audio: migrate to new ClockTime design

6 files changed, 104 insertions, 89 deletions

diff --git a/audio/audiofx/src/audioecho/imp.rs b/audio/audiofx/src/audioecho/imp.rs
index b3524c12..03353535 100644
--- a/audio/audiofx/src/audioecho/imp.rs
+++ b/audio/audiofx/src/audioecho/imp.rs
@@ -30,15 +30,15 @@ static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
 
 use super::ring_buffer::RingBuffer;
 
-const DEFAULT_MAX_DELAY: u64 = gst::SECOND_VAL;
-const DEFAULT_DELAY: u64 = 500 * gst::MSECOND_VAL;
+const DEFAULT_MAX_DELAY: gst::ClockTime = gst::ClockTime::SECOND;
+const DEFAULT_DELAY: gst::ClockTime = gst::ClockTime::from_seconds(500);
 const DEFAULT_INTENSITY: f64 = 0.5;
 const DEFAULT_FEEDBACK: f64 = 0.0;
 
 #[derive(Debug, Clone, Copy)]
 struct Settings {
-    pub max_delay: u64,
-    pub delay: u64,
+    pub max_delay: gst::ClockTime,
+    pub delay: gst::ClockTime,
     pub intensity: f64,
     pub feedback: f64,
 }
@@ -71,10 +71,10 @@ impl AudioEcho {
         state: &mut State,
         settings: &Settings,
     ) {
-        let delay_frames = (settings.delay as usize)
-            * (state.info.channels() as usize)
-            * (state.info.rate() as usize)
-            / (gst::SECOND_VAL as usize);
+        let delay_frames = (settings.delay
+            * (state.info.channels() as u64)
+            * (state.info.rate() as u64))
+            .seconds() as usize;
 
         for (i, (o, e)) in data.iter_mut().zip(state.buffer.iter(delay_frames)) {
             let inp = (*i).to_f64().unwrap();
@@ -99,8 +99,8 @@ impl ObjectImpl for AudioEcho {
                 glib::ParamSpec::new_uint64("max-delay",
                     "Maximum Delay",
                     "Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state)",
-                    0, u64::MAX,
-                    DEFAULT_MAX_DELAY,
+                    0, u64::MAX - 1,
+                    DEFAULT_MAX_DELAY.nseconds(),
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
                 glib::ParamSpec::new_uint64(
@@ -108,8 +108,8 @@ impl ObjectImpl for AudioEcho {
                     "Delay",
                     "Delay of the echo in nanoseconds",
                     0,
-                    u64::MAX,
-                    DEFAULT_DELAY,
+                    u64::MAX - 1,
+                    DEFAULT_DELAY.nseconds(),
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_PLAYING,
                 ),
                 glib::ParamSpec::new_double(
@@ -147,12 +147,14 @@ impl ObjectImpl for AudioEcho {
             "max-delay" => {
                 let mut settings = self.settings.lock().unwrap();
                 if self.state.lock().unwrap().is_none() {
-                    settings.max_delay = value.get().expect("type checked upstream");
+                    settings.max_delay =
+                        gst::ClockTime::from_nseconds(value.get().expect("type checked upstream"));
                 }
             }
             "delay" => {
                 let mut settings = self.settings.lock().unwrap();
-                settings.delay = value.get().expect("type checked upstream");
+                settings.delay =
+                    gst::ClockTime::from_nseconds(value.get().expect("type checked upstream"));
             }
             "intensity" => {
                 let mut settings = self.settings.lock().unwrap();
@@ -292,12 +294,12 @@ impl BaseTransformImpl for AudioEcho {
         let info = gst_audio::AudioInfo::from_caps(incaps)
             .map_err(|_| gst::loggable_error!(CAT, "Failed to parse input caps"))?;
         let max_delay = self.settings.lock().unwrap().max_delay;
-        let size = max_delay * (info.rate() as u64) / gst::SECOND_VAL;
-        let buffer_size = size * (info.channels() as u64);
+        let size = (max_delay * (info.rate() as u64)).seconds() as usize;
+        let buffer_size = size * (info.channels() as usize);
 
