gst_rtp_base_audio_payload_flush

gstreamer_rtp_sys

Function gst_rtp_base_audio_payload_flush 

Source 
pub unsafe extern "C" fn gst_rtp_base_audio_payload_flush(
    baseaudiopayload: *mut GstRTPBaseAudioPayload,
    payload_len: c_uint,
    timestamp: GstClockTime,
) -> GstFlowReturn