGER", "setCommunicationDevice targetType ALREADY SET UP!!");
result = true;
break;
}
}
}
if (result == null) {
List<AudioDeviceInfo> devices = audioManager.getAvailableCommunicationDevices();
outer:
for (Integer targetType : targetTypes) {
for (AudioDeviceInfo device : devices) {
if (device.getType() == targetType) {
result = audioManager.setCommunicationDevice(device);
Log.i("AUDIO_MANAGER", "setCommunicationDevice type:" + targetType + " result:" + result);
if (result) break outer;
}
}
}
}
if (result == null) {
Log.i("AUDIO_MANAGER", "setCommunicationDevice targetType NOT FOUND!!");
}
else if (!result) {
Log.i("AUDIO_MANAGER", "setCommunicationDevice targetType FAILED TO SET!!");
}

If the ear piece is connected to the phone with bluetooth (which I assume it is), have you tried calling AudioManager.setBluetoothScoOn(true)?
I've been through the Android reference and this is the only thing I could find that you didn't mention trying.