Develop Reference

Android oboe c++ Some sounds distorted on playback

I'm using the Android oboe library for high performance audio in a music game.
In the assets folder I have 2 .raw files (both 48000Hz 16 bit PCM wavs and about 60kB)
std_kit_sn.raw
std_kit_ht.raw
These are loaded into memory as SoundRecordings and added to a Mixer. kSampleRateHz is 48000:
stdSN= SoundRecording::loadFromAssets(mAssetManager, "std_kit_sn.raw");
stdHT= SoundRecording::loadFromAssets(mAssetManager, "std_kit_ht.raw");
mMixer.addTrack(stdSN);
mMixer.addTrack(stdFT);
// Create a builder
AudioStreamBuilder builder;
builder.setFormat(AudioFormat::I16);
builder.setChannelCount(1);
builder.setSampleRate(kSampleRateHz);
builder.setCallback(this);
builder.setPerformanceMode(PerformanceMode::LowLatency);
builder.setSharingMode(SharingMode::Exclusive);
LOGD("After creating a builder");
// Open stream
Result result = builder.openStream(&mAudioStream);
if (result != Result::OK){
LOGE("Failed to open stream. Error: %s", convertToText(result));
}
LOGD("After openstream");
// Reduce stream latency by setting the buffer size to a multiple of the burst size
mAudioStream->setBufferSizeInFrames(mAudioStream->getFramesPerBurst() * 2);
// Start the stream
result = mAudioStream->requestStart();
if (result != Result::OK){
LOGE("Failed to start stream. Error: %s", convertToText(result));
}
LOGD("After starting stream");
They are called appropriately to play with standard code (as per Google tutorials) at required times:
stdSN->setPlaying(true);
stdHT->setPlaying(true); //Nasty Sound
The audio callback is standard (as per Google tutorials):
DataCallbackResult SoundFunctions::onAudioReady(AudioStream *mAudioStream, void *audioData, int32_t numFrames) {
// Play the stream
mMixer.renderAudio(static_cast<int16_t*>(audioData), numFrames);
return DataCallbackResult::Continue;
}
The std_kit_sn.raw plays fine. But std_kit_ht.raw has a nasty distortion. Both play with low latency. Why is one playing fine and the other has a nasty distortion?

I loaded your sample project and I believe the distortion you hear is caused by clipping/wraparound during mixing of sounds.
The Mixer object from the sample is a summing mixer. It just adds the values of each track together and outputs the sum.
You need to add some code to reduce the volume of each track to avoid exceeding the limits of an int16_t (although you're welcome to file a bug on the oboe project and I'll try to add this in an upcoming version). If you exceed this limit you'll get wraparound which is causing the distortion.
Additionally, your app is hardcoded to run at 22050 frames/sec. This will result in sub-optimal latency across most mobile devices because the stream is forced to upsample to the audio device's native frame rate. A better approach would be to leave the sample rate undefined when opening the stream - this will give you the optimal frame rate for the current audio device - then use a resampler on your source files to supply audio at this frame rate.

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.

MediaCodec audio/video muxing issues ond Android

I am transcoding videos based on the example given by Google (https://android.googlesource.com/platform/cts/+/master/tests/tests/media/src/android/media/cts/ExtractDecodeEditEncodeMuxTest.java)
Basically, transocding of MP4 files works, but on some phones I get some weird results. If for example I transcode a video with audio on an HTC One, the code won't give any errors but the file cannot play afterward on the phone. If I have a 10 seconds video it jumps to almost the last second and you only here some crackling noise. If you play the video with VLC the audio track is completely muted.
I did not alter the code in terms of encoding/decoding and the same code gives correct results on a Nexus 5 or MotoX for example.
Anybody having an idea why it might fail on that specific device?
Best regard and thank you,
Florian

I made it work in Android 4.4.2 devices by following changes:
Set AAC profile to AACObjectLC instead of AACObjectHE
private static final int OUTPUT_AUDIO_AAC_PROFILE = MediaCodecInfo.CodecProfileLevel.AACObjectLC;
During creation of output audio format, use sample rate and channel count of input format instead of fixed values
MediaFormat outputAudioFormat = MediaFormat.createAudioFormat(OUTPUT_AUDIO_MIME_TYPE,
inputFormat.getInteger(MediaFormat.KEY_SAMPLE_RATE),
inputFormat.getInteger(MediaFormat.KEY_CHANNEL_COUNT));
Put a check just before audio muxing audio track to control presentation timestamps. (To avoid timestampUs X < lastTimestampUs X for Audio track error)
if (audioPresentationTimeUsLast == 0) { // Defined in the begining of method
audioPresentationTimeUsLast = audioEncoderOutputBufferInfo.presentationTimeUs;
} else {
if (audioPresentationTimeUsLast > audioEncoderOutputBufferInfo.presentationTimeUs) {
audioEncoderOutputBufferInfo.presentationTimeUs = audioPresentationTimeUsLast + 1;
}
audioPresentationTimeUsLast = audioEncoderOutputBufferInfo.presentationTimeUs;
}
// Write data
if (audioEncoderOutputBufferInfo.size != 0) {
muxer.writeSampleData(outputAudioTrack, encoderOutputBuffer, audioEncoderOutputBufferInfo);
}
Hope this helps...

