Fix various new clippy warnings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>

audio/audiofx/src/audioecho/ring_buffer.rs

@@ -15,8 +15,8 @@ pub struct RingBuffer {
 
 impl RingBuffer {
     pub fn new(size: usize) -> Self {
-        let mut buffer = Vec::with_capacity(size as usize);
+        let mut buffer = Vec::with_capacity(size);
-        buffer.extend(iter::repeat(0.0).take(size as usize));
+        buffer.extend(iter::repeat(0.0).take(size));
 
         Self {
             buffer: buffer.into_boxed_slice(),

audio/audiofx/src/audioloudnorm/imp.rs

@@ -611,9 +611,8 @@ impl State {
         // the position where we have to start writing the next 100ms in the next
         // iteration.
 
-        let mut outbuf = gst::Buffer::with_size(
+        let mut outbuf =
-            self.current_samples_per_frame as usize * self.info.bpf() as usize,
+            gst::Buffer::with_size(self.current_samples_per_frame * self.info.bpf() as usize)
-        )
                 .map_err(|_| gst::FlowError::Error)?;
         {
             let outbuf = outbuf.get_mut().unwrap();
@@ -819,7 +818,7 @@ impl State {
         // adjustment. frame_type should only ever be set to Final at the end if we ended up in
         // Inner state before.
         if self.frame_type == FrameType::First
-            && (src.len() / self.info.channels() as usize) < self.current_samples_per_frame as usize
+            && (src.len() / self.info.channels() as usize) < self.current_samples_per_frame
         {
             self.process_first_frame_is_last(imp)?;
         }

audio/audiofx/src/hrtfrender/imp.rs

@@ -373,7 +373,7 @@ impl HrtfRender {
             let (prev_offset, _) = state.adapter.prev_offset();
             let offset = prev_offset.checked_add(distance_samples).unwrap_or(0);
 
-            let duration_samples = outputsz / outbpf as usize;
+            let duration_samples = outputsz / outbpf;
             let duration = samples_to_time(duration_samples as u64);
 
             (pts, offset, duration)

audio/csound/tests/csound_filter.rs

@@ -260,10 +260,7 @@ fn csound_filter_underflow() {
     }
 
     assert_eq!(num_buffers, UNDERFLOW_NUM_BUFFERS / 2);
-    assert_eq!(
+    assert_eq!(num_samples, UNDERFLOW_NUM_SAMPLES * UNDERFLOW_NUM_BUFFERS);
-        num_samples as usize,
-        UNDERFLOW_NUM_SAMPLES * UNDERFLOW_NUM_BUFFERS
-    );
 }
 
 // Verifies that the caps negotiation is properly done, by pushing buffers whose caps

audio/lewton/src/lewtondec/imp.rs

@@ -405,7 +405,7 @@ impl LewtonDec {
         let outbuf = if let Some(ref reorder_map) = state.reorder_map {
             let mut outbuf = self
                 .obj()
-                .allocate_output_buffer(sample_count as usize * audio_info.bpf() as usize);
+                .allocate_output_buffer(sample_count * audio_info.bpf() as usize);
             {
                 // And copy the decoded data into our output buffer while reordering the channels to the
                 // GStreamer channel order

generic/sodium/src/decrypter/imp.rs

@@ -498,7 +498,7 @@ impl Decrypter {
         gst::debug!(CAT, obj: pad, "Requested offset: {}", offset);
         gst::debug!(CAT, obj: pad, "Requested size: {}", requested_size);
 
-        let chunk_index = offset as u64 / block_size as u64;
+        let chunk_index = offset / block_size as u64;
         gst::debug!(CAT, obj: pad, "Stream Block index: {}", chunk_index);
 
         let pull_offset = offset - (chunk_index * block_size as u64);

generic/threadshare/examples/benchmark.rs

@@ -158,7 +158,7 @@ fn main() {
                 let context = build_context();
                 let source = gst::ElementFactory::make("ts-tonesrc")
                     .name(format!("source-{}", i).as_str())
-                    .property("samples-per-buffer", (wait as u32) * 8000 / 1000)
+                    .property("samples-per-buffer", wait * 8000 / 1000)
                     .property("context", &context)
                     .property("context-wait", wait)
                     .build()

generic/threadshare/src/jitterbuffer/imp.rs

@@ -28,7 +28,7 @@ use gst_rtp::RTPBuffer;
 
 use once_cell::sync::Lazy;
 
