Function gstreamer_rtp_sys::gst_rtp_base_audio_payload_push [−][src]
pub unsafe extern "C" fn gst_rtp_base_audio_payload_push(
baseaudiopayload: *mut GstRTPBaseAudioPayload,
data: *const u8,
payload_len: c_uint,
timestamp: GstClockTime
) -> GstFlowReturn