         *self.state.lock().unwrap() = Some(State {
             info,
-            buffer: RingBuffer::new(buffer_size as usize),
+            buffer: RingBuffer::new(buffer_size),
         });
 
         Ok(())
diff --git a/audio/audiofx/src/audioloudnorm/imp.rs b/audio/audiofx/src/audioloudnorm/imp.rs
index 2925e774..e6d7e614 100644
--- a/audio/audiofx/src/audioloudnorm/imp.rs
+++ b/audio/audiofx/src/audioloudnorm/imp.rs
@@ -232,10 +232,10 @@ impl State {
         {
             let (pts, distance) = self.adapter.prev_pts();
             let distance_samples = distance / self.info.bpf() as u64;
-            let pts = pts
-                + gst::ClockTime::from(
-                    distance_samples.mul_div_floor(gst::SECOND_VAL, self.info.rate() as u64),
-                );
+            let distance_ts = distance_samples
+                .mul_div_floor(*gst::ClockTime::SECOND, self.info.rate() as u64)
+                .map(gst::ClockTime::from_nseconds);
+            let pts = pts.zip(distance_ts).map(|(pts, dist)| pts + dist);
 
             let inbuf = self
                 .adapter
@@ -253,8 +253,11 @@ impl State {
                 outbuf.set_pts(pts);
                 outbuf.set_duration(
                     (outbuf.size() as u64)
-                        .mul_div_floor(gst::SECOND_VAL, (self.info.bpf() * self.info.rate()) as u64)
-                        .into(),
+                        .mul_div_floor(
+                            *gst::ClockTime::SECOND,
+                            (self.info.bpf() * self.info.rate()) as u64,
+                        )
+                        .map(gst::ClockTime::from_nseconds),
                 );
             }
 
@@ -270,10 +273,10 @@ impl State {
 
         let (pts, distance) = self.adapter.prev_pts();
         let distance_samples = distance / self.info.bpf() as u64;
-        let pts = pts
-            + gst::ClockTime::from(
-                distance_samples.mul_div_floor(gst::SECOND_VAL, self.info.rate() as u64),
-            );
+        let distance_ts = distance_samples
+            .mul_div_floor(*gst::ClockTime::SECOND, self.info.rate() as u64)
+            .map(gst::ClockTime::from_nseconds);
+        let pts = pts.zip(distance_ts).map(|(pts, dist)| pts + dist);
 
         let mut _mapped_inbuf = None;
         let src = if self.adapter.available() > 0 {
@@ -310,8 +313,11 @@ impl State {
             outbuf.set_pts(pts);
             outbuf.set_duration(
                 (outbuf.size() as u64)
-                    .mul_div_floor(gst::SECOND_VAL, (self.info.bpf() * self.info.rate()) as u64)
-                    .into(),
+                    .mul_div_floor(
+                        *gst::ClockTime::SECOND,
+                        (self.info.bpf() * self.info.rate()) as u64,
+                    )
+                    .map(gst::ClockTime::from_nseconds),
             );
         }
 
@@ -370,8 +376,8 @@ impl State {
         &mut self,
         element: &super::AudioLoudNorm,
         src: &[f64],
-        pts: gst::ClockTime,
-    ) -> Result<(gst::Buffer, gst::ClockTime), gst::FlowError> {
+        pts: impl Into<Option<gst::ClockTime>>,
+    ) -> Result<(gst::Buffer, Option<gst::ClockTime>), gst::FlowError> {
         // Fill our whole buffer here with the initial input, i.e. 3000ms of samples.
         self.buf.copy_from_slice(src);
 
