#include <AudioStreamBuilder.h>
Public Member Functions inherited from Static Public Member Functions | |
| static bool | isAAudioSupported () |
| static bool | isAAudioRecommended () |
Detailed Description
Factory class for an audio Stream.
Member Function Documentation
◆ getAudioApi()
|
inline |
Get the audio API which will be requested when opening the stream. No guarantees that this is the API which will actually be used. Query the stream itself to find out the API which is being used.
If you do not specify the API, then AAudio will be used if isAAudioRecommended() returns true. Otherwise OpenSL ES will be used.
- Returns
- the requested audio API
◆ isAAudioRecommended()
Is the AAudio API recommended this device?
AAudio may be supported but not recommended because of version specific issues. AAudio is not recommended for Android 8.0 or earlier versions.
- Returns
- true if recommended
◆ isAAudioSupported()
Is the AAudio API supported on this device?
AAudio was introduced in the Oreo 8.0 release.
- Returns
- true if supported
◆ openManagedStream()
| Result oboe::AudioStreamBuilder::openManagedStream | ( | ManagedStream & | stream | ) |
Create and open a ManagedStream object based on the current builder state.
The caller must create a unique ptr, and pass by reference so it can be modified to point to an opened stream. The caller owns the unique ptr, and it will be automatically closed and deleted when going out of scope.
- Parameters
-
stream Reference to the ManagedStream (uniqueptr) used to keep track of stream
- Returns
- OBOE_OK if successful or a negative error code.
◆ openStream() [1/2]
| Result oboe::AudioStreamBuilder::openStream | ( | AudioStream ** | stream | ) |
Create and open a stream object based on the current settings.
The caller owns the pointer to the AudioStream object and must delete it when finished.
- Parameters
-
stream pointer to a variable to receive the stream address
- Returns
- OBOE_OK if successful or a negative error code
◆ openStream() [2/2]
| Result oboe::AudioStreamBuilder::openStream | ( | std::shared_ptr< oboe::AudioStream > & | stream | ) |
Create and open a stream object based on the current settings.
The caller shares the pointer to the AudioStream object. The shared_ptr is used internally by Oboe to prevent the stream from being deleted while it is being used by callbacks.
- Parameters
-
stream reference to a shared_ptr to receive the stream address
- Returns
- OBOE_OK if successful or a negative error code
◆ setAllowedCapturePolicy()
|
inline |
Specify whether this stream audio may or may not be captured by other apps or the system.
The default is AllowedCapturePolicy::Unspecified which maps to AAUDIO_ALLOW_CAPTURE_BY_ALL.
Note that an application can also set its global policy, in which case the most restrictive policy is always applied. See android.media.AudioAttributes.setAllowedCapturePolicy.
Added in API level 29 to AAudio.
- Parameters
-
inputPreset the desired level of opt-out from being captured.
- Returns
- pointer to the builder so calls can be chained
◆ setAttributionTag()
|
inline |
Declare the attribution tag of the context creating the stream.
This is usually Context#getAttributionTag().
The default, if you do not call this function, is null.
Available since API level 31.
- Parameters
-
attributionTag attributionTag of the calling context.
◆ setAudioApi()
|
inline |
If you leave this unspecified then Oboe will choose the best API for the device and SDK version at runtime.
This should almost always be left unspecified, except for debugging purposes. Specifying AAudio will force Oboe to use AAudio on 8.0, which is extremely risky. Specifying OpenSLES should mainly be used to test legacy performance/functionality.
If the caller requests AAudio and it is supported then AAudio will be used.
- Parameters
-
audioApi Must be AudioApi::Unspecified, AudioApi::OpenSLES or AudioApi::AAudio.
- Returns
- pointer to the builder so calls can be chained
◆ setBufferCapacityInFrames()
|
inline |
Set the requested buffer capacity in frames. BufferCapacityInFrames is the maximum possible BufferSizeInFrames.
The final stream capacity may differ. For AAudio it should be at least this big. For OpenSL ES, it could be smaller.
Default is kUnspecified.
- Parameters
-
bufferCapacityInFrames the desired buffer capacity in frames or kUnspecified
- Returns
- pointer to the builder so calls can be chained
◆ setCallback()
|
inline |
Specifies an object to handle data or error related callbacks from the underlying API.
This is the equivalent of calling both setDataCallback() and setErrorCallback().
Important: See AudioStreamCallback for restrictions on what may be called from the callback methods.
- Deprecated:
- Call setDataCallback(std::shared_ptr<AudioStreamDataCallback>) and setErrorCallback(std::shared_ptr<AudioStreamErrorCallback>) instead.
- Parameters
-
streamCallback
- Returns
- pointer to the builder so calls can be chained
◆ setChannelConversionAllowed()
|
inline |
If true then Oboe might convert channel counts to achieve optimal results. On some versions of Android for example, stereo streams could not use a FAST track. So a mono stream might be used instead and duplicated to two channels. On some devices, mono streams might be broken, so a stereo stream might be opened and converted to mono.