If original CTS tests fail you need to go to device vendors and ask for fixes

precision of Android MediaPlayer seekTo

I have a number of mp3 files that I use with Android MediaPlayer to play from certain offsets.
Using seekTo() seems to stop at correct location. player.getCurrrentPosition() returns the correct offset, but in some cases the real position is off for as much as 200 ms. The files are about 3 minutes worth of recording and the incorrect offsets seem to appear at the end. Of some of the files.
I have the same effect either trying with Android 4.0.3 device or 4.3 emulator.
Anybody has experience with "finetuning" MediaPlayer offsets? Any experience why MediaPlayer might not be working correctly with some files? They are all CBR, stereo, some have sampling frequency 22050, some 44100, different bitrates.
I'm setting the offsets from another program and saving to mp3 tags, then in case of doubt verifying manually using Audacity. Audacity agrees with my estimate of what the correct offset is, MediaPlayer seems to disagree.
I'm aware that I could use AudioTrack with raw sound files and have a better control, however it might be impractical as there are many mp3 files, so using raw sound data will make pretty large application or many large data files.
The code is nothing fancy:
player.seekTo(start);
player.start();
CountDownTimer timer = new CountDownTimer(length, 100) {
#Override
public void onTick(long millisUntilFinished) {
if (player!=null) setInt(R.id.nLocation, player.getCurrentPosition());
}
#Override
public void onFinish() {
if (player!=null) {
if (player.isPlaying()) {
player.pause();
}
setInt(R.id.nLocation, player.getCurrentPosition());
player.stop();
player.release();
player = null;
}
}
};
timer.start();

I did not manage to find the rule why the MediaPlayer interprets offset (seekTo) differently for a group of MP3 files. For example when creating a new MP3 file with the same parameters from Audacity+Lame (MPEG1, Layer III, 44100 Hz, 192 Kb/s) it worked perfectly.
However:
this can be reproduced - rip MP3 file using Windows Media Player, settings: MP3, 192 kb/s [added when edited]
I found the workaround that seems to work for any recording.
The background - in order to tell MediaPlayer to play from certain offset, I store certain data in MP3 tags. I use a separate program to set up the playback (in frames): Label A, start frame=1000, length=100 frames, Label B, start #1500 etc. Now when I need to play it back, I read the MP3 headers, determine the frame length, for example 26.12245 ms/frame and calculate the offset (1000 frames will be 26122 ms).
The workaround is to store in MP3 tag also the frame count and length in ms (or pass through again and count the frames). Then when start MediaPlayer, compare MediaPlayer.getDuration() (MediaPlayer estimate) with the duration stored in MP3 tag. Then adjust the frame size:
adjustedFrameSizeMs = realFrameSizeMs + (player.getDuration()-storedDurationMs)/storedframeCount;
In my case (for the files with incorrect offset) the adjusted frame length always was between 26.08 and 26.09 ms (instead of 26.12245).
I attempted to try see if this is because Android plays the recording quicker (so it estimates the "real time", not the time according to frame size and frame count). It seems that it really does plays quicker. But even quicker than its own estimate. For example a recording of about 1 hour:
my estimate: 2448 s
MediaPlayer: 2444 s (4 sec difference)
Audacity: 2442 s (here we are in disagreement)
Foobar: 2448 s (another witness that agrees with my estimate :-)
MediaPlayer, real play time: 2438 s
The real playtime was 6 s (0.25%) less than MediaPlayer own estimate. Another attempt on a different sample gave the same percentage difference. However the fact that Audacity and Foobar did not always agree with my estimates, does not let me put all the blame on MediaPlayer.

AudioFlinger could not create track. status: -12

I am programming for android 2.2 and am trying to using the
SoundPool class to play several sounds simultaneously but at what feel like random times sound will stop coming out of the speakers.
for each sound that would have been played this is printed in the logcat:
AudioFlinger could not create track. status: -12
Error creating AudioTrack
Audio track delete
No exception is thrown and the program continues to execute without any changes except for the lack of volume. I've had a really hard time tracking down what conditions cause the error or recreating it after it happens. I can't find the error in the documentation anywhere and am pretty much at a loss.
Any help would be greatly appreciated!
Edit: I forgot to mention that I am loading mp3 files, not ogg.