@@ -439,7 +445,7 @@ impl State {
 
         // PTS is the input PTS for the first frame, we output the first 100ms of the input
         // buffer here
-        Ok((outbuf, pts))
+        Ok((outbuf, pts.into()))
     }
 
     fn process_fill_inner_frame(&mut self, element: &super::AudioLoudNorm, src: &[f64]) {
@@ -600,8 +606,8 @@ impl State {
         &mut self,
         element: &super::AudioLoudNorm,
         src: &[f64],
-        pts: gst::ClockTime,
-    ) -> Result<(gst::Buffer, gst::ClockTime), gst::FlowError> {
+        pts: impl Into<Option<gst::ClockTime>>,
+    ) -> Result<(gst::Buffer, Option<gst::ClockTime>), gst::FlowError> {
         // Fill in these 100ms and adjust its gain according to previous measurements, and
         // at the same time copy 100ms over to the limiter_buf.
         self.process_fill_inner_frame(element, src);
@@ -632,7 +638,9 @@ impl State {
 
         // PTS is 2.9s seconds before the input PTS as we buffer 3s of samples and just
         // outputted here the first 100ms of that.
-        let pts = pts + 100 * gst::MSECOND - 3 * gst::SECOND;
+        let pts = pts
+            .into()
+            .map(|pts| pts + 100 * gst::ClockTime::MSECOND - 3 * gst::ClockTime::SECOND);
         Ok((outbuf, pts))
     }
 
@@ -693,8 +701,8 @@ impl State {
         &mut self,
         element: &super::AudioLoudNorm,
         src: &[f64],
-        pts: gst::ClockTime,
-    ) -> Result<(gst::Buffer, gst::ClockTime), gst::FlowError> {
+        pts: impl Into<Option<gst::ClockTime>>,
+    ) -> Result<(gst::Buffer, Option<gst::ClockTime>), gst::FlowError> {
         let channels = self.info.channels() as usize;
         let num_samples = src.len() / channels;
 
@@ -769,7 +777,9 @@ impl State {
 
         // PTS is 2.9s seconds before the input PTS as we buffer 3s of samples and just
         // outputted here the first 100ms of that.
-        let pts = pts + 100 * gst::MSECOND - 3 * gst::SECOND;
+        let pts = pts
+            .into()
+            .map(|pts| pts + 100 * gst::ClockTime::MSECOND - 3 * gst::ClockTime::SECOND);
         Ok((outbuf, pts))
     }
 
@@ -777,8 +787,8 @@ impl State {
         &mut self,
         element: &super::AudioLoudNorm,
         src: &[f64],
-        pts: gst::ClockTime,
-    ) -> Result<(gst::Buffer, gst::ClockTime), gst::FlowError> {
+        pts: impl Into<Option<gst::ClockTime>>,
+    ) -> Result<(gst::Buffer, Option<gst::ClockTime>), gst::FlowError> {
         // Apply a linear scale factor to the whole buffer
 
         gst_debug!(
@@ -807,15 +817,15 @@ impl State {
         }
 
         // PTS is input PTS as we just pass through the data without latency.
-        Ok((outbuf, pts))
+        Ok((outbuf, pts.into()))
     }
 
     fn process(
         &mut self,
         element: &super::AudioLoudNorm,
         src: &[f64],
-        pts: gst::ClockTime,
-    ) -> Result<(gst::Buffer, gst::ClockTime), gst::FlowError> {
+        pts: impl Into<Option<gst::ClockTime>>,
+    ) -> Result<(gst::Buffer, Option<gst::ClockTime>), gst::FlowError> {
         self.r128_in
             .add_frames_f64(src)
             .map_err(|_| gst::FlowError::Error)?;
@@ -1690,8 +1700,8 @@ impl AudioLoudNorm {
                     let (live, min_latency, max_latency) = peer_query.result();
                     q.set(
                         live,
-                        min_latency + 3 * gst::SECOND,
-                        max_latency + 3 * gst::SECOND,
+                        min_latency + 3 * gst::ClockTime::SECOND,
+                        max_latency.map(|max| max + 3 * gst::ClockTime::SECOND),
                     );
                     true
                 } else {
diff --git a/audio/audiofx/src/audiornnoise/imp.rs b/audio/audiofx/src/audiornnoise/imp.rs
index 42efd72f..33ae01b8 100644
--- a/audio/audiofx/src/audiornnoise/imp.rs
+++ b/audio/audiofx/src/audiornnoise/imp.rs
@@ -52,16 +52,18 @@ pub struct AudioRNNoise {
 
