Develop Reference

how to control the sound volume of my own android app

I imagine the answer to this question must be something quite typical. The point is that I have some buttons that play sounds when you click on them. I want to control the sound volume. How do I control the sound volume? I am using the following code:
private fun playSound() {
val sound = getSystemService(Context.AUDIO_SERVICE) as AudioManager
sound.playSoundEffect(AudioManager.FX_KEY_CLICK, 1f)
}

https://developer.android.com/reference/android/media/AudioManager#playSoundEffect(int,%20float)
volume float: Sound effect volume. The volume value is a raw scalar so UI controls should be scaled logarithmically. If a volume of -1 is specified, the AudioManager.STREAM_MUSIC stream volume minus 3dB will be used. NOTE: This version is for applications that have their own settings panel for enabling and controlling volume.
So you need to adjust that second parameter. I haven't used it, but it says that's a scalar so I'd imagine 1 is full volume? It doesn't mention any other value as a max constant or anything. -1 gives you a sound a little quieter than the current media volume setting, if that's convenient. You'll need to play around with different values between 0 and 1 (I assume!) and see what you need

audio latency issues

In the application which I want to create, I face some technical obstacles. I have two music tracks in the application. For example, a user imports the music background as a first track. The second path is a voice recorded by the user to the rhythm of the first track played by the speaker device (or headphones). At this moment we face latency. After recording and playing back in the app, the user hears the loss of synchronisation between tracks, which occurs because of the microphone and speaker latencies.
Firstly, I try to detect the delay by filtering the input sound. I use android’s AudioRecord class, and the method read(). This method fills my short array with audio data.
I found that the initial values of this array are zeros so I decided to cut them out before I will start to write them into the output stream.
So I consider those zeros as a „warmup” latency of the microphone. Is this approach correct? This operation gives some results, but it doesn’t resolve the problem, and at this stage, I’m far away from that.
But the worse case is with the delay between starting the speakers and playing the music. This delay I cannot filter or detect. I tried to create some calibration feature which counts the delay. I play a „beep” sound through the speakers, and when I start to play it, I also begin to measure time. Then, I start recording and listen for this sound being detected by the microphone. When I recognise this sound in the app, I stop measuring time. I repeat this process several times, and the final value is the average from those results. That is how I try to measure the latency of the device. Now, when I have this value, I can simply shift the second track backwards to achieve synchronisation of both records (I will lose some initial milliseconds of the recording, but I skip this case, for now, there are some possibilities to fix it).
I thought that this approach would resolve the problem, but it turned out this is not as simple as I thought. I found two issues here:
1. Delay while playing two tracks simultaneously
2. Random in device audio latency.
The first: I play two tracks using AudioTrack class and I run method play() like this:
val firstTrack = //creating a track
val secondTrack = //creating a track
firstTrack.play()
secondTrack.play()
This code causes delays at the stage of playing tracks. Now, I don’t even have to think about latency while recording; I cannot play two tracks simultaneously without delays. I tested this with some external audio file (not recorded in my app) - I’m starting the same audio file using the code above, and I can see a delay. I also tried it with MediaPlayer class, and I have the same results. In this case, I even try to play tracks when callback OnPreparedListener invoke:
val firstTrack = //AudioPlayer
val secondTrack = //AudioPlayer
second.setOnPreparedListener {
first.start()
second.start()
}
And it doesn’t help.
I know that there is one more class provided by Android called SoundPool. According to the documentation, it can be better with playing tracks simultaneously, but I can’t use it because it supports only small audio files and that can't limit me.
How can I resolve this problem? How can I start playing two tracks precisely at the same time?
The second: Audio latency is not deterministic - sometimes it is smaller, and sometimes it’s huge, and it’s out of my hands. So measuring device latency can help but again - it cannot resolve the problem.
To sum up: is there any solution, which can give me exact latency per device (or app session?) or other triggers which detect actual delay, to provide the best synchronisation while playback two tracks at the same time?
Thank you in advance!

