I am Developing an app in android live streaming. I am able to stream the live videos to the YouTube channel. But the problem was, getting no audio to that live streaming video.
My code will like below
private static final int frequency= 44100;
public void recordThread() {
Log.d(MainActivity.APP_NAME, "recordThread");
int audioEncoding = AudioFormat.ENCODING_PCM_16BIT;
int channelConfiguration = AudioFormat.CHANNEL_IN_STEREO;
int bufferSize = AudioRecord.getMinBufferSize(frequency, channelConfiguration, audioEncoding);
Log.i(MainActivity.APP_NAME, "AudioRecord buffer size: " + bufferSize);
// 16 bit PCM stereo recording was chosen as example.
AudioRecord recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, frequency, channelConfiguration,
audioEncoding, bufferSize);
recorder.startRecording();
// Make bufferSize be in samples instead of bytes.
bufferSize /= 2;
short[] buffer = new short[bufferSize];
while (!cancel) {
int bufferReadResult = recorder.read(buffer, 0, bufferSize);
// Utils.Debug("bufferReadResult: " + bufferReadResult);
if (bufferReadResult > 0) {
frameCallback.handleFrame(buffer, bufferReadResult);
} else if (bufferReadResult < 0) {
Log.w(MainActivity.APP_NAME, "Error calling recorder.read: " + bufferReadResult);
}
}
recorder.stop();
Log.d(MainActivity.APP_NAME, "exit recordThread");
}
please some suggest me the solution to get out of this issue.
Search in github. There you will find a sample project(yasea) for audio encoding. Intermix those two projects, you will get the solution.
Related
I am trying to send audio between windows and android, I was successfully able to do that windows to windows but when I stream audio from android, it produces a white noise only. I think it is an issue with the AudioFormat in android and Windows because when I changed the sample Bits to 8 I guess, I heard the voice in one side of my headphones but then it went away too.
On Android Side
int BUFFER_MS = 15; // do not buffer more than BUFFER_MS milliseconds
int bufferSize = 48000 * 2 * BUFFER_MS / 1000;
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, 48000, 2,
AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
byte[] buffer = new byte[bufferSize];
int bytesRead;
audioTrack.play();
while (socket.isConnected()) {
bytesRead = inputStream.read(buffer, 0, buffer.length);
audioTrack.write(buffer,0,bytesRead);
}
On Windows Side
AudioFormat format = getAudioFormat();
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
// checks if system supports the data line
if (!AudioSystem.isLineSupported(info)) {
throw new LineUnavailableException(
"The system does not support the specified format.");
}
TargetDataLine audioLine = AudioSystem.getTargetDataLine(format);
audioLine.open(format);
audioLine.start();
byte[] buffer = new byte[BUFFER_SIZE];
int bytesRead;
while (socket.isConnected()) {
bytesRead = audioLine.read(buffer, 0, buffer.length);
outputStream.write(buffer,0,bytesRead);
}
and getAudioFormat function is
AudioFormat getAudioFormat() {
float sampleRate = 48000;
int sampleSizeInBits = 16;
int channels = 2;
boolean signed = true;
boolean bigEndian = true;
return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed,
bigEndian);
}
Only hearing a white noise, if someone can help please do.
Okayyyy So I found out the problem. I just had to put bigEndian to false -_-
It's the byte order difference. I don't understand why it's different in android and pc but seems like it does the trick.
I am trying to play a wav file of size 230mb and 20 min whose properties are as below:
ffmpeg -i 1.wav
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
I am learning how to use audiotrack.
I found two solutions to play this audio play using audiotrack.
Solution 1: the following plays the audio
int frequency = 44100;
int channelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
int audioEncoding = AudioFormat.ENCODING_PCM_16BIT;
int bufferSize = AudioTrack.getMinBufferSize(frequency, channelConfiguration,audioEncoding);
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, frequency,
channelConfiguration, audioEncoding, bufferSize,
AudioTrack.MODE_STREAM);
int count = 0;
byte[] data = new byte[bufferSize];
try{
FileInputStream fileInputStream = new FileInputStream(listMusicFiles.get(0).listmusicfiles_fullfilepath);
DataInputStream dataInputStream = new DataInputStream(fileInputStream);
audioTrack.play();
while((count = dataInputStream.read(data, 0, bufferSize)) > -1){
audioTrack.write(data, 0, count);
}
audioTrack.stop();
audioTrack.release();
dataInputStream.close();
fileInputStream.close();
}
catch (FileNotFoundException e){
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
Second Solution: Playing noise
int frequency = 44100;
int channelConfiguration = AudioFormat.CHANNEL_CONFIGURATION_STEREO;
int audioEncoding = AudioFormat.ENCODING_PCM_16BIT;
int bufferSize = AudioTrack.getMinBufferSize(frequency, channelConfiguration,audioEncoding);
short[] audiodata = new short[bufferSize];
try {
DataInputStream dis = new DataInputStream(
new BufferedInputStream(new FileInputStream(
listMusicFiles.get(0).listmusicfiles_fullfilepath)));
AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, frequency,
channelConfiguration, audioEncoding, bufferSize,
AudioTrack.MODE_STREAM);
audioTrack.play();
while (dis.available() > 0) {
int i = 0;
while (dis.available() > 0 && i < audiodata.length) {
audiodata[i] = dis.readShort();
i++;
}
audioTrack.write(audiodata, 0, audiodata.length);
}
dis.close();
} catch (Throwable t) {
Log.e("AudioTrack", "Playback Failed");
}
I am new to short sample and byte samples. I tried to understand but it not so easy.