Default is false.
◆ setChannelCount()
|
inline |
Request a specific number of channels.
Default is kUnspecified. If the value is unspecified then the application should query for the actual value after the stream is opened.
As the channel count here may be different from the corresponding channel count of provided channel mask used in setChannelMask(). The last called will be respected if this function and setChannelMask() are called.
◆ setChannelMask()
|
inline |
Request a specific channel mask.
Default is kUnspecified. If the value is unspecified then the application should query for the actual value after the stream is opened.
As the corresponding channel count of provided channel mask here may be different from the channel count used in setChannelCount(). The last called will be respected if this function and setChannelCount() are called.
As the setChannelMask API is available on Android 32+, this call will only take effects on Android 32+.
◆ setContentType()
|
inline |
Set the type of audio data that an output stream will carry.
The system will use this information to optimize the behavior of the stream. This could, for example, affect whether a stream is paused when a notification occurs. The contentType is ignored for input streams.
The default, if you do not call this function, is ContentType::Music.
Added in API level 28.
- Parameters
-
contentType the type of audio data, eg. ContentType::Speech
◆ setDataCallback() [1/2]
|
inline |
Pass a raw pointer to a data callback. This is not recommended because the dataCallback object might get deleted by the app while it is being used.
- Parameters
-
dataCallback
- Returns
- pointer to the builder so calls can be chained
◆ setDataCallback() [2/2]
|
inline |
Specifies an object to handle data related callbacks from the underlying API.
Important: See AudioStreamCallback for restrictions on what may be called from the callback methods.
We pass a shared_ptr so that the sharedDataCallback object cannot be deleted before the stream is deleted.
- Parameters
-
sharedDataCallback
- Returns
- pointer to the builder so calls can be chained
◆ setDeviceId()
|
inline |
Request a stream to a specific audio input/output device given an audio device ID.
In most cases, the primary device will be the appropriate device to use, and the deviceId can be left kUnspecified.
The ID could be obtained from the Java AudioManager. AudioManager.getDevices() returns an array of AudioDeviceInfo, which contains a getId() method. That ID can be passed to this function.
It is possible that you may not get the device that you requested. So if it is important to you, you should call stream->getDeviceId() after the stream is opened to verify the actual ID.
Note that when using OpenSL ES, this will be ignored and the created stream will have deviceId kUnspecified.
- Parameters
-
deviceId device identifier or kUnspecified
- Returns
- pointer to the builder so calls can be chained
◆ setDirection()
|
inline |
Request the direction for a stream. The default is Direction::Output.
- Parameters
-
direction Direction::Output or Direction::Input
◆ setErrorCallback() [1/2]
|
inline |
Pass a raw pointer to an error callback. This is not recommended because the errorCallback object might get deleted by the app while it is being used.
- Parameters
-
errorCallback
- Returns
- pointer to the builder so calls can be chained
◆ setErrorCallback() [2/2]
|
inline |
Specifies an object to handle error related callbacks from the underlying API. This can occur when a stream is disconnected because a headset is plugged in or unplugged. It can also occur if the audio service fails or if an exclusive stream is stolen by another stream.
Important: See AudioStreamCallback for restrictions on what may be called from the callback methods.
When an error callback occurs, the associated stream must be stopped and closed in a separate thread.
We pass a shared_ptr so that the errorCallback object cannot be deleted before the stream is deleted. If the stream was created using a shared_ptr then the stream cannot be deleted before the error callback has finished running.
- Parameters
-
sharedErrorCallback
- Returns
- pointer to the builder so calls can be chained
◆ setFormat()
|
inline |
Request a sample data format, for example Format::Float.
Default is Format::Unspecified. If the value is unspecified then the application should query for the actual value after the stream is opened.
◆ setFormatConversionAllowed()
|
inline |
If true then Oboe might convert data formats to achieve optimal results. On some versions of Android, for example, a float stream could not get a low latency data path. So an I16 stream might be opened and converted to float.
Default is false.
◆ setFramesPerCallback()
|
inline |
- Deprecated:
- use
setFramesPerDataCallbackinstead.
◆ setFramesPerDataCallback()
|
inline |
Request a specific number of frames for the data callback.
Default is kUnspecified. If the value is unspecified then the actual number may vary from callback to callback.
If an application can handle a varying number of frames then we recommend leaving this unspecified. This allow the underlying API to optimize the callbacks. But if your application is, for example, doing FFTs or other block oriented operations, then call this function to get the sizes you need.
Calling setFramesPerDataCallback() does not guarantee anything about timing. This just collects the data into a the number of frames that your app requires. We encourage leaving this unspecified in most cases.
If this number is larger than the burst size, some bursts will not receive a callback. If this number is smaller than the burst size, there may be multiple callbacks in a single burst.
- Parameters
-
framesPerCallback
- Returns
- pointer to the builder so calls can be chained
◆ setInputPreset()
|
inline |
Set the input (capture) preset for the stream.
