Develop Reference

Android AudioRecord delay when using bytes, not when using and shorts

I'm developing an Xamarin Android application where video and audio are recorded with some playing music. I need to merge all these streams together.
I'm trying to figure out what I'm missing when recording audio using byte[] (or ByteBuffer which I also tried out) in the audioRecord.read() function. The output WAV file seems right (is clearly playable at a 44100Hz sample rate), but a delay appears after a couple of seconds and tends to get bigger and bigger.
When using shorts, I don't have any delay in the MIC recorded audio. The big issue using shorts is that no mather what I do, I can't have a sample rate higher then 8000hz (but this isn't the current issue although if someone knows how to fix it I'll take it :) )
The final merged file is an mp4 with AAC audio, merged using ffmpeg, but I don't think this is the issue.
Could it be related to 8000Hz (using short) and 44100Hz (using byte) ? Or I'm a adding something when using byte[] since I don't check how many bytes are read ?
Here are the parts involved in the issue:
//output file initialization
mDataOutputStream = new FileOutputStream(new Java.IO.File(mRawFilePath));
public void Run()
{
...
short[] shortBuf = new short[bufferSize / 2];
//byte[] byteBuf = new byte[bufferSize];
while(isRecording) {
//using shorts
audioRecorder.Read(shortBuf, 0, shortBuf.Length);
WriteShortsToFile(buf);
//using byte[]
//audioRecorder.Read(byteBuf, 0, byteBuf.Length);
//WriteBytesToFile(buf);
}
...
}
public void WriteShortsToFile(short[] shorts)
{
for (int i = 0; i < shorts.Length; i++)
{
mDataOutputStream.WriteByte(shorts[i] & 0xFF);
mDataOutputStream.WriteByte((shorts[i] >> 8) & 0xFF);
}
}
public void WriteBytesToFile(byte[] buf)
{
mDataOutputStream.Write(buf);
}

Finally got it working as it should.
I changed the allocation size of the short array to bufferSize.

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.

To mute microphone while recording audio

I am currently working on an app which is running in the background and muting the microphone during incoming and outgoing calls.But am unable to mute the microphone when there is any type audio recording.
It will be of great help if I am provided with any type of solutions.
Thanks in advance

I assume your talking about a Android app, but you should be more concise in your questions.
Regardless, this is a post about muting the microphone which should answer your question.
How does setMicrophoneMute() work?

AudioManager::setMicrophoneMute only applies to voice calls (and VoIP). It's possible that it will affect recordings as well on some products, but there's no guarantee that it will, so you can't rely on it.
It should still mute the voice call uplink so that the other party can't hear what you're saying even if there's a recording ongoing. If it doesn't I would consider that a bug in the implementation of the device you're testing this on. However, what you say will end up in the recording that you do locally (unless you're using the VOICE_DOWNLINK AudioSource).

If you don't want to record audio while video recording. You can set
AudioManager.setStreamMute(AudioManager.STREAM_MUSIC, true);
It also work by AudioManager.STREAM_SYSTEM for some devices.

There is no direct method to mute on AudioRecorder. We need to add some trick for mute the recording.
What I can do is, I download silence wav file and convert it to byte and add on the byte array.
When user click on Mute Button, isMuteClick = true and when unmute it will be false.
while (isStreaming)
{
if(!isMuteClick){
// read() is a blocking call // can set blocking see docs
int bytesRead = recorder.read(readBuffer, 0,bytesReadTotal);
bytesReadTotal += bytesRead; // above ...chunk - bytesReadTotal);
mainBuffer.write(readBuffer, 0, bytesRead);
}else{
int bytesRead = recorder.read(WavToByteArray(R.raw.silence), 0, chunk - bytesReadTotal); // 505 // 4410 //chunk - bytesReadTotal
bytesReadTotal += bytesRead; // above ...chunk - bytesReadTotal);
mainBuffer.write(WavToByteArray(R.raw.silence), 0, bytesRead);
}
} /
And here is code for converting silence wav file to byte array
private byte[] WavToByteArray(int resourceId) {
byte[] filteredByteArray = new byte[1024];
try {
InputStream inputStream = this.getResources().openRawResource(resourceId);
ByteArrayOutputStream outputStream = new ByteArrayOutputStream();
byte[] soundBytes = new byte[1024];
int i = Integer.MAX_VALUE;
while ((i = inputStream.read(soundBytes, 0, soundBytes.length)) > 0) {
outputStream.write(soundBytes, 0, i);
}
inputStream.close();
outputStream.close();
// remove .wav header
byte[] audioBytes = outputStream.toByteArray();
filteredByteArray = Arrays.copyOfRange(audioBytes, 44, audioBytes.length);
} catch (Exception e) {
e.printStackTrace();
}
return filteredByteArray;
}

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.

Using AudioTrack in Android to play a WAV file

I'm working with Android, trying to make my AudioTrack application play a Windows .wav file (Tada.wav). Frankly, it shouldn't be this hard, but I'm hearing a lot of strange stuff. The file is saved on my phone's mini SD card and reading the contents doesn't seem to be a problem, but when I play the file (with parameters I'm only PRETTY SURE are right), I get a few seconds of white noise before the sound seems to resolve itself into something that just may be right.
I have successfully recorded and played my own voice back on the phone -- I created a .pcm file according to the directions in this example:
http://emeadev.blogspot.com/2009/09/raw-audio-manipulation-in-android.html
(without the backwards masking)...
Anybody got some suggestions or awareness of an example on the web for playing a .wav file on an Android??
Thanks,
R.

