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Full Guide To Oboe

Oboe is a C++ library which makes it easy to build high-performance audio apps on Android. Apps communicate with Oboe by reading and writing data to streams.

Audio streams

Oboe moves audio data between your app and the audio inputs and outputs on your Android device. Your app passes data in and out by reading from and writing to audio streams, represented by the class AudioStream. The read/write calls can be blocking or non-blocking.

A stream is defined by the following:

  • The audio device that is the source or sink for the data in the stream.
  • The sharing mode that determines whether a stream has exclusive access to an audio device that might otherwise be shared among multiple streams.
  • The format of the audio data in the stream.

Audio device

Each stream is attached to a single audio device.

An audio device is a hardware interface or virtual endpoint that acts as a source or sink for a continuous stream of digital audio data. Don't confuse an audio device (a built-in mic or bluetooth headset) with the Android device (the phone or watch) that is running your app.

On API 23 and above you can use the AudioManager method getDevices() to discover the audio devices that are available on your Android device. The method returns information about the type of each device.

Each audio device has a unique ID on the Android device. You can use the ID to bind an audio stream to a specific audio device. However, in most cases you can let Oboe choose the default primary device rather than specifying one yourself.

The audio device attached to a stream determines whether the stream is for input or output. A stream can only move data in one direction. When you define a stream you also set its direction. When you open a stream Android checks to ensure that the audio device and stream direction agree.

Sharing mode

A stream has a sharing mode:

  • SharingMode::Exclusive (available on API 26+) means the stream has exclusive access to an endpoint on its audio device; the endpoint cannot be used by any other audio stream. If the exclusive endpoint is already in use, it might not be possible for the stream to obtain access to it. Exclusive streams provide the lowest possible latency by bypassing the mixer stage, but they are also more likely to get disconnected. You should close exclusive streams as soon as you no longer need them, so that other apps can access that endpoint. Not all audio devices provide exclusive endpoints. System sounds and sounds from other apps can still be heard when an exclusive stream is in use as they use a different endpoint.

Oboe exclusive sharing mode diagram

  • SharingMode::Shared allows Oboe to mix audio. Oboe mixes all the shared streams assigned to the same endpoint on the audio device.

Oboe exclusive sharing mode diagram

You can explicitly request the sharing mode when you create a stream, although you are not guaranteed to receive that mode. By default, the sharing mode is Shared.

Audio format

The data passed through a stream has the usual digital audio attributes, which you must specify when you define a stream. These are as follows:

  • Sample format
  • Samples per frame
  • Sample rate

Oboe permits these sample formats:

AudioFormat C data type Notes
I16 int16_t common 16-bit samples, Q0.15 format
Float float -1.0 to +1.0

Oboe might perform sample conversion on its own. For example, if an app is writing FLOAT data but the HAL uses PCM_I16, Oboe might convert the samples automatically. Conversion can happen in either direction. If your app processes audio input, it is wise to verify the input format and be prepared to convert data if necessary, as in this example:

AudioFormat dataFormat = stream->getDataFormat();
//... later
if (dataFormat == AudioFormat::I16) {

Creating an audio stream

The Oboe library follows a builder design pattern and provides the class AudioStreamBuilder.

  1. Set the audio stream configuration using an AudioStreamBuilder. Use the builder functions that correspond to the stream parameters. These optional set functions are available:
AudioStreamBuilder streamBuilder;


Note that these methods do not report errors, such as an undefined constant or value out of range.

In most cases, you will not call the method setAPIIndex(). This determines whether Oboe will use AAudio or OpenSL ES as the audio engine for you app. Oboe will automatically select the best implementation available on your device. If you want to specifically select AAudio or OpenSL, set the APIIndex yourself. After a stream has been opened, you can verify that the API you specified was chosen by calling AudioStreamBuilder::getAPIIndex(). The allowable indexes are AAudio and OpenSLES.

If you do not specify the deviceId, the default is the primary output device. If you do not specify the stream direction, the default is an output stream. For all other parameters, you can explicitly set a value, or let the system assign the optimal value by not specifying the parameter at all or setting it to kUnspecified.

To be safe, check the state of the audio stream after you create it, as explained in step 3, below.

  1. After you've configured the AudioStreamBuilder, call openStream() to open the stream:
Result result = streamBuilder.openStream(&stream_);
if (result != OK){
                        "Error opening stream %s",
  1. You should verify the stream's configuration after opening it. If you specified sample format, sample rate, or samples per frame they will not change. However, sharing mode and buffer capacity might change (whether or not you set them) depending on the capabilities of the stream's audio device and the Android device on which it's running. As a matter of good defensive programming, you should check the stream's configuration before using it. There are functions to retrieve the stream setting that corresponds to each builder setting:
AudioStreamBuilder set methods AudioStream get methods
setDeviceId() getDeviceId()
setDirection() getDirection()
setSharingMode() getSharingMode()
setSampleRate() getSampleRate()
setChannelCount() getChannelCount()
setFormat() getFormat()
setBufferCapacityInFrames() getBufferCapacityInFrames()