I could understand the first solution is using byte sample and the second solution is using short samples.
So why is the second solution not working.
The default size of a short type is two bytes. You might have a look at this documentation as well.
The audio track has a recommended buffer size and sample rate which can be found using the way suggested here in this answer. Please have a look.
However, it is important to play the audio using the recommended sample rate which is 44100 Hz in your case and the recommended buffer size that you get using the following code segment.
AudioTrack.getMinBufferSize(frequency, channelConfiguration, audioEncoding)
In your implementation with short array, the buffer size is doubled and hence its creating noises in case of playing the audio. I would suggest, you might consider changing the buffer size by dividing the size by two in your implementation using short.
short[] audiodata = new short[(int) bufferSize / 2];
Hope you have understood the problem.
Im trying to record speech using AudioRecord object.
It works great on all devices except Galaxy S6 edge.
Here the code
AudioRecord recorder = null;
int desiredRate = 0;
for (int rate : new int[] {44100, 8000, 11025, 16000, 22050}) {
int bufferSize = AudioRecord.getMinBufferSize(rate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT);
if (bufferSize > 0) {
// buffer size is valid, Sample rate supported
recorder = new AudioRecord(MediaRecorder.AudioSource.MIC, rate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize);
if (recorder.getState() != AudioRecord.STATE_INITIALIZED) {
recorder.release();
} else {
break;
}
}
}
In the Galaxy the state is 0.
What can be the reason for it?
I am using AudioRecord to capture audio packets and stream them to a voice recognition server.
In my Galaxy Note 4, Android M device, it works perfectly fine.
However, when I use other devices (Nexus 7/Android L and HTC combo/android ICS) the resulting audio is choppy, with glitchy noises in the sound every half a second that spoil the speech recognition process at the server.
I know this is a complicated topic, does somebody know how to deal with this audio capture irregularities in android?
This is my code setup:
private static final int AUDIO_SOURCE = MediaRecorder.AudioSource.VOICE_RECOGNITION;
private static final int SAMPLING_RATE = 16000; //44100,
private static final int CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_MONO;
private static final int BIT_RATE = AudioFormat.ENCODING_PCM_16BIT;
bufferSize = AudioRecord.getMinBufferSize(SAMPLING_RATE,
CHANNEL_CONFIG,
BIT_RATE;
audioBuffer = new byte[bufferSize];
AudioRecord recorder = new AudioRecord(AUDIO_SOURCE,
SAMPLING_RATE,
CHANNEL_CONFIG,
BIT_RATE,
bufferSize);
recorder.startRecording();
while (isRecording) {
short[] buffer = new short[bufferSize];
int shorts_recorded = recorder.read(buffer, 0, buffer.length);
byte[] audioBytes = new byte[bufferSize*2]; //bufferSize*2
ByteBuffer.wrap(audioBytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(buffer);//runningBuffer.get(0));
Integer.toString(audioBytes.length)+","+audioBuffer.length);
handler.onAudioDataCapture(audioBytes); //expose audio data to upload callback interface
proceed();
}
we need recording should be done in a manner that records and overwrites the exiting recording so it will always provide the last X minutes as defined in the app settings.
int bufferSize = 2 * AudioRecord.getMinBufferSize(RECORDER_SAMPLERATE,
RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING);
recorder = new AudioRecord(MediaRecorder.AudioSource.MIC,
RECORDER_SAMPLERATE, RECORDER_CHANNELS,
RECORDER_AUDIO_ENCODING, bufferSize);
recorder.startRecording();
int b = recorder.getState();
if (b == 1)
recorder.startRecording();