-use std::cmp::{max, min, Ordering};
+use std::cmp::Ordering;
 use std::collections::{BTreeSet, VecDeque};
 use std::mem;
 use std::sync::Arc;
@@ -412,7 +412,7 @@ impl SinkHandler {
         }
 
         if let Some(last_in_seqnum) = inner.last_in_seqnum {
-            let gap = gst_rtp::compare_seqnum(last_in_seqnum as u16, seq);
+            let gap = gst_rtp::compare_seqnum(last_in_seqnum, seq);
             if gap == 1 {
                 self.calculate_packet_spacing(inner, &mut state, rtptime, pts);
             } else {
@@ -463,7 +463,7 @@ impl SinkHandler {
             state.equidistant += 1;
         }
 
-        state.equidistant = min(max(state.equidistant, -7), 7);
+        state.equidistant = state.equidistant.clamp(-7, 7);
 
         inner.last_rtptime = Some(rtptime);
 
@@ -679,7 +679,7 @@ impl SrcHandler {
             // FIXME reason why we can expect Some for the 2 lines below
             let mut last_popped_pts = state.last_popped_pts.unwrap();
             let interval = pts.into().unwrap().saturating_sub(last_popped_pts);
-            let spacing = interval / (gap as u64 + 1);
+            let spacing = interval / (gap + 1);
 
             *discont = true;
 

mux/fmp4/examples/dash_vod.rs

@@ -225,7 +225,7 @@ fn main() -> Result<(), Error> {
 "###,
                     duration = duration, segment_timeline = segment_timeline);
 
-                std::fs::write(path, &manifest).expect("failed to write manifest");
+                std::fs::write(path, manifest).expect("failed to write manifest");
             })
             .build(),
     );

mux/fmp4/src/fmp4mux/boxes.rs

@@ -1111,8 +1111,8 @@ fn write_visual_sample_entry(
 
         if let Ok(cll) = gst_video::VideoContentLightLevel::from_caps(&stream.caps) {
             write_box(v, b"clli", move |v| {
-                v.extend((cll.max_content_light_level() as u16).to_be_bytes());
+                v.extend((cll.max_content_light_level()).to_be_bytes());
-                v.extend((cll.max_frame_average_light_level() as u16).to_be_bytes());
+                v.extend((cll.max_frame_average_light_level()).to_be_bytes());
                 Ok(())
             })?;
         }

mux/mp4/src/mp4mux/boxes.rs

@@ -928,8 +928,8 @@ fn write_visual_sample_entry(
 
         if let Ok(cll) = gst_video::VideoContentLightLevel::from_caps(&stream.caps) {
             write_box(v, b"clli", move |v| {
-                v.extend((cll.max_content_light_level() as u16).to_be_bytes());
+                v.extend((cll.max_content_light_level()).to_be_bytes());
-                v.extend((cll.max_frame_average_light_level() as u16).to_be_bytes());
+                v.extend((cll.max_frame_average_light_level()).to_be_bytes());
                 Ok(())
             })?;
         }
@@ -1454,7 +1454,7 @@ fn write_cslg(
     let composition_start_time = stream
         .earliest_pts
         .nseconds()
-        .mul_div_round(timescale as u64, gst::ClockTime::SECOND.nseconds() as u64)
+        .mul_div_round(timescale as u64, gst::ClockTime::SECOND.nseconds())
         .context("too earliest PTS")?;
     v.extend(composition_start_time.to_be_bytes());
 
@@ -1462,7 +1462,7 @@ fn write_cslg(
     let composition_end_time = stream
         .end_pts
         .nseconds()
-        .mul_div_round(timescale as u64, gst::ClockTime::SECOND.nseconds() as u64)
+        .mul_div_round(timescale as u64, gst::ClockTime::SECOND.nseconds())
         .context("too end PTS")?;
     v.extend(composition_end_time.to_be_bytes());
 

net/aws/src/aws_transcribe_parse/imp.rs

@@ -170,8 +170,8 @@ impl TranscribeParse {
                         }
                     };
 
-                    let start_pts = ((start_time as f64 * 1_000_000_000.0) as u64).nseconds();
+                    let start_pts = ((start_time * 1_000_000_000.0) as u64).nseconds();
-                    let end_pts = ((end_time as f64 * 1_000_000_000.0) as u64).nseconds();
+                    let end_pts = ((end_time * 1_000_000_000.0) as u64).nseconds();
                     let duration = end_pts.saturating_sub(start_pts);
 
                     if start_pts > last_pts {

net/hlssink3/src/playlist.rs

@@ -93,7 +93,7 @@ impl Playlist {
         }
 
         self.playlist_index += 1;
-        self.inner.media_sequence = self.playlist_index as u64 - self.inner.segments.len() as u64;
+        self.inner.media_sequence = self.playlist_index - self.inner.segments.len() as u64;
     }
 
     /// Sets the playlist to started state.