Using an audio stream

State transitions

An Oboe stream is usually in one of five stable states (the error state, Disconnected, is described at the end of this section):

  • Open
  • Started
  • Paused
  • Flushed
  • Stopped

Data only flows through a stream when the stream is in the Started state. To move a stream between states, use one of the functions that request a state transition:

Result result;
result = stream->requestStart();
result = stream->requestStop();
result = stream->requestPause();
result = stream->requestFlush();

Note that you can only request pause or flush on an output stream:

These functions are asynchronous, and the state change doesn't happen immediately. When you request a state change, the stream moves one of the corresponding transient states:

  • Starting
  • Pausing
  • Flushing
  • Stopping
  • Closing

The state diagram below shows the stable states as rounded rectangles, and the transient states as dotted rectangles. Though it's not shown, you can call close() from any state

Oboe Lifecycle

Oboe doesn't provide callbacks to alert you to state changes. One special function, AudioStream::waitForStateChange() can be used to wait for a state change.

The function does not detect a state change on its own, and does not wait for a specific state. It waits until the current state is different than inputState, which you specify.

For example, after requesting to pause, a stream should immediately enter the transient state Pausing, and arrive sometime later at the Paused state - though there's no guarantee it will. Since you can't wait for the Paused state, use waitForStateChange() to wait for any state other than Pausing. Here's how that's done:

StreamState inputState = StreamState::Pausing;
StreamState nextState = StreamState::Uninitialized;
int64_t timeoutNanos = 100 * kNanosPerMillisecond;
result = stream->requestPause();
result = stream->waitForStateChange(inputState, &nextState, timeoutNanos);

If the stream's state is not Pausing (the inputState, which we assumed was the current state at call time), the function returns immediately. Otherwise, it blocks until the state is no longer Pausing or the timeout expires. When the function returns, the parameter nextState shows the current state of the stream.

You can use this same technique after calling request start, stop, or flush, using the corresponding transient state as the inputState. Do not call waitForStateChange() after calling AudioStream::close() since the stream will be deleted as soon as it closes. And do not call close() while waitForStateChange() is running in another thread.

Reading and writing to an audio stream

After the stream is started you can read or write to it using the methods AudioStream::read(buffer, numFrames, timeoutNanos) and AudioStream::write(buffer, numFrames, timeoutNanos).

For a blocking read or write that transfers the specified number of frames, set timeoutNanos greater than zero. For a non-blocking call, set timeoutNanos to zero. In this case the result is the actual number of frames transferred.

When you read input, you should verify the correct number of frames was read. If not, the buffer might contain unknown data that could cause an audio glitch. You can pad the buffer with zeros to create a silent dropout:

Result result =, numFrames, timeout);
if (result < 0) {
  // Error!
if (result != numFrames) {
  // pad the buffer with zeros
  memset(static_cast<sample_type*>(audioData) + result * samplesPerFrame, 0,
      sizeof(sample_type) * (numFrames - result) * samplesPerFrame);

You can prime the stream's buffer before starting the stream by writing data or silence into it. This must be done in a non-blocking call with timeoutNanos set to zero.

The data in the buffer must match the data format returned by stream.getDataFormat().

Closing an audio stream

When you are finished using a stream, close it:


After you close a stream you cannot call any of its methods.

Disconnected audio stream

An audio stream can become disconnected at any time if one of these events happens:

  • The associated audio device is no longer connected (for example when headphones are unplugged).
  • An error occurs internally.
  • An audio device is no longer the primary audio device.

When a stream is disconnected, it has the state "Disconnected" and calls to write() or other functions will return Result::ErrorDisconnected. When a stream is disconnected, all you can do is close it.

If you need to be informed when an audio device is disconnected, write a class which extends AudioStreamCallback and implements one of the following methods:

  • onErrorBeforeClose(stream, error) - called when the stream has been stopped but not yet closed, so you can still query the stream for its properties (e.g. for number of underruns).
  • onErrorAfterClose(stream, error) - called when the stream has been closed so the stream cannot be referenced.

Register your class using builder.setCallback(yourCallbackClass).

The onError*() methods should check the state of the stream as shown in the following example. You should not close or reopen the stream from the callback, use another thread instead. Note that if you open a new stream it might have different characteristics than the original stream (for example framesPerBurst):

void PlayAudioEngine::onErrorBeforeClose(AudioStream *audioStream, Result error) {
    if (error == Result::ErrorDisconnected) {
        // Handle stream restart on a separate thread
        std::function<void(void)> restartStream = std::bind(&PlayAudioEngine::restartStream, this);
        mStreamRestartThread = new std::thread(restartStream);
    // See Definitions.h for other Result::Error* codes

Optimizing performance

You can optimize the performance of an audio application by using special high-priority threads.

Using a high priority callback

If your app reads or writes audio data from an ordinary thread, it may be preempted or experience timing jitter. This can cause audio glitches. Using larger buffers might guard against such glitches, but a large buffer also introduces longer audio latency. For applications that require low latency, an audio stream can use an asynchronous callback function to transfer data to and from your app. The callback runs in a high-priority thread that has better performance.