 impl State {
     // The following three functions are copied from the csound filter.
-    fn buffer_duration(&self, buffer_size: u64) -> gst::ClockTime {
+    fn buffer_duration(&self, buffer_size: u64) -> Option<gst::ClockTime> {
         let samples = buffer_size / self.in_info.bpf() as u64;
         self.samples_to_time(samples)
     }
 
-    fn samples_to_time(&self, samples: u64) -> gst::ClockTime {
-        gst::ClockTime(samples.mul_div_round(gst::SECOND_VAL, self.in_info.rate() as u64))
+    fn samples_to_time(&self, samples: u64) -> Option<gst::ClockTime> {
+        samples
+            .mul_div_round(*gst::ClockTime::SECOND, self.in_info.rate() as u64)
+            .map(gst::ClockTime::from_nseconds)
     }
 
-    fn current_pts(&self) -> gst::ClockTime {
+    fn current_pts(&self) -> Option<gst::ClockTime> {
         // get the last seen pts and the amount of bytes
         // since then
         let (prev_pts, distance) = self.adapter.prev_pts();
@@ -71,7 +73,9 @@ impl State {
         // can be added to the prev_pts to get the
         // pts at the beginning of the adapter.
         let samples = distance / self.in_info.bpf() as u64;
-        prev_pts + self.samples_to_time(samples)
+        prev_pts
+            .zip(self.samples_to_time(samples))
+            .map(|(prev_pts, time_offset)| prev_pts + time_offset)
     }
 
     fn needs_more_data(&self) -> bool {
@@ -359,11 +363,12 @@ impl BaseTransformImpl for AudioRNNoise {
                         "Peer latency: live {} min {} max {}",
                         live,
                         min,
-                        max
+                        max.display(),
                     );
 
                     min += gst::ClockTime::from_seconds((FRAME_SIZE / 48000) as u64);
-                    max += gst::ClockTime::from_seconds((FRAME_SIZE / 48000) as u64);
+                    max = max
+                        .map(|max| max + gst::ClockTime::from_seconds((FRAME_SIZE / 48000) as u64));
                     q.set(live, min, max);
                     return true;
                 }
diff --git a/audio/audiofx/src/ebur128level/imp.rs b/audio/audiofx/src/ebur128level/imp.rs
index 940c21d2..3033db17 100644
--- a/audio/audiofx/src/ebur128level/imp.rs
+++ b/audio/audiofx/src/ebur128level/imp.rs
@@ -81,13 +81,13 @@ impl From<Mode> for ebur128::Mode {
 
 const DEFAULT_MODE: Mode = Mode::all();
 const DEFAULT_POST_MESSAGES: bool = true;
-const DEFAULT_INTERVAL: u64 = gst::SECOND_VAL;
+const DEFAULT_INTERVAL: gst::ClockTime = gst::ClockTime::SECOND;
 
 #[derive(Debug, Clone, Copy)]
 struct Settings {
     mode: Mode,
     post_messages: bool,
-    interval: u64,
+    interval: gst::ClockTime,
 }
 
 impl Default for Settings {
@@ -104,8 +104,8 @@ struct State {
     info: gst_audio::AudioInfo,
     ebur128: ebur128::EbuR128,
     num_frames: u64,
-    interval_frames: u64,
-    interval_frames_remaining: u64,
+    interval_frames: gst::ClockTime,
+    interval_frames_remaining: gst::ClockTime,
 }
 