The system will use this information to optimize the behavior of the stream. This could, for example, affect which microphones are used and how the recorded data is processed.
The default, if you do not call this function, is InputPreset::VoiceRecognition. That is because VoiceRecognition is the preset with the lowest latency on many platforms.
Added in API level 28.
- Parameters
-
inputPreset the desired configuration for recording
◆ setIsContentSpatialized()
|
inline |
Specifies whether the audio data of this output stream has already been processed for spatialization.
If the stream has been processed for spatialization, setting this to true will prevent issues such as double-processing on platforms that will spatialize audio data.
This is false by default.
Available since API level 32.
- Parameters
-
isContentSpatialized whether the content is already spatialized
- Returns
- pointer to the builder so calls can be chained
◆ setPackageName()
|
inline |
Declare the name of the package creating the stream.
This is usually Context#getPackageName().
The default, if you do not call this function, is a random package in the calling uid. The vast majority of apps have only one package per calling UID. If an invalid package name is set, input streams may not be given permission to record when started.
The package name is usually the applicationId in your app's build.gradle file.
Available since API level 31.
- Parameters
-
packageName packageName of the calling app.
◆ setPerformanceMode()
|
inline |
Request a performance level for the stream. This will determine the latency, the power consumption, and the level of protection from glitches.
- Parameters
-
performanceMode for example, PerformanceMode::LowLatency
- Returns
- pointer to the builder so calls can be chained
◆ setPrivacySensitiveMode()
|
inline |
Indicates whether this input stream must be marked as privacy sensitive or not.
When PrivacySensitiveMode::Enabled, this input stream is privacy sensitive and any concurrent capture is not permitted.
This is off (PrivacySensitiveMode::Disabled) by default except when the input preset is InputPreset::VoiceRecognition or InputPreset::Camcorder
Always takes precedence over default from input preset when set explicitly.
Only relevant if the stream direction is Direction::Input and AAudio is used.
Added in API level 30 to AAudio.
- Parameters
-
privacySensitive PrivacySensitiveMode::Enabled if capture from this stream must be marked as privacy sensitive, PrivacySensitiveMode::Disabled if stream should be marked as not sensitive.
- Returns
- pointer to the builder so calls can be chained
◆ setSampleRate()
|
inline |
Request a specific sample rate in Hz.
Default is kUnspecified. If the value is unspecified then the application should query for the actual value after the stream is opened.
Technically, this should be called the "frame rate" or "frames per second", because it refers to the number of complete frames transferred per second. But it is traditionally called "sample rate". Se we use that term.
◆ setSampleRateConversionQuality()
|
inline |
Specify the quality of the sample rate converter in Oboe.
If set to None then Oboe will not do sample rate conversion. But the underlying APIs might still do sample rate conversion if you specify a sample rate. That can prevent you from getting a low latency stream.
If you do the conversion in Oboe then you might still get a low latency stream.
Default is SampleRateConversionQuality::None
◆ setSessionId()
|
inline |
Set the requested session ID.
The session ID can be used to associate a stream with effects processors. The effects are controlled using the Android AudioEffect Java API.
The default, if you do not call this function, is SessionId::None.
If set to SessionId::Allocate then a session ID will be allocated when the stream is opened.
The allocated session ID can be obtained by calling AudioStream::getSessionId() and then used with this function when opening another stream. This allows effects to be shared between streams.
Session IDs from Oboe can be used the Android Java APIs and vice versa. So a session ID from an Oboe stream can be passed to Java and effects applied using the Java AudioEffect API.
Allocated session IDs will always be positive and nonzero.
Added in API level 28.
- Parameters
-
sessionId an allocated sessionID or SessionId::Allocate
◆ setSharingMode()
|
inline |
Request a mode for sharing the device. The requested sharing mode may not be available. So the application should query for the actual mode after the stream is opened.
- Parameters
-
sharingMode SharingMode::Shared or SharingMode::Exclusive
- Returns
- pointer to the builder so calls can be chained
◆ setSpatializationBehavior()
|
inline |
Sets the behavior affecting whether spatialization will be used.
The AAudio system will use this information to select whether the stream will go through a spatializer effect or not when the effect is supported and enabled.
This is SpatializationBehavior::Never by default.
Available since API level 32.
- Parameters
-
spatializationBehavior the desired spatialization behavior
- Returns
- pointer to the builder so calls can be chained
◆ setUsage()
|
inline |
Set the intended use case for an output stream.
The system will use this information to optimize the behavior of the stream. This could, for example, affect how volume and focus is handled for the stream. The usage is ignored for input streams.
The default, if you do not call this function, is Usage::Media.
Added in API level 28.
- Parameters
-
usage the desired usage, eg. Usage::Game
◆ willUseAAudio()
|
inline |
- Returns
- true if AAudio will be used based on the current settings.
The documentation for this class was generated from the following file:
- include/oboe/AudioStreamBuilder.h
