Apply Clippy suggestions

These are the suggestion that could be applied automatically; others will require manual review.

Author: ladvoc

libwebrtc/src/audio_source.rs

@@ -161,7 +161,7 @@ pub mod native {
             if self.queue_size_samples == 0 {
                 // frame size must be 10ms for fast path
                 let expected_frames_per_ch = (self.sample_rate / 100) as usize;
-                if frame.data.len() % (self.num_channels as usize) != 0 {
+                if !frame.data.len().is_multiple_of(self.num_channels as usize) {
                     return Err(RtcError {
                         error_type: RtcErrorType::InvalidState,
                         message: "frame.data length not divisible by channel count".to_owned(),

libwebrtc/src/data_channel.rs

@@ -135,7 +135,7 @@ impl DataChannel {
                 observer_ptr as *mut ::std::os::raw::c_void,
             );
         }
-        Self { ffi: sys_handle, observer: observer }
+        Self { ffi: sys_handle, observer }
     }
 
     pub fn send(&self, data: &[u8], binary: bool) -> Result<(), DataChannelError> {

libwebrtc/src/ice_candidate.rs

@@ -42,7 +42,7 @@ impl IceCandidate {
         })?;
 
         let ffi = unsafe {
-            sys::lkCreateIceCandidate(c_sdp_mid.as_ptr(), sdp_mline_index as i32, c_sdp.as_ptr())
+            sys::lkCreateIceCandidate(c_sdp_mid.as_ptr(), sdp_mline_index, c_sdp.as_ptr())
         };
         Ok(IceCandidate { ffi: unsafe { sys::RefCounted::from_raw(ffi) } })
     }
@@ -56,7 +56,7 @@ impl IceCandidate {
     }
 
     pub fn sdp_mline_index(&self) -> i32 {
-        unsafe { sys::lkIceCandidateGetMlineIndex(self.ffi.as_ptr()).into() }
+        unsafe { sys::lkIceCandidateGetMlineIndex(self.ffi.as_ptr()) }
     }
 
     pub fn candidate(&self) -> String {

libwebrtc/src/native/apm.rs

@@ -46,7 +46,7 @@ impl AudioProcessingModule {
     ) -> Result<(), RtcError> {
         let samples_per_10ms = (sample_rate as usize / 100) * num_channels as usize;
         assert!(
-            data.len() % samples_per_10ms == 0 && data.len() >= samples_per_10ms,
+            data.len().is_multiple_of(samples_per_10ms) && data.len() >= samples_per_10ms,
             "slice must have a multiple of 10ms worth of samples"
         );
 
@@ -79,7 +79,7 @@ impl AudioProcessingModule {
     ) -> Result<(), RtcError> {
         let samples_per_10ms = (sample_rate as usize / 100) * num_channels as usize;
         assert!(
-            data.len() % samples_per_10ms == 0 && data.len() >= samples_per_10ms,
+            data.len().is_multiple_of(samples_per_10ms) && data.len() >= samples_per_10ms,
             "slice must have a multiple of 10ms worth of samples"
         );
 

libwebrtc/src/native/audio_mixer.rs

@@ -50,6 +50,12 @@ pub static SYS_AUDIO_MIXER_CALLBACKS: sys::lkAudioMixerSourceCallback =
         getAudioFrameWithInfo: Some(AudioMixer::audio_mixer_source_get_audio_frame_with_info),
     };
 
+impl Default for AudioMixer {
+    fn default() -> Self {
+        Self::new()
+    }
+}
+
 impl AudioMixer {
     pub fn new() -> Self {
         let ffi = unsafe { sys::lkCreateAudioMixer() };
@@ -103,9 +109,9 @@ impl AudioMixer {
         let source =
             unsafe { &*(userdata as *const AudioMixerSourceImpl<Box<dyn AudioMixerSource>>) };
         if let Some(frame) = source.inner.get_audio_frame_with_info(target_sample_rate) {
-            let samples_count = (frame.sample_rate as usize / 100) as usize;
+            let samples_count = (frame.sample_rate as usize / 100);
             assert_eq!(
-                frame.sample_rate, target_sample_rate as u32,
+                frame.sample_rate, target_sample_rate,
                 "sample rate must match target_sample_rate"
             );
             assert_eq!(
@@ -128,9 +134,9 @@ impl AudioMixer {
                     frame.num_channels,
                 );
             }
-            return sys::lkAudioFrameInfo::AUDIO_FRAME_INFO_NORMAL;
+            sys::lkAudioFrameInfo::AUDIO_FRAME_INFO_NORMAL
         } else {
-            return sys::lkAudioFrameInfo::AUDIO_FRAME_INFO_MUTE;
+            sys::lkAudioFrameInfo::AUDIO_FRAME_INFO_MUTE
         }
     }
 }

