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https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-11-29 06:50:59 +00:00
livesync: Split fallback_duration into in_ and out_duration
Make it independent of the `latency`; this was inconsistent anyway, where the default latency of zero got you a fallback duration of 100 ms and something else got you half the latency. Maintain a separate duration for the `in` and the `out` side so we change the duration of repeat buffers after a caps change, not just before. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
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f94ecfc7a6
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1 changed files with 36 additions and 37 deletions
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@ -36,6 +36,16 @@ fn audio_info_from_caps(
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.transpose()
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}
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fn duration_from_caps(caps: &gst::CapsRef) -> Option<gst::ClockTime> {
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caps.structure(0)
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.filter(|s| s.name().starts_with("video/"))
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.and_then(|s| s.get::<gst::Fraction>("framerate").ok())
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.filter(|framerate| framerate.denom() > 0 && framerate.numer() > 0)
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.and_then(|framerate| {
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gst::ClockTime::SECOND.mul_div_round(framerate.denom() as u64, framerate.numer() as u64)
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})
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}
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#[derive(Debug, Clone, Copy, PartialEq, Eq)]
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enum BufferLateness {
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OnTime,
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@ -82,9 +92,6 @@ struct State {
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/// Latency reported by upstream
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upstream_latency: Option<gst::ClockTime>,
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/// Duration we assume for buffers without one
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fallback_duration: gst::ClockTime,
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/// Whether we're in PLAYING state
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playing: bool,
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@ -118,6 +125,12 @@ struct State {
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/// Audio format of our srcpad
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out_audio_info: Option<gst_audio::AudioInfo>,
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/// Duration from caps on our sinkpad
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in_duration: Option<gst::ClockTime>,
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/// Duration from caps on our srcpad
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out_duration: Option<gst::ClockTime>,
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/// Queue between sinkpad and srcpad
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queue: VecDeque<Item>,
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@ -159,7 +172,9 @@ const PROP_OUT: &str = "out";
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const PROP_DUPLICATE: &str = "duplicate";
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const DEFAULT_LATENCY: gst::ClockTime = gst::ClockTime::ZERO;
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const MINIMUM_DURATION: gst::ClockTime = gst::ClockTime::from_mseconds(8);
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const DEFAULT_DURATION: gst::ClockTime = gst::ClockTime::from_mseconds(100);
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const MAXIMUM_DURATION: gst::ClockTime = gst::ClockTime::from_seconds(10);
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const MINIMUM_LATE_THRESHOLD: gst::ClockTime = gst::ClockTime::ZERO;
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const DEFAULT_LATE_THRESHOLD: Option<gst::ClockTime> = Some(gst::ClockTime::from_seconds(2));
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@ -170,7 +185,6 @@ impl Default for State {
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late_threshold: DEFAULT_LATE_THRESHOLD,
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single_segment: false,
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upstream_latency: None,
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fallback_duration: DEFAULT_DURATION,
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playing: false,
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eos: false,
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srcresult: Err(gst::FlowError::Flushing),
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@ -180,6 +194,8 @@ impl Default for State {
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out_segment: None,
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in_caps: None,
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pending_caps: None,
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in_duration: None,
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out_duration: None,
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in_audio_info: None,
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out_audio_info: None,
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queue: VecDeque::with_capacity(32),
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@ -346,7 +362,6 @@ impl ObjectImpl for LiveSync {
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match pspec.name() {
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PROP_LATENCY => {
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state.latency = value.get().unwrap();
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state.update_fallback_duration();
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let _ = self.obj().post_message(gst::message::Latency::new());
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}
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@ -488,31 +503,6 @@ impl State {
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})
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}
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fn update_fallback_duration(&mut self) {
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self.fallback_duration = self
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// First, try 1/framerate from the caps
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.in_caps
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.as_ref()
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.and_then(|c| c.structure(0))
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.filter(|s| s.name().starts_with("video/"))
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.and_then(|s| s.get::<gst::Fraction>("framerate").ok())
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.filter(|framerate| framerate.denom() > 0 && framerate.numer() > 0)
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.and_then(|framerate| {
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gst::ClockTime::SECOND
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.mul_div_round(framerate.denom() as u64, framerate.numer() as u64)
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})
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.filter(|&dur| dur > 8.mseconds() && dur < 10.seconds())
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// Otherwise, half the configured latency
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.or_else(|| Some(self.latency / 2))
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// In any case, don't allow a zero duration
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.filter(|&dur| dur > gst::ClockTime::ZERO)
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// Safe default
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.unwrap_or(DEFAULT_DURATION);
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// Change the duration of the next duplicate buffer
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self.out_buffer_duplicate = false;
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}
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fn pending_events(&self) -> bool {
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self.pending_caps.is_some() || self.pending_segment.is_some()
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}
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@ -583,8 +573,8 @@ impl LiveSync {
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state.in_segment = None;
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state.in_caps = None;
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state.in_audio_info = None;
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state.in_duration = None;
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state.in_timestamp = None;
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state.update_fallback_duration();
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}
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fn src_reset(&self, state: &mut State) {
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@ -592,6 +582,7 @@ impl LiveSync {
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state.out_segment = None;
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state.pending_caps = None;
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state.out_audio_info = None;
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state.out_duration = None;
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state.out_buffer = None;
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state.out_buffer_duplicate = false;
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state.out_timestamp = None;
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@ -669,10 +660,12 @@ impl LiveSync {
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}
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};
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let duration = duration_from_caps(&caps);
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let mut state = self.state.lock();
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state.in_caps = Some(caps);
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state.in_audio_info = audio_info;
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state.update_fallback_duration();
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state.in_duration = duration;
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}
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gst::EventView::Gap(_) => {
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@ -890,13 +883,17 @@ impl LiveSync {
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);
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}
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} else {
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gst::debug!(CAT, imp: self, "Incoming buffer without duration");
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gst::debug!(CAT, imp: self, "Patching incoming buffer with duration {calc_duration}");
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}
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buf_mut.set_duration(calc_duration);
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} else if buf_mut.duration().is_none() {
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gst::debug!(CAT, imp: self, "Incoming buffer without duration");
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buf_mut.set_duration(state.fallback_duration);
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let duration = state.in_duration.map_or(DEFAULT_DURATION, |dur| {
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dur.clamp(MINIMUM_DURATION, MAXIMUM_DURATION)
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});
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gst::debug!(CAT, imp: self, "Patching incoming buffer with duration {duration}");
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buf_mut.set_duration(duration);
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}
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// At this stage we should really really have a segment
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@ -1071,7 +1068,6 @@ impl LiveSync {
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gst::EventView::Caps(e) => {
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state.pending_caps = Some(e.caps_owned());
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state.update_fallback_duration();
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push = false;
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}
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@ -1158,6 +1154,7 @@ impl LiveSync {
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state.srcresult?;
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state.out_audio_info = audio_info_from_caps(&caps).unwrap();
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state.out_duration = duration_from_caps(&caps);
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}
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if let Some(segment) = segment {
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@ -1305,7 +1302,9 @@ impl LiveSync {
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let buffer = out_buffer.make_mut();
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if !duplicate {
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let duration = state.fallback_duration;
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let duration = state.out_duration.map_or(DEFAULT_DURATION, |dur| {
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dur.clamp(MINIMUM_DURATION, MAXIMUM_DURATION)
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});
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if let Some(audio_info) = &state.out_audio_info {
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let Some(size) = audio_info
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