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I am currently developing an Android Application that has audio recording and playing. I am new to dealing with audio and I'm having some trouble with encoding and formats.
I am able to record and play the audio in my application, but when exporting I am not able to reproduce the audio. The only way I found was exporting my .pcm file and converting using Audacity.
This is my code to record the audio is:
private Thread recordingThread
private AudioRecord mRecorder;
private boolean isRecording = false;
private void startRecording() {
mRecorder = new AudioRecord(MediaRecorder.AudioSource.MIC,
Constants.RECORDER_SAMPLERATE, Constants.RECORDER_CHANNELS,
Constants.RECORDER_AUDIO_ENCODING, Constants.BufferElements2Rec * Constants.BytesPerElement);
mRecorder.startRecording();
isRecording = true;
recordingThread = new Thread(new Runnable() {
public void run() {
writeAudioDataToFile();
}
}, "AudioRecorder Thread");
recordingThread.start();
}
private void writeAudioDataToFile() {
// Write the output audio in byte
FileOutputStream os = null;
try {
os = new FileOutputStream(mFileName);
} catch (FileNotFoundException e) {
e.printStackTrace();
}
while (isRecording) {
// gets the voice output from microphone to byte format
mRecorder.read(sData, 0, Constants.BufferElements2Rec);
try {
// // writes the data to file from buffer
// // stores the voice buffer
byte bData[] = short2byte(sData);
os.write(bData, 0, Constants.BufferElements2Rec * Constants.BytesPerElement);
} catch (IOException e) {
e.printStackTrace();
}
}
try {
os.close();
} catch (IOException e) {
e.printStackTrace();
}
}
To play the recorded audio, the code is:
private void startPlaying() {
new Thread(new Runnable() {
public void run() {
try {
File file = new File(mFileName);
byte[] audioData = null;
InputStream inputStream = new FileInputStream(mFileName);
audioData = new byte[Constants.BufferElements2Rec];
mPlayer = new AudioTrack(AudioManager.STREAM_MUSIC, Constants.RECORDER_SAMPLERATE,
AudioFormat.CHANNEL_OUT_MONO, Constants.RECORDER_AUDIO_ENCODING,
Constants.BufferElements2Rec * Constants.BytesPerElement, AudioTrack.MODE_STREAM);
final float duration = (float) file.length() / Constants.RECORDER_SAMPLERATE / 2;
Log.i(TAG, "PLAYBACK AUDIO");
Log.i(TAG, String.valueOf(duration));
mPlayer.setPositionNotificationPeriod(Constants.RECORDER_SAMPLERATE / 10);
mPlayer.setNotificationMarkerPosition(Math.round(duration * Constants.RECORDER_SAMPLERATE));
mPlayer.play();
int i = 0;
while ((i = inputStream.read(audioData)) != -1) {
try {
mPlayer.write(audioData, 0, i);
} catch (Exception e) {
Log.e(TAG, "Exception: " + e.getLocalizedMessage());
}
}
} catch (FileNotFoundException fe) {
Log.e(TAG, "File not found: " + fe.getLocalizedMessage());
} catch (IOException io) {
Log.e(TAG, "IO Exception: " + io.getLocalizedMessage());
}
}
}).start();
}
The constants defined in a Constants class are:
public class Constants {
final static public int RECORDER_SAMPLERATE = 44100;
final static public int RECORDER_CHANNELS = AudioFormat.CHANNEL_IN_MONO;
final static public int RECORDER_AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT;
final static public int BufferElements2Rec = 1024; // want to play 2048 (2K) since 2 bytes we use only 1024
final static public int BytesPerElement = 2; // 2 bytes in 16bit format
}
If I export the file as it is, I convert it with Audacity and it plays. I do, however, need to export it in a format that can be played automatically.
I've seen answers to implement Lame and am currently working on it. I've also found an answer to convert it using:
private File rawToWave(final File rawFile, final String filePath) throws IOException {
File waveFile = new File(filePath);
byte[] rawData = new byte[(int) rawFile.length()];
DataInputStream input = null;
try {
input = new DataInputStream(new FileInputStream(rawFile));
input.read(rawData);
} finally {
if (input != null) {
input.close();
}
}
DataOutputStream output = null;
try {
output = new DataOutputStream(new FileOutputStream(waveFile));
// WAVE header
// see http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
writeString(output, "RIFF"); // chunk id
writeInt(output, 36 + rawData.length); // chunk size
writeString(output, "WAVE"); // format
writeString(output, "fmt "); // subchunk 1 id
writeInt(output, 16); // subchunk 1 size
writeShort(output, (short) 1); // audio format (1 = PCM)
writeShort(output, (short) 1); // number of channels
writeInt(output, Constants.RECORDER_SAMPLERATE); // sample rate
writeInt(output, Constants.RECORDER_SAMPLERATE * 2); // byte rate
writeShort(output, (short) 2); // block align
writeShort(output, (short) 16); // bits per sample
writeString(output, "data"); // subchunk 2 id
writeInt(output, rawData.length); // subchunk 2 size
// Audio data (conversion big endian -> little endian)
short[] shorts = new short[rawData.length / 2];
ByteBuffer.wrap(rawData).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shorts);
ByteBuffer bytes = ByteBuffer.allocate(shorts.length * 2);
for (short s : shorts) {
bytes.putShort(s);
}
output.write(bytes.array());
} finally {
if (output != null) {
output.close();
}
}
return waveFile;
}
private void writeInt(final DataOutputStream output, final int value) throws IOException {
output.write(value >> 0);
output.write(value >> 8);
output.write(value >> 16);
output.write(value >> 24);
}
private void writeShort(final DataOutputStream output, final short value) throws IOException {
output.write(value >> 0);
output.write(value >> 8);
}
private void writeString(final DataOutputStream output, final String value) throws IOException {
for (int i = 0; i < value.length(); i++) {
output.write(value.charAt(i));
}
}
But this, when exported, plays with the correct duration but just white noise.