 #[derive(Default)]
@@ -167,8 +167,8 @@ impl ObjectImpl for EbuR128Level {
                     "Interval",
                     "Interval in nanoseconds for posting messages",
                     0,
-                    u64::MAX,
-                    DEFAULT_INTERVAL,
+                    u64::MAX - 1,
+                    DEFAULT_INTERVAL.nseconds(),
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
             ]
@@ -209,13 +209,14 @@ impl ObjectImpl for EbuR128Level {
                 settings.post_messages = post_messages;
             }
             "interval" => {
-                let interval = value.get().expect("type checked upstream");
+                let interval =
+                    gst::ClockTime::from_nseconds(value.get().expect("type checked upstream"));
                 gst_info!(
                     CAT,
                     obj: obj,
                     "Changing interval from {} to {}",
-                    gst::ClockTime::from(settings.interval),
-                    gst::ClockTime::from(interval)
+                    settings.interval,
+                    interval,
                 );
                 settings.interval = interval;
             }
@@ -396,7 +397,7 @@ impl BaseTransformImpl for EbuR128Level {
 
         let interval_frames = settings
             .interval
-            .mul_div_floor(info.rate() as u64, gst::SECOND_VAL)
+            .mul_div_floor(info.rate() as u64, *gst::ClockTime::SECOND)
             .unwrap();
 
         *self.state.borrow_mut() = Some(State {
@@ -459,7 +460,10 @@ impl BaseTransformImpl for EbuR128Level {
                 state.num_frames = 0;
             }
 
-            let to_process = u64::min(state.interval_frames_remaining, frames.num_frames() as u64);
+            let to_process = u64::min(
+                state.interval_frames_remaining.nseconds(),
+                frames.num_frames() as u64,
+            );
 
             frames
                 .process(to_process, &mut state.ebur128)
@@ -472,30 +476,25 @@ impl BaseTransformImpl for EbuR128Level {
                     gst::FlowError::Error
                 })?;
 
-            state.interval_frames_remaining -= to_process;
+            state.interval_frames_remaining -= gst::ClockTime::from_nseconds(to_process);
             state.num_frames += to_process;
 
             // The timestamp we report in messages is always the timestamp until which measurements
             // are included, not the starting timestamp.
-            timestamp += gst::ClockTime::from(
-                to_process
-                    .mul_div_floor(gst::SECOND_VAL, state.info.rate() as u64)
-                    .unwrap(),
-            );
+            timestamp = timestamp.map(|ts| {
+                ts + to_process
+                    .mul_div_floor(*gst::ClockTime::SECOND, state.info.rate() as u64)
+                    .map(gst::ClockTime::from_nseconds)
+                    .unwrap()
+            });
 
             // Post a message whenever an interval is full
-            if state.interval_frames_remaining == 0 {
+            if state.interval_frames_remaining.is_zero() {
                 state.interval_frames_remaining = state.interval_frames;
 
                 if settings.post_messages {
-                    let running_time = segment
-                        .as_ref()
-                        .map(|s| s.to_running_time(timestamp))
-                        .unwrap_or(gst::CLOCK_TIME_NONE);
-                    let stream_time = segment
-                        .as_ref()
-                        .map(|s| s.to_stream_time(timestamp))
-                        .unwrap_or(gst::CLOCK_TIME_NONE);
+                    let running_time = segment.as_ref().and_then(|s| s.to_running_time(timestamp));
+                    let stream_time = segment.as_ref().and_then(|s| s.to_stream_time(timestamp));
 
