Fix various new clippy warnings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
This commit is contained in:
Sebastian Dröge 2022-12-13 11:43:16 +02:00
parent 289e8a08c3
commit 3f904553ea
27 changed files with 57 additions and 66 deletions

View file

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

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@ -611,10 +611,9 @@ impl State {
// the position where we have to start writing the next 100ms in the next
// iteration.
let mut outbuf = gst::Buffer::with_size(
self.current_samples_per_frame as usize * self.info.bpf() as usize,
)
.map_err(|_| gst::FlowError::Error)?;
let mut outbuf =
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();
let mut dst = outbuf.map_writable().map_err(|_| gst::FlowError::Error)?;
@ -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)?;
}

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@ -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)

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

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@ -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

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@ -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);

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@ -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()

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@ -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;

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@ -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(),
);

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@ -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_frame_average_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()).to_be_bytes());
Ok(())
})?;
}

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@ -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_frame_average_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()).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());

View file

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

View file

@ -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.

View file

@ -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,

View file

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

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@ -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();

View file

@ -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))

View file

@ -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 {

View file

@ -397,8 +397,8 @@ impl Detector {
- self.last_received_packets.iter().next().unwrap().1.arrival;
let bits = self
.last_received_packets
.iter()
.map(|(_seqnum, p)| p.size as f64)
.values()
.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,
160.,
),
),
// 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
(threshold_on_effective_bitrate - self.target_bitrate_on_delay as f64)
.clamp(160.0, alpha * avg_packet_size_bits),
);
/* 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,
DEFAULT_MIN_BITRATE as u32,
u32::MAX,
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,
),

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@ -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,
),

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@ -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()
}

View file

@ -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)
}

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@ -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(

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@ -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(

View file

@ -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,

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@ -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();

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@ -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