net/ndi/src/ndisrc/receiver.rs

@@ -915,14 +915,14 @@ impl Receiver {
             real_time_now,
         );
 
-        let res_timestamp = self.0.observations_timestamp[if is_audio { 0 } else { 1 }].process(
+        let res_timestamp = self.0.observations_timestamp[usize::from(!is_audio)].process(
             element,
             timestamp,
             receive_time,
             duration,
         );
 
-        let res_timecode = self.0.observations_timecode[if is_audio { 0 } else { 1 }].process(
+        let res_timecode = self.0.observations_timecode[usize::from(!is_audio)].process(
             element,
             Some(timecode),
             receive_time,

net/onvif/src/onvifmetadataoverlay/imp.rs

@@ -187,8 +187,8 @@ impl OnvifMetadataOverlay {
             gst_video::VideoFormat::Bgra,
             #[cfg(target_endian = "big")]
             gst_video::VideoFormat::Argb,
-            total_width as u32,
+            total_width,
-            total_height as u32,
+            total_height,
         )
         .ok()?;
 

net/raptorq/src/raptorqenc/imp.rs

@@ -421,7 +421,7 @@ impl RaptorqEnc {
 
         assert_eq!(state.packets.len(), state.seqnums.len());
 
-        if state.packets.len() == state.protected_packets_num as usize {
+        if state.packets.len() == state.protected_packets_num {
             // We use current buffer timing as a base for repair packets timestamps
             let now_pts = buffer.pts();
             let now_dts = buffer.dts_or_pts();

net/rtp/src/av1/depay/imp.rs

@@ -370,12 +370,12 @@ impl RTPAv1Depay {
                 let mut bitreader = BitReader::endian(reader, ENDIANNESS);
                 parse_leb128(&mut bitreader)
                     .map_err(err_opt!(self, leb_read))
-                    .ok()? as u32
+                    .ok()?
             }
         } else {
             element_size = parse_leb128(&mut BitReader::endian(&mut *reader, ENDIANNESS))
                 .map_err(err_opt!(self, leb_read))
-                .ok()? as u32;
+                .ok()?;
             is_last_obu = match rtp
                 .payload_size()
                 .cmp(&(reader.position() as u32 + element_size))

net/rtp/src/av1/pay/imp.rs

@@ -471,7 +471,7 @@ impl RTPAv1Pay {
                     state.open_obu_fragment = true;
                     size
                 } else {
-                    last_obu.bytes.len() as u32 - last_obu.offset as usize as u32
+                    last_obu.bytes.len() as u32 - last_obu.offset as u32
                 };
 
                 if !packet.omit_last_size_field {

net/rtp/src/gcc/imp.rs

@@ -397,8 +397,8 @@ impl Detector {
             - self.last_received_packets.iter().next().unwrap().1.arrival;
         let bits = self
             .last_received_packets
-            .iter()
+            .values()
-            .map(|(_seqnum, p)| p.size as f64)
+            .map(|p| p.size as f64)
             .sum::<f64>()
             * 8.;
 
@@ -523,7 +523,7 @@ impl Detector {
             }
         }
 
-        self.compute_loss_average(lost_packets as f64 / n_packets as f64);
+        self.compute_loss_average(lost_packets / n_packets as f64);
     }
 
     fn compute_loss_average(&mut self, loss_fraction: f64) {
@@ -828,7 +828,7 @@ impl State {
             }
         };
 
-        if effective_bitrate as f64 - target_bitrate as f64 > 5. * target_bitrate / 100. {
+        if effective_bitrate as f64 - target_bitrate > 5. * target_bitrate / 100. {
             gst::info!(
                 CAT,
                 "Effective rate {} >> target bitrate {} - we should avoid that \
@@ -850,16 +850,11 @@ impl State {
             let threshold_on_effective_bitrate = 1.5 * effective_bitrate as f64;
             let increase = f64::max(
                 1000.0f64,
-                f64::min(
-                    alpha * avg_packet_size_bits,
                 // Stuffing should ensure that the effective bitrate is not
                 // < target bitrate, still, make sure to always increase
                 // the bitrate by a minimum amount of 160.bits
-                    f64::max(
+                (threshold_on_effective_bitrate - self.target_bitrate_on_delay as f64)
-                        threshold_on_effective_bitrate - self.target_bitrate_on_delay as f64,
+                    .clamp(160.0, alpha * avg_packet_size_bits),
-                        160.,
-                    ),
-                ),
             );
 