Your code can access the callback mechanism by implementing the virtual class AudioStreamCallback. The stream periodically executes onAudioReady() (the callback function) to acquire the data for its next burst.

class AudioEngine : AudioStreamCallback {
    DataCallbackResult AudioEngine::onAudioReady(
            AudioStream *oboeStream,
            void *audioData,
            int32_t numFrames){
        oscillator_->render(static_cast<float *>(audioData), numFrames);
        return DataCallbackResult::Continue;

    bool AudioEngine::start() {
        // register the callback
    // application data
    Oscillator* oscillator_;

Note that the callback must be registered on the stream with setCallback. Any application-specific data (such as oscillator_ in this case) can be included within the class itself.

The callback function should not perform a read or write on the stream that invoked it. If the callback belongs to an input stream, your code should process the data that is supplied in the audioData buffer (specified as the second argument). If the callback belongs to an output stream, your code should place data into the buffer.

It is possible to process more than one stream in the callback. You can use one stream as the master, and pass pointers to other streams in the class's private data. Register a callback for the master stream. Then use non-blocking I/O on the other streams. Here is an example of a round-trip callback that passes an input stream to an output stream. The master calling stream is the output stream. The input stream is included in the class.

The callback does a non-blocking read from the input stream placing the data into the buffer of the output stream.

class AudioEngine : AudioStreamCallback {

    oboe_data_callback_result_t AudioEngine::onAudioReady(
            AudioStream *oboeStream,
            void *audioData,
            int32_t numFrames){
        auto result =, numFrames, timeout);
        // result has type ResultWithValue<int32_t>, which for convenience is coerced to a Result type when compared with another Result.
        if (result == Result::OK){
		    if (result.value() == numFrames)
                return DataCallbackResult::Continue;
		    if (result.value() >= 0) {
                memset(static_cast<sample_type*>(audioData) + result.value() * samplesPerFrame, 0,
                    sizeof(sample_type) * (numFrames - result.value()) * samplesPerFrame);
                return DataCallbackResult::Continue;
        return DataCallbackResult::Stop;

    bool AudioEngine::start() {

    void setRecordingStream(AudioStream *stream) {
      recordingStream = stream;

    AudioStream *recordingStream;

Note that in this example it is assumed the input and output streams have the same number of channels, format and sample rate. The format of the streams can be mismatched - as long as the code handles the translations properly.

Callback do's and don'ts

These are things the onAudioReady method should NOT do:

  • allocate memory using, for example, malloc() or new
  • any file operations such as opening, closing, reading or writing
  • any network operations such as streaming
  • use any mutexes or other synchronization primitives
  • sleep
  • stop or close the stream
  • Call read() or write() on the stream which invoked it

The following methods are OK to call:

  • AudioStream::get*()
  • oboe::convertResultToText()

Setting performance mode

Every AudioStream has a performance mode which has a large effect on your app's behavior. There are three modes:

  • PerformanceMode::None is the default mode. It uses a basic stream that balances latency and power savings.
  • PerformanceMode::LowLatency uses smaller buffers and an optimized data path for reduced latency.
  • PerformanceMode::PowerSaving uses larger internal buffers and a data path that trades off latency for lower power.

You can select the performance mode by calling setPerformanceMode(), and discover the current mode by calling getPerformanceMode().

If low latency is more important than power savings in your application, use PerformanceMode::LowLatency. This is useful for apps that are very interactive, such as games or keyboard synthesizers.

If saving power is more important than low latency in your application, use PerformanceMode::PowerSaving. This is typical for apps that play back previously generated music, such as streaming audio or MIDI file players.

In the current version of Oboe, in order to achieve the lowest possible latency you must use the PerformanceMode::LowLatency performance mode along with a high-priority callback. Follow this example:

// Create a callback object
MyOboeStreamCallback myCallback;

// Create a stream builder
AudioStreamBuilder builder;

// Use it to create the stream
AudioStream *stream;

Thread safety

The Oboe API is not completely thread safe. You cannot call some of the Oboe functions concurrently from more than one thread at a time. This is because Oboe avoids using mutexes, which can cause thread preemption and glitches.

To be safe, don't call waitForStateChange() or read or write to the same stream from two different threads. Similarly, don't close a stream in one thread while reading or writing to it in another thread.

Calls that return stream settings, like AudioStream::getSampleRate() and AudioStream::getChannelCount(), are thread safe.

These calls are also thread safe:

  • convertToText()
  • AudioStream::get*() except for getTimestamp()

Note: When a stream uses a callback function, it's safe to read/write from the callback thread while also closing the stream from the thread in which it is running.

Code samples

Code samples are available in the samples folder.

Known Issues

The following methods are defined, but will return Result::ErrorUnimplemented:

  • setBufferSizeInFrames()
  • getBufferSizeInFrames()
  • getXRunCount()
  • getFramesRead()
  • getTimestamp()