i had almost this exact same problem with some sounds i was attempting to load and play recently.
i even broke it down to loading a single mp3 that was causing this error.
one thing i noted: when i loaded with a loop of -1, it would fail with the "status 12" error, but when i loaded it to loop 0 times, it would succeed. even attempting to load 1 time failed.
the final solution was to open the mp3 in an audio editor and re-edit it with slightly lesser quality so that the file is now smaller, and doesn't seem to take up quite as many resources in the system.
finally, there is this discussion that encourages performing a release on the objects you are using, because there is indeed a hard limit on the resources that can be used, and it is system-wide, so if you use several of the resources, other apps will not be able to use them.
https://groups.google.com/forum/#!topic/android-platform/tyITQ09vV3s/discussion%5B1-25%5D
For audio, there's a hard limit of 32 active AudioTrack objects per
device (not per app: you need to share those 32 with rest of the system), and AudioTrack is used internally beneath SoundPool,
ToneGenerator, MediaPlayer, native audio based on OpenSL ES, etc. But
the actual AudioTrack limit is < 32; it depends more on soft factors
such as memory, CPU load, etc. Also note that the limiter in the
Android audio mixer does not currently have dynamic range compression,
so it is possible to clip if you have a large number of active sounds
and they're all loud.
For video players the limit is much much lower due to the intense load
that video puts on the device.
I'll use this as an opportunity to remind media developers: please
remember to call release() for media objects when your app is paused.
This frees up the underlying resources that other apps will need.
Don't rely on the media objects being cleaned up in finalize by the
garbage collector, as that has unpredictable timing.

I had a similar issue where the music tracker within my Android game would drop notes and I got the Audioflinger error (although my status was -22). I got it working however so this might help some people.
The problem occurred when a single sample was being output multiple times simultaneously. So in my case it was a single sample being played on two or more tracks. This seemed to occasionally deadlock or something and one of the two notes would be dropped. The solution was to have two copies of the sample (two actual ogg files - identical but both in the assets). Then on each track even although I was playing the same sample, it was coming from a different file. This totally fixed the issue for me.
Not sure why it works as I cache the samples into memory, but even loading the same file into two different sounds didn't fix it. Only when the samples came out of two different files did the errors go away.
I'm sure this won't help everyone and it's not the prettiest fix but it might help someone.

john.k.doe is right. You must reduce the size of your mp3 file. You should keep the size under 100kb per file. I had to reduce my 200kb file to 72kb using a constante bit rate(CBR) of 32kbps instead of the usual 128kbps. That worked for me!

Try
final ToneGenerator tg = new ToneGenerator(AudioManager.STREAM_NOTIFICATION, 50);
tg.startTone(ToneGenerator.TONE_PROP_BEEP, 200);
tg.release();
Releasing should keep your resources.

I was with this problem. In order to solve it i run the method .release() of SoundPool object after finish playing the sound.
Here's my code:
SoundPool pool = new SoundPool(10, AudioManager.STREAM_MUSIC, 50);
final int teste = pool.load(this.ctx,this.soundS,1);
pool.setOnLoadCompleteListener(new OnLoadCompleteListener(){
#Override
public void onLoadComplete(SoundPool sound,int sampleId,int status){
pool.play(teste, 20,20, 1, 0, 1);
new Thread(new Runnable(){
#Override
public void run(){
try {
Thread.sleep(2000);
pool.release();
} catch (InterruptedException e) { e.printStackTrace(); }
}
}).start();
}
});
Note that in my case my sounds had length 1-2 seconds max, so i put the value of 2000 miliseconds in Thread.sleep(), in order to only release the resources after the player have had finished.

Like said above, there is a problem with looping: when I set repeat to -1 I get this error, but with 0 everything is working properly.
I've noticed that some sounds give this error when I'm trying to play them one by one. For example:
mSoundPool.stop(mStreamID);
mStreamID = mSoundPool.play(mRandID, mVolume, mVolume, 1, -1, 1f);
In such case, first track is played ok, but when I switch sounds, next track gives this error. It seems that using looping, a buffer is somehow overloaded, and mSoundPool.stop cannot release resources immediately.
Solution:
final Handler handler = new Handler();
handler.postDelayed(new Runnable() {
#Override
public void run() {
mStreamID = mSoundPool.play(mRandID, mVolume, mVolume, 1, -1, 1f);
}, 350);
And it's working, but delay is different for different devices.

In my case, reducing the quality and thereby the file sizes of the MP3's to under 100kb wasn't sufficient, as some 51kb files worked while some longer duration 41kb files still did not.
What helped us was reducing the sample rate from 44100 to 22050 or shortening the duration to less than 5 seconds.

I see too many overcomplicated answer. Error -12 means that you did not release the variables.
I had the same problem after I played an OGG audio file 8 times.
This worked for me:
SoundPoolPlayer onBeep; //Global variable
if(onBeep!=null){
onBeep.release();
}
onBeep = SoundPoolPlayer.create(getContext(), R.raw.micon);
onBeep.setOnCompletionListener(
new MediaPlayer.OnCompletionListener() {
#Override
public void onCompletion(MediaPlayer mp) { //mp will be null here
loge("ON Beep! END");
startGoogleASR_API_inner();
}
}
);
onBeep.play();
Releasing the variable right after .play() would mess things up, and it is not possible to release the variable inside onCompletion, so notice how I release the variable before using it(and checking for null to avoid nullpointer exceptions).
It works like charm!

A single soundPool has an internal memory limitation of 1 (one) Mb. You might be hitting this if your sound is very high quality. If you have many sounds and are hitting this limit, just create more soundpools, and distribute your sounds across them.
You may not even be able to reach the hard track limit if you are running out of memory before you get there.
That error not only appears when the stream or track limit has been reached, but also the memory limit. Soundpool will stop playing old and/or de-prioritized sounds in order to play a new sound.