             /* Additive increase */
@@ -1259,8 +1254,8 @@ impl ObjectImpl for BandwidthEstimator {
                      the element to configure the starting bitrate, in which case the
                      encoder should also use it as target bitrate",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
-                    DEFAULT_MIN_BITRATE as u32,
+                    DEFAULT_MIN_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
                 glib::ParamSpecUInt::new(
@@ -1268,7 +1263,7 @@ impl ObjectImpl for BandwidthEstimator {
                     "Minimal Bitrate",
                     "Minimal bitrate to use (in bit/sec) when computing it through the bandwidth estimation algorithm",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
                     DEFAULT_MIN_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
@@ -1277,7 +1272,7 @@ impl ObjectImpl for BandwidthEstimator {
                     "Maximal Bitrate",
                     "Maximal bitrate to use (in bit/sec) when computing it through the bandwidth estimation algorithm",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
                     DEFAULT_MAX_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),

net/webrtc/src/webrtcsink/imp.rs

@@ -275,7 +275,7 @@ impl Default for Settings {
             cc_info: CCInfo {
                 heuristic: WebRTCSinkCongestionControl::GoogleCongestionControl,
                 min_bitrate: DEFAULT_MIN_BITRATE,
-                max_bitrate: DEFAULT_MAX_BITRATE as u32,
+                max_bitrate: DEFAULT_MAX_BITRATE,
                 start_bitrate: DEFAULT_START_BITRATE,
             },
             do_fec: DEFAULT_DO_FEC,
@@ -2400,7 +2400,7 @@ impl ObjectImpl for WebRTCSink {
                     "Minimal Bitrate",
                     "Minimal bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
                     DEFAULT_MIN_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
@@ -2409,7 +2409,7 @@ impl ObjectImpl for WebRTCSink {
                     "Minimal Bitrate",
                     "Minimal bitrate to use (in bit/sec) when computing it through the congestion control algorithm",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
                     DEFAULT_MAX_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),
@@ -2418,7 +2418,7 @@ impl ObjectImpl for WebRTCSink {
                     "Start Bitrate",
                     "Start bitrate to use (in bit/sec)",
                     1,
-                    u32::MAX as u32,
+                    u32::MAX,
                     DEFAULT_START_BITRATE,
                     glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
                 ),

utils/uriplaylistbin/tests/uriplaylistbin.rs

@@ -26,7 +26,7 @@ fn file_name_to_uri(name: &str) -> String {
         r
     };
 
-    let url = url::Url::from_file_path(&input_path).unwrap();
+    let url = url::Url::from_file_path(input_path).unwrap();
     url.to_string()
 }
 

video/closedcaption/src/caption_frame.rs

@@ -57,7 +57,7 @@ impl CaptionFrame {
                 data.as_ptr() as *mut _,
                 i32::from(full),
             );
-            data.set_len(len as usize);
+            data.set_len(len);
 
             String::from_utf8(data).map_err(|_| Error)
         }

video/closedcaption/src/cea608tott/imp.rs

@@ -184,7 +184,7 @@ impl Cea608ToTt {
         m %= 60;
         let ns = time % 1_000_000_000;
 
-        (h as u64, m as u8, s as u8, (ns / 1_000_000) as u16)
+        (h, m as u8, s as u8, (ns / 1_000_000) as u16)
     }
 
     fn create_vtt_buffer(

video/closedcaption/src/jsontovtt/imp.rs

@@ -131,7 +131,7 @@ impl State {
         m %= 60;
         let ns = time % 1_000_000_000;
 
-        (h as u64, m as u8, s as u8, (ns / 1_000_000) as u16)
+        (h, m as u8, s as u8, (ns / 1_000_000) as u16)
     }
 
     fn create_vtt_buffer(

video/closedcaption/src/mcc_enc/imp.rs

@@ -103,7 +103,7 @@ impl MccEnc {
                 FixedOffset::east_opt(creation_date.utc_offset().as_seconds() as i32)
                     .and_then(|tz| {
                         tz.with_ymd_and_hms(
-                            creation_date.year() as i32,
+                            creation_date.year(),
                             creation_date.month() as u32,
                             creation_date.day_of_month() as u32,
                             creation_date.hour() as u32,

video/rav1e/src/rav1enc/imp.rs

@@ -468,7 +468,7 @@ impl ObjectImpl for Rav1Enc {
             }
             "rdo-lookahead-frames" => {
                 let settings = self.settings.lock().unwrap();
-                (settings.rdo_lookahead_frames as i32).to_value()
+                (settings.rdo_lookahead_frames).to_value()
             }
             "tune" => {
                 let settings = self.settings.lock().unwrap();

video/videofx/src/videocompare/imp.rs

@@ -355,7 +355,7 @@ impl VideoAggregatorImpl for VideoCompare {
 
         let max_distance_threshold = {
             let settings = self.settings.lock().unwrap();
-            settings.max_distance_threshold as f64
+            settings.max_distance_threshold
         };
 
         if message