                     let mut s = gst::Structure::builder("ebur128-level")
                         .field("timestamp", &timestamp)
diff --git a/audio/audiofx/tests/audioloudnorm.rs b/audio/audiofx/tests/audioloudnorm.rs
index b8163359..10ad8b23 100644
--- a/audio/audiofx/tests/audioloudnorm.rs
+++ b/audio/audiofx/tests/audioloudnorm.rs
@@ -54,7 +54,7 @@ fn run_test(
         second_input = second_input,
         num_buffers = num_buffers,
         samples_per_buffer = samples_per_buffer,
-        output_buffer_duration = samples_per_buffer as u64 * gst::SECOND_VAL / 192_000,
+        output_buffer_duration = samples_per_buffer as u64 * *gst::ClockTime::SECOND / 192_000,
         format = format,
         ))
     } else {
@@ -102,7 +102,7 @@ fn run_test(
 
     let mut eos = false;
     let bus = pipeline.bus().unwrap();
-    while let Some(msg) = bus.timed_pop(gst::CLOCK_TIME_NONE) {
+    while let Some(msg) = bus.timed_pop(gst::ClockTime::NONE) {
         use gst::MessageView;
         match msg.view() {
             MessageView::Eos(..) => {
@@ -127,17 +127,17 @@ fn run_test(
     .unwrap();
 
     let mut num_samples = 0;
-    let mut expected_ts = gst::ClockTime::from(0);
+    let mut expected_ts = gst::ClockTime::ZERO;
     for sample in samples.iter() {
         use std::cmp::Ordering;
 
         let buf = sample.buffer().unwrap();
 
-        let ts = buf.pts();
+        let ts = buf.pts().expect("undefined pts");
         match ts.cmp(&expected_ts) {
             Ordering::Greater => {
                 assert!(
-                    ts - expected_ts <= gst::ClockTime::from(1),
+                    ts - expected_ts <= gst::ClockTime::NSECOND,
                     "TS is {} instead of {}",
                     ts,
                     expected_ts
@@ -145,7 +145,7 @@ fn run_test(
             }
             Ordering::Less => {
                 assert!(
-                    expected_ts - ts <= gst::ClockTime::from(1),
+                    expected_ts - ts <= gst::ClockTime::NSECOND,
                     "TS is {} instead of {}",
                     ts,
                     expected_ts
@@ -160,8 +160,7 @@ fn run_test(
         num_samples += data.len() / channels as usize;
         r128.add_frames_f64(data).unwrap();
 
-        expected_ts +=
-            gst::ClockTime::from((data.len() as u64 / channels as u64) * gst::SECOND_VAL / 192_000);
+        expected_ts += gst::ClockTime::from_seconds(data.len() as u64 / channels as u64) / 192_000;
     }
 
     assert_eq!(
diff --git a/audio/audiofx/tests/ebur128level.rs b/audio/audiofx/tests/ebur128level.rs
index 776014a6..95a8f6d1 100644
--- a/audio/audiofx/tests/ebur128level.rs
+++ b/audio/audiofx/tests/ebur128level.rs
@@ -125,9 +125,9 @@ fn run_test(layout: gst_audio::AudioLayout, format: gst_audio::AudioFormat) {
                     let timestamp = s.get::<u64>("timestamp").unwrap();
                     let running_time = s.get::<u64>("running-time").unwrap();
                     let stream_time = s.get::<u64>("stream-time").unwrap();
-                    assert_eq!(timestamp, num_msgs * 500 * gst::MSECOND_VAL);
-                    assert_eq!(running_time, num_msgs * 500 * gst::MSECOND_VAL);
-                    assert_eq!(stream_time, num_msgs * 500 * gst::MSECOND_VAL);
+                    assert_eq!(timestamp, num_msgs * 500 * *gst::ClockTime::MSECOND);
+                    assert_eq!(running_time, num_msgs * 500 * *gst::ClockTime::MSECOND);
+                    assert_eq!(stream_time, num_msgs * 500 * *gst::ClockTime::MSECOND);
 
                     // Check if all these exist
                     let _momentary_loudness = s.get::<f64>("momentary-loudness").unwrap();