Synchronising audio for karaoke apps is tough. The main issue you seem to be facing is variable latency in the output stream.
This is almost certainly caused by "warm up" latency: the time it takes from hitting "play" on your backing track to the first frame of audio data being rendered by the audio device (e.g. headphones). This can have large variance and is difficult to measure.
The first (and easiest) thing to try is to use MODE_STREAM when constructing your AudioTrack and prime it with bufferSizeInBytes of data prior to calling play (more here). This should result in lower, more consistent "warm up" latency.
A better way is to use the Android NDK to have a continuously running audio stream which is just outputting silence until the moment you hit play, then start sending audio frames immediately. The only latency you have here is the continuous output latency.
If you decide to go down this route I recommend taking a look at the Oboe library (full disclosure: I am one of the authors).
To answer one of your specific questions...
Is there a way to calculate the latency of the audio output stream programatically?
Yes. The easiest way to explain this is with a code sample (this is C++ for the AAudio API but the principle is the same using Java AudioTrack):
// Get the index and time that a known audio frame was presented for playing
int64_t existingFrameIndex;
int64_t existingFramePresentationTime;
AAudioStream_getTimestamp(stream, CLOCK_MONOTONIC, &existingFrameIndex, &existingFramePresentationTime);
// Get the write index for the next audio frame
int64_t writeIndex = AAudioStream_getFramesWritten(stream);
// Calculate the number of frames between our known frame and the write index
int64_t frameIndexDelta = writeIndex - existingFrameIndex;
// Calculate the time which the next frame will be presented
int64_t frameTimeDelta = (frameIndexDelta * NANOS_PER_SECOND) / sampleRate_;
int64_t nextFramePresentationTime = existingFramePresentationTime + frameTimeDelta;
// Assume that the next frame will be written into the stream at the current time
int64_t nextFrameWriteTime = get_time_nanoseconds(CLOCK_MONOTONIC);
// Calculate the latency
*latencyMillis = (double) (nextFramePresentationTime - nextFrameWriteTime) / NANOS_PER_MILLISECOND;
A caveat: This method relies on accurate timestamps being reported by the audio hardware. I know this works on Google Pixel devices but have heard reports that it isn't so accurate on other devices so YMMV.

Following the answer of donturner, here's a Java version (that also uses other methods depending on the SDK version)
/** The audio latency has not been estimated yet */
private static long AUDIO_LATENCY_NOT_ESTIMATED = Long.MIN_VALUE+1;
/** The audio latency default value if we cannot estimate it */
private static long DEFAULT_AUDIO_LATENCY = 100L * 1000L * 1000L; // 100ms
/**
* Estimate the audio latency
*
* Not accurate at all, depends on SDK version, etc. But that's the best
* we can do.
*/
private static void estimateAudioLatency(AudioTrack track, long audioFramesWritten) {
long estimatedAudioLatency = AUDIO_LATENCY_NOT_ESTIMATED;
// First method. SDK >= 19.
if (Build.VERSION.SDK_INT >= 19 && track != null) {
AudioTimestamp audioTimestamp = new AudioTimestamp();
if (track.getTimestamp(audioTimestamp)) {
// Calculate the number of frames between our known frame and the write index
long frameIndexDelta = audioFramesWritten - audioTimestamp.framePosition;
// Calculate the time which the next frame will be presented
long frameTimeDelta = _framesToNanoSeconds(frameIndexDelta);
long nextFramePresentationTime = audioTimestamp.nanoTime + frameTimeDelta;
// Assume that the next frame will be written at the current time
long nextFrameWriteTime = System.nanoTime();
// Calculate the latency
estimatedAudioLatency = nextFramePresentationTime - nextFrameWriteTime;
}
}
// Second method. SDK >= 18.
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED && Build.VERSION.SDK_INT >= 18) {
Method getLatencyMethod;
try {
getLatencyMethod = AudioTrack.class.getMethod("getLatency", (Class<?>[]) null);
estimatedAudioLatency = (Integer) getLatencyMethod.invoke(track, (Object[]) null) * 1000000L;
} catch (Exception ignored) {}
}
// If no method has successfully gave us a value, let's try a third method
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED) {
AudioManager audioManager = (AudioManager) CRT.getInstance().getSystemService(Context.AUDIO_SERVICE);
try {
Method getOutputLatencyMethod = audioManager.getClass().getMethod("getOutputLatency", int.class);
estimatedAudioLatency = (Integer) getOutputLatencyMethod.invoke(audioManager, AudioManager.STREAM_MUSIC) * 1000000L;
} catch (Exception ignored) {}
}
// No method gave us a value. Let's use a default value. Better than nothing.
if (estimatedAudioLatency == AUDIO_LATENCY_NOT_ESTIMATED) {
estimatedAudioLatency = DEFAULT_AUDIO_LATENCY;
}
return estimatedAudioLatency
}
private static long _framesToNanoSeconds(long frames) {
return frames * 1000000000L / SAMPLE_RATE;
}