libwebrtc/src/native/frame_cryptor.rs

@@ -200,7 +200,7 @@ impl FrameCryptor {
         let observer = unsafe { &*(userdata as *const Arc<RtcFrameCryptorObserver>) };
         let str: String =
             unsafe { std::ffi::CStr::from_ptr(participant_id).to_str().unwrap().to_string() };
-        observer.on_frame_cryption_state_change(str, state.into());
+        observer.on_frame_cryption_state_change(str, state);
     }
 
     pub fn new_for_rtp_sender(
@@ -224,7 +224,7 @@ impl FrameCryptor {
                 Some(FrameCryptor::on_encryption_state_changed),
                 observer_box as *mut ::std::os::raw::c_void,
             );
-            Self { observer: observer, ffi: sys::RefCounted::from_raw(ffi) }
+            Self { observer, ffi: sys::RefCounted::from_raw(ffi) }
         }
     }
 
@@ -249,7 +249,7 @@ impl FrameCryptor {
                 Some(FrameCryptor::on_encryption_state_changed),
                 observer_box as *mut ::std::os::raw::c_void,
             );
-            Self { observer: observer, ffi: sys::RefCounted::from_raw(ffi) }
+            Self { observer, ffi: sys::RefCounted::from_raw(ffi) }
         }
     }
 

libwebrtc/src/native/yuv_helper.rs

@@ -31,7 +31,7 @@ fn i420_assert_safety(
     height: i32,
 ) {
     let height_abs = height.unsigned_abs();
-    let chroma_height = (height_abs + 1) / 2;
+    let chroma_height = height_abs.div_ceil(2);
     let min_y = (src_stride_y * height_abs) as usize;
     let min_u = (src_stride_u * chroma_height) as usize;
     let min_v = (src_stride_v * chroma_height) as usize;
@@ -50,7 +50,7 @@ fn nv12_assert_safety(
     height: i32,
 ) {
     let height_abs = height.unsigned_abs();
-    let chroma_height = (height_abs + 1) / 2;
+    let chroma_height = height_abs.div_ceil(2);
 
     let min_y = (src_stride_y * height_abs) as usize;
     let min_uv = (src_stride_uv * chroma_height) as usize;

libwebrtc/src/peer_connection.rs

@@ -100,6 +100,7 @@ impl From<OfferOptions> for lkOfferAnswerOptions {
 }
 
 #[derive(Debug, Clone)]
+#[derive(Default)]
 pub struct AnswerOptions {
     pub offer_to_receive_audio: bool,
     pub offer_to_receive_video: bool,
@@ -116,11 +117,6 @@ impl From<AnswerOptions> for lkOfferAnswerOptions {
     }
 }
 
-impl Default for AnswerOptions {
-    fn default() -> Self {
-        Self { offer_to_receive_audio: false, offer_to_receive_video: false }
-    }
-}
 
 #[derive(Debug, Clone)]
 pub struct IceCandidateError {
@@ -639,7 +635,7 @@ impl PeerConnection {
         if ffi.is_null() {
             return Err(RtcError {
                 error_type: RtcErrorType::Internal,
-                message: format!("Failed to create data channel"),
+                message: "Failed to create data channel".to_string(),
             });
         }
 

libwebrtc/src/peer_connection_factory.rs

@@ -148,7 +148,7 @@ impl PeerConnectionFactory {
                 observer_ptr as *mut ::std::os::raw::c_void,
             )
         };
-        if sys_peer == std::ptr::null_mut() {
+        if sys_peer.is_null() {
             unsafe {
                 let _ = Rc::from_raw(observer_ptr);
             }
@@ -158,7 +158,7 @@ impl PeerConnectionFactory {
             });
         }
         let ffi = unsafe { sys::RefCounted::from_raw(sys_peer) };
-        let peer = PeerConnection { observer: observer, ffi: ffi };
+        let peer = PeerConnection { observer, ffi };
         Ok(peer)
     }
 

libwebrtc/src/sys/conv.rs

@@ -300,7 +300,7 @@ pub fn rtp_parameters_to_native(params: RtpParameters) -> sys::RefCounted<sys::l
 
             sys::lkRtpCodecParametersSetPayloadType(
                 c_codec.as_ptr(),
-                codec.payload_type.try_into().unwrap(),
+                codec.payload_type.into(),
             );
             sys::lkRtpCodecParametersSetMimeType(
                 c_codec.as_ptr(),
@@ -315,7 +315,7 @@ pub fn rtp_parameters_to_native(params: RtpParameters) -> sys::RefCounted<sys::l
             if let Some(channels) = codec.channels {
                 sys::lkRtpCodecParametersSetChannels(
                     c_codec.as_ptr(),
-                    channels.try_into().unwrap(),
+                    channels.into(),
                 );
             }
             lk_codecs_vec.push_back(c_codec);