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https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-11-26 13:31:00 +00:00
ndisrc: Keep track of audio/video and timestamp/timecode observations separately
Audio/video are in practice not always from the same clock and can have different behaviours with regards to clock rate and jitter. Handling them separately generally gives better results for the timestamps output by the source element.
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1 changed files with 32 additions and 24 deletions
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@ -191,7 +191,9 @@ pub struct ReceiverInner {
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queue: ReceiverQueue,
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max_queue_length: usize,
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observations: Observations,
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// Audio/video time observations
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observations_timestamp: [Observations; 2],
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observations_timecode: [Observations; 2],
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element: glib::WeakRef<gst_base::BaseSrc>,
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timestamp_mode: TimestampMode,
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@ -227,6 +229,7 @@ struct ReceiverQueueInner {
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const WINDOW_LENGTH: u64 = 512;
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const WINDOW_DURATION: u64 = 2_000_000_000;
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#[derive(Default)]
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struct Observations(AtomicRefCell<ObservationsInner>);
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struct ObservationsInner {
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@ -254,26 +257,21 @@ impl Default for ObservationsInner {
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}
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impl Observations {
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fn new() -> Self {
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Self(AtomicRefCell::new(ObservationsInner::default()))
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}
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// Based on the algorithm used in GStreamer's rtpjitterbuffer, which comes from
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// Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation over Network Delays":
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// http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
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fn process(
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&self,
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element: &gst_base::BaseSrc,
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time: (Option<gst::ClockTime>, gst::ClockTime),
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remote_time: Option<gst::ClockTime>,
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local_time: gst::ClockTime,
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duration: Option<gst::ClockTime>,
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) -> (gst::ClockTime, Option<gst::ClockTime>, bool) {
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if time.0.is_none() {
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return (time.1, duration, false);
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}
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let time = (time.0.unwrap(), time.1);
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let remote_time = time.0.nseconds();
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let local_time = time.1.nseconds();
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let remote_time = match remote_time {
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None => return (local_time, duration, false),
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Some(remote_time) => remote_time.nseconds(),
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};
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let local_time = local_time.nseconds();
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gst_trace!(
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CAT,
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@ -488,7 +486,8 @@ impl Receiver {
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Condvar::new(),
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))),
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max_queue_length,
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observations: Observations::new(),
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observations_timestamp: Default::default(),
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observations_timecode: Default::default(),
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element: element.downgrade(),
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timestamp_mode,
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timeout,
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@ -780,6 +779,7 @@ impl Receiver {
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fn calculate_timestamp(
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&self,
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element: &gst_base::BaseSrc,
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is_audio: bool,
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timestamp: i64,
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timecode: i64,
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duration: Option<gst::ClockTime>,
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@ -805,17 +805,23 @@ impl Receiver {
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real_time_now,
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);
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let res_timestamp = self.0.observations_timestamp[if is_audio { 0 } else { 1 }].process(
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element,
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timestamp,
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receive_time,
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duration,
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);
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let res_timecode = self.0.observations_timecode[if is_audio { 0 } else { 1 }].process(
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element,
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Some(timecode),
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receive_time,
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duration,
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);
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let (pts, duration, discont) = match self.0.timestamp_mode {
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TimestampMode::ReceiveTimeTimecode => {
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self.0
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.observations
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.process(element, (Some(timecode), receive_time), duration)
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}
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TimestampMode::ReceiveTimeTimestamp => {
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self.0
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.observations
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.process(element, (timestamp, receive_time), duration)
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}
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TimestampMode::ReceiveTimeTimecode => res_timecode,
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TimestampMode::ReceiveTimeTimestamp => res_timestamp,
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TimestampMode::Timecode => (timecode, duration, false),
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TimestampMode::Timestamp if timestamp.is_none() => (receive_time, duration, false),
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TimestampMode::Timestamp => {
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@ -888,6 +894,7 @@ impl Receiver {
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self.calculate_timestamp(
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element,
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false,
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video_frame.timestamp(),
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video_frame.timecode(),
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duration,
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@ -1438,6 +1445,7 @@ impl Receiver {
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self.calculate_timestamp(
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element,
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true,
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audio_frame.timestamp(),
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audio_frame.timecode(),
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duration,
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