The android MediaPlayer class is notoriously slow to begin audio playback, I experienced an issue in an app I was creating where there was a greater than one second delay to begin playing an audio clip. I resolved it by switching to ExoPlayer which resulted in the playback starting within 100ms. I've also read that ffmpeg has even faster start audio startup time than ExoPlayer but I haven't used it so I can't make any promises.

Increase MediaPlayer volume beyond 100%

Below code is working but not increasing the media player volume higher than the default max volume.Please help
AudioManager am =
(AudioManager) getSystemService(Context.AUDIO_SERVICE);
am.setStreamVolume(
AudioManager.STREAM_MUSIC,
am.getStreamMaxVolume(AudioManager.STREAM_MUSIC),
0);

The MediaPlayer class's setVolume() method only accepts scalars in the range [0.0, 1.0], but the classes deriving from AudioEffect can be used to amplify the MediaPlayer's audio session.
For example, LoudnessEnhancer amplifies samples by a gain specified in millibels (i.e. hundredths of decibels):
MediaPlayer player = new MediaPlayer();
player.setDataSource("https://www.example.org/song.mp3");
player.prepare();
// Increase amplitude by 20%.
double audioPct = 1.2;
int gainmB = (int) Math.round(Math.log10(audioPct) * 2000);
LoudnessEnhancer enhancer = new LoudnessEnhancer(player.getAudioSessionId());
enhancer.setTargetGain(gainmB);
It's unclear from the documentation, but it appeared to me that LoudnessEnhancer doesn't work properly with negative gains, so you may still need to use MediaPlayer's setVolume() method if you want to decrease the volume.
DynamicsProcessing provides multiple stages across multiple channels, including an input gain stage.

For increasing the volume of the device beyond the system volume u have to go in engineers mode.for that save below code.and paste it in number entering box In calling option it will directly redirect you to the engineers mode
*#*#3646633#*#*
By this you can access the system settings one thing make sure that don't use this without care it may affect your system performance.

How avoid automatic gain control with AudioRecord?

How can I do audio recordings using android.media.AudioRecord without any smartphone-manufacturer-dependent fancy signal processing like automatic gain control (AGC) and/or equalization, noise suppression, echo cancellation, ... just the pure microphone signal?
Background
MediaRecorder.AudioSource provides nine constants,
DEFAULT and MIC initially being there,
VOICE_UPLINK, VOICE_DOWNLINK, and VOICE_CALL added in API level 4,
CAMCORDER and VOICE_RECOGNITION added in API 7,
VOICE_COMMUNICATION added in API 11,
REMOTE_SUBMIX added in API 19 but not available to third-party applications.
But none of them does a clean job across all smartphones. Rather, I have to find out myself it seems, which device uses which combinations of signal processing blocks for which MediaRecorder.AudioSource constant.
Would be nice to have a tenth constant like PURE_MIC added in API level 20.
But as long as this is not available, what can I do instead?

Short answer is "Nothing".
The AudioSources correspond to various logical audio input devices depending on the accessories that you have connected to the phone and the current use-case, which in turn corresponds to physical devices (primary built-in mic, secondary mic, wired headset mic, etc) with different tunings.
Each such combination of physical device and tuning is trimmed by the OEM to meet both external requirements (e.g. CTS, operator requirements, etc) and internal acoustic requirements set by the OEM itself. This process may cause the introduction of various filters - such as AGC, noise suppression, equalization, etc - into the audio input path at the hardware codec or multimedia DSP level.
While a PURE_MIC source might be useful in for some applications, it's not something that's available today.
On many devices you can control things like microphone gain, and possibly even the filter chain, by using amixer to write to the hardware codec's ALSA controls. However, this would obviously be a very platform-specific approach, and I also suspect that you have to be running as either the root or audio user to be allowed to do this.