I stumbled on the answer (frankly, by trying &^#! I didn't think would work), in case anybody's interested... In my original code (which is derived from the example in the link in the original post), the data is read from the file like so:
InputStream is = new FileInputStream (file);
BufferedInputStream bis = new BufferedInputStream (is, 8000);
DataInputStream dis = new DataInputStream (bis); // Create a DataInputStream to read the audio data from the saved file
int i = 0; // Read the file into the "music" array
while (dis.available() > 0)
{
music[i] = dis.readShort(); // This assignment does not reverse the order
i++;
}
dis.close(); // Close the input stream
In this version, music[] is array of SHORTS. So, the readShort() method would seem to make sense here, since the data is 16-bit PCM... However, on the Android that seems to be the problem. I changed that code to the following:
music=new byte[(int) file.length()];//size & length of the file
InputStream is = new FileInputStream (file);
BufferedInputStream bis = new BufferedInputStream (is, 8000);
DataInputStream dis = new DataInputStream (bis); // Create a DataInputStream to read the audio data from the saved file
int i = 0; // Read the file into the "music" array
while (dis.available() > 0)
{
music[i] = dis.readByte(); // This assignment does not reverse the order
i++;
}
dis.close(); // Close the input stream
In this version, music[] is an array of BYTES. I'm still telling the AudioTrack that it's 16-bit PCM data, and my Android doesn't seem to have a problem with writing an array of bytes into an AudioTrack thus configured... Anyway, it finally sounds right, so if anyone else wants to play Windows sounds on their Android, for some reason, that's the solution. Ah, Endianness......
R.

I found a lot of long answers to this question. My final solution, which given all the cutting and pasting is hardly mine, comes down to:
public boolean play() {
int i = 0;
byte[] music = null;
InputStream is = mContext.getResources().openRawResource(R.raw.noise);
at = new AudioTrack(AudioManager.STREAM_MUSIC, 44100,
AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
minBufferSize, AudioTrack.MODE_STREAM);
try{
music = new byte[512];
at.play();
while((i = is.read(music)) != -1)
at.write(music, 0, i);
} catch (IOException e) {
e.printStackTrace();
}
at.stop();
at.release();
return STOPPED;
}
STOPPED is just a "true" sent back as a signal to reset the pause/play button.
And in the class initializer:
public Mp3Track(Context context) {
mContext = context;
minBufferSize = AudioTrack.getMinBufferSize(44100,
AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT);
}
Context is just "this" from the calling activity.
You can use a FileInputStream on the sdcard, etc. My files are in res/raw

Are you skipping the first 44 bytes of the file before you dump the rest of the file's data into the buffer? The first 44 bytes are the WAVE header and they would sound like random noise if you tried to play them.
Also, are you sure you are creating the AudioTrack with the same properties as the WAVE you are trying to play (sample rate, bit rate, number of channels, etc)? Windows actually does a good job of giving this information to you in the File Properties page:

As said by Aaron C, you have to skip initial 44 bytes or (as I prefer) read first 44 bytes that are the WAVE header. In this way you know how many channels, bits per sample, length, etc... the WAVE contains.
Here you can find a good implementation of a WAVE header parser/writer.

Please don't perpetuate terrible parsing code. WAV parsing is trivial to implement
http://soundfile.sapp.org/doc/WaveFormat/
and you will thank yourself by being able to parse things such as the sampling rate, bit depth, and number of channels.
Also x86 and ARM (at least by default) are both little endian , so native-endian WAV files should be fine without any shuffling.

Just confirm if you have AudioTrack.MODE_STREAM and not AudioTrack.MODE_STATIC in the AudioTrack constructor:
AudioTrack at = new AudioTrack(
AudioManager.STREAM_MUSIC,
sampleRate,
AudioFormat.CHANNEL_IN_STEREO,
AudioFormat.ENCODING_PCM_16BIT,
// buffer length in bytes
outputBufferSize,
AudioTrack.MODE_STREAM
);

Sample wav file:
http://www.mauvecloud.net/sounds/pcm1644m.wav
Sample Code:
public class AudioTrackPlayer {
Context mContext;
int minBufferSize;
AudioTrack at;
boolean STOPPED;
public AudioTrackPlayer(Context context) {
Log.d("------","init");
mContext = context;
minBufferSize = AudioTrack.getMinBufferSize(44100,
AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT);
}
public boolean play() {
Log.d("------","play");
int i = 0;
byte[] music = null;
InputStream is = mContext.getResources().openRawResource(R.raw.pcm1644m);
at = new AudioTrack(AudioManager.STREAM_MUSIC, 44100,
AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT,
minBufferSize, AudioTrack.MODE_STREAM);
try {
music = new byte[512];
at.play();
while ((i = is.read(music)) != -1)
at.write(music, 0, i);
} catch (IOException e) {
e.printStackTrace();
}
at.stop();
at.release();
return STOPPED;
}
}