Some of the answers that I've tried but wasn't able to work:
Anyone can point out what is the best solution? Is it really implementing lame or can it be done on a more straight forward way? If so, why is the code sample converting the file to just white noise?
657
– Go to file open to find the audio file you want to convert. – Once opened, go to file, export audio. – Use drop down menu to select save as type; two of the options are the WAV 32-bit float PCM and MP3.
You can easily use Media Encoder to convert MP3s to WAV file format for all your sound engineering and audio editing needs.
You've got most of the code correct. The only issue that I can see is the part where you write the PCM data to the WAV file. This should be quite simple to do because WAV = Metadata + PCM (in that order). This should work:
private void rawToWave(final File rawFile, final File waveFile) throws IOException {
byte[] rawData = new byte[(int) rawFile.length()];
DataInputStream input = null;
try {
input = new DataInputStream(new FileInputStream(rawFile));
input.read(rawData);
} finally {
if (input != null) {
input.close();
}
}
DataOutputStream output = null;
try {
output = new DataOutputStream(new FileOutputStream(waveFile));
// WAVE header
// see http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
writeString(output, "RIFF"); // chunk id
writeInt(output, 36 + rawData.length); // chunk size
writeString(output, "WAVE"); // format
writeString(output, "fmt "); // subchunk 1 id
writeInt(output, 16); // subchunk 1 size
writeShort(output, (short) 1); // audio format (1 = PCM)
writeShort(output, (short) 1); // number of channels
writeInt(output, 44100); // sample rate
writeInt(output, RECORDER_SAMPLERATE * 2); // byte rate
writeShort(output, (short) 2); // block align
writeShort(output, (short) 16); // bits per sample
writeString(output, "data"); // subchunk 2 id
writeInt(output, rawData.length); // subchunk 2 size
// Audio data (conversion big endian -> little endian)
short[] shorts = new short[rawData.length / 2];
ByteBuffer.wrap(rawData).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(shorts);
ByteBuffer bytes = ByteBuffer.allocate(shorts.length * 2);
for (short s : shorts) {
bytes.putShort(s);
}
output.write(fullyReadFileToBytes(rawFile));
} finally {
if (output != null) {
output.close();
}
}
}
byte[] fullyReadFileToBytes(File f) throws IOException {
int size = (int) f.length();
byte bytes[] = new byte[size];
byte tmpBuff[] = new byte[size];
FileInputStream fis= new FileInputStream(f);
try {
int read = fis.read(bytes, 0, size);
if (read < size) {
int remain = size - read;
while (remain > 0) {
read = fis.read(tmpBuff, 0, remain);
System.arraycopy(tmpBuff, 0, bytes, size - remain, read);
remain -= read;
}
}
} catch (IOException e){
throw e;
} finally {
fis.close();
}
return bytes;
}
private void writeInt(final DataOutputStream output, final int value) throws IOException {
output.write(value >> 0);
output.write(value >> 8);
output.write(value >> 16);
output.write(value >> 24);
}
private void writeShort(final DataOutputStream output, final short value) throws IOException {
output.write(value >> 0);
output.write(value >> 8);
}
private void writeString(final DataOutputStream output, final String value) throws IOException {
for (int i = 0; i < value.length(); i++) {
output.write(value.charAt(i));
}
}
How to use
It's quite simple to use. Just call it like this:
File f1 = new File("/sdcard/44100Sampling-16bit-mono-mic.pcm"); // The location of your PCM file
File f2 = new File("/sdcard/44100Sampling-16bit-mono-mic.wav"); // The location where you want your WAV file
try {
rawToWave(f1, f2);
} catch (IOException e) {
e.printStackTrace();
}
How all this works
As you can see, the WAV header is the only difference between WAV and PCM file formats. The assumption is that you are recording 16 bit PCM MONO audio (which according to your code, you are). The rawToWave function just neatly adds headers to the WAV file, so that music players know what to expect when your file is opened, and then after the headers, it just writes the PCM data from the last bit onwards.
Cool Tip
If you want to shift the pitch of your voice, or make a voice changer app, all you got to do is increase/decrease the value of writeInt(output, 44100); // sample rate in your code. Decreasing it will tell the player to play it at a different rate thereby changing the output pitch. Just a little extra 'good to know' thing. :)
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