Some devices add AGC effect to the sound input tract by default. Therefore, you need to obtain reference to corresponding AudioEffect object and force it to disable.
First, obtain AutomaticGainControl object linked to the AudioRecord audio session, and then just set it disabled:
if (AutomaticGainControl.isAvailable()) {
AutomaticGainControl agc = AutomaticGainControl.create(
myAudioRecord.getAudioSessionId()
);
agc.setEnabled(false);
}

Note: Most of the audio sources (including DEFAULT) apply processing to the audio signal. To record raw audio select UNPROCESSED. Some devices do not support unprocessed input. Call AudioManager.getProperty("PROPERTY_SUPPORT_AUDIO_SOURCE_UNPROCESSED") first to verify it's available. If it is not, try using VOICE_RECOGNITION instead, which does not employ AGC or noise suppression. You can use UNPROCESSED as an audio source even when the property is not supported, but there is no guarantee whether the signal will be unprocessed or not in that case.
Android documentation Link https://developer.android.com/guide/topics/media/mediarecorder.html#example
AudioManager audioManager = (AudioManager)getSystemService(Context.AUDIO_SERVICE);
if(audioManager.getProperty(AudioManager.PROPERTY_SUPPORT_AUDIO_SOURCE_UNPROCESSED) !=null)
mRecorder.setAudioSource(MediaRecorder.AudioSource.UNPROCESSED);
else
mRecorder.setAudioSource(MediaRecorder.AudioSource.VOICE_RECOGNITION);

MIC should be fine, and for the rest you need to know if they are supported.
I've made a class for this:
enum class AudioSource(val audioSourceValue: Int, val minApi: Int) {
VOICE_CALL(MediaRecorder.AudioSource.VOICE_CALL, 4), DEFAULT(MediaRecorder.AudioSource.DEFAULT, 1), MIC(MediaRecorder.AudioSource.MIC, 1),
VOICE_COMMUNICATION(MediaRecorder.AudioSource.VOICE_COMMUNICATION, 11), CAMCORDER(MediaRecorder.AudioSource.CAMCORDER, 7),
VOICE_RECOGNITION(MediaRecorder.AudioSource.VOICE_RECOGNITION, 7),
VOICE_UPLINK(MediaRecorder.AudioSource.VOICE_UPLINK, 4), VOICE_DOWNLINK(MediaRecorder.AudioSource.VOICE_DOWNLINK, 4),
#TargetApi(Build.VERSION_CODES.KITKAT)
REMOTE_SUBMIX(MediaRecorder.AudioSource.REMOTE_SUBMIX, 19),
#TargetApi(Build.VERSION_CODES.N)
UNPROCESSED(MediaRecorder.AudioSource.UNPROCESSED, 24);
fun isSupported(context: Context): Boolean =
when {
Build.VERSION.SDK_INT < minApi -> false
this != UNPROCESSED -> true
else -> {
val audioManager: AudioManager = context.getSystemService(Context.AUDIO_SERVICE) as AudioManager
Build.VERSION.SDK_INT >= Build.VERSION_CODES.N && "true" == audioManager.getProperty(AudioManager.PROPERTY_SUPPORT_AUDIO_SOURCE_UNPROCESSED)
}
}
companion object {
fun getAllSupportedValues(context: Context): ArrayList<AudioSource> {
val values = AudioSource.values()
val result = ArrayList<AudioSource>(values.size)
for (value in values)
if (value.isSupported(context))
result.add(value)
return result
}
}
}

Set the volume of an android application

In order for the user to control the volume , my android application has a menu consisting of a slider that provides int values from 0 to 10 , when dragged. After I obtain a value , I must set the volume to the corresponding value chosen by the user , and well , this is the part that I don't know to implement and I 'd like to find about it.

Use the AudioManager class. Essentially the code goes as follows:
AudioManager audioManager = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
audioManager.setStreamVolume(streamType, volume, flags);
The problem is that the volume of the device isn't necessarily mapped from 0 to 10 as you have in your slider. On my emulator, it's from 0 to 7. So what you need to do is getStreamMaxVolume(...) to know what your max is, and then work out your value as a fraction of that. As an example, if your user chooses volume 8 out of 10, that's equivalent to 0.8 * 7 = 5.6, which you should round to 6 out of 7.
The "stream" refers to things like ringer volume, notification volume, music volume, etc. If you want to change the volume of the ringer, you need to make sure all your commands have AudioManager.STREAM_RING as the streamType.