Returns information about how the application was started. This function provides details about the launch intent, such as the type of intent field FOvrLaunchDetails::LaunchType and any additional data that was passed along with it. By calling this function, you can gain insight into how your application was launched and take appropriate action based on that information.

Logs if the user successfully deeplinked to a destination. This function takes two parameters: a string tracking ID and a launch result. The tracking ID is used to identify the specific deeplink attempt, while the launch result indicates whether the deeplink was EOvrLaunchResult::Success or not. By logging this information, you can track the effectiveness of your deeplinking efforts and make adjustments as needed.

Accepts a VoIP connection from a given user.

Gets whether or not a voice connection is using discontinuous transmission (DTX). Both sides must set to using DTX when their connection is established in order for this to be true. Returns unknown if there is no connection.

Gets the current local bitrate used for the connection to the specified user. This is set by the current client. Returns unknown if there is no connection.

Returns the size of the internal ringbuffer used by the voip system in elements. This size is the maximum number of elements that can ever be returned by Voip_GetPCM().This function can be safely called from any thread.

Gets all available samples of voice data for the specified peer and copies it into outputBuffer. The voip system will generate data at roughly the rate of 480 samples per 10ms. This function should be called every frame with 50ms (2400 elements) of buffer size to account for frame rate variations. The data format is 16 bit fixed point 48khz mono.This function can be safely called from any thread. Gets all available samples of voice data for the specified peer and copies it into outputBuffer. The voip system will generate data at roughly the rate of 480 samples per 10ms. This function should be called every frame with 50ms (2400 elements) of buffer size to account for frame rate variations. The data format is 32 bit floating point 48khz mono.This function can be safely called from any thread. Returns the current number of audio samples available to read for the specified user. This function is inherently racy; it's possible that more data can be added between a call to this function and a subsequent call to Voip_GetPCM().This function can be safely called from any thread.

Like Voip_GetPCM(), this function copies available audio samples into a provided buffer, but also stores the timestamp of the first sample in the output parameter 'timestamp'. This timestamp can be used for synchronization; see Voip_GetSyncTimestamp() for more details.This function returns a number of samples copied. Note that it may return early, even if there's more data available, in order to keep the batch of audio samples with a single timestamp small; for example, if there's 30ms worth of audio in the buffer, this function may return 480 samples (10ms) each time it's called. Therefore, it's recommended to call this as long as there's data in the buffer (i.e. the function returns a non-zero result). See Voip_GetPCMWithTimestamp(). Uses a 32-bit floating-point data format. Gets the current remote bitrate used for the connection to the specified user. This is set by the client on the other side of the connection. Returns unknown if there is no connection.

Returns a timestamp used for synchronizing audio samples sent to the given user with an external data stream.Timestamps associated with audio frames are implicitly transmitted to remote peers; on the receiving side, they can be obtained by using Voip_GetPCMWithTimestamp(). Voip_GetPCMWithTimestamp() is used to fetch those timestamps on the sending side an application can insert the value returned by this function into each data packet and compare it to the value returned by GetPCMWithTimestamp() on the receiving side in order to determine the ordering of two events (sampling audio and composing a data packet).Note: the timestamp is generated by an unspecified clock; it's doesn't generally represent wall-clock time. Use Voip_GetSyncTimestampDifference() to convert the difference between two timestamps to microseconds.This function assumes that a voice connection to the user already exists; it returns 0 if that isn't the case.

Calculates the difference between two sync timestamps, returned by either Voip_GetSyncTimestamp() or Voip_GetPCMWithTimestamp(), and converts it to microseconds.Return value will be negative if lhs is smaller than rhs, zero if both timestamps are the same, and positive otherwise. The absolute value of the result is the time in microseconds between two sync timestamps.

Returns SystemVoip microphone's mute state.

Returns SystemVoip status.

This function allows you to set a callback that will be called every time audio data is captured by the microphone. The callback function must match this signature: void filterCallback(int16_t pcmData[], size_t pcmDataLength, int frequency, int numChannels); The pcmData param is used for both input and output. pcmDataLength is the size of pcmData in elements. numChannels will be 1 or 2. If numChannels is 2, then the channel data will be interleaved in pcmData. Frequency is the input data sample rate in hertz.This function can be safely called from any thread. This function is used to enable or disable the local microphone. When muted, the microphone will not transmit any audio. Voip connections are unaffected by this state. New connections can be established or closed whether the microphone is muted or not. This can be used to implement push-to-talk, or a local mute button. The default state is unmuted.This function can be safely called from any thread.

The options set for newly created connections to use. Existing connections will continue to use their current settings until they are destroyed and recreated.

Sets the output sample rate. Audio data will be resampled as it is placed into the internal ringbuffer.This function can be safely called from any thread.

Attempts to establish a VoIP session with the specified user.A message of type FOvrNotification_Voip_StateChange will be posted when the session is established.This function can be safely called from any thread.

Terminates a VoIP session with the specified user. Note that a muting functionality should be used to temporarily stop sending audio; restarting a VoIP session after tearing it down may be an expensive operation.This function can be safely called from any thread.