[][src]Function gstreamer_rtp_sys::gst_rtp_base_audio_payload_set_frame_options

pub unsafe extern "C" fn gst_rtp_base_audio_payload_set_frame_options(
    rtpbaseaudiopayload: *mut GstRTPBaseAudioPayload,
    frame_duration: c_int,
    frame_size: c_int
)