fix errors

livekit · Dec 4, 2023 · 5c6996c · 5c6996c
1 parent ed08dcb
commit 5c6996c
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Showing 4 changed files with 6 additions and 11 deletions.
diff --git a/libwebrtc/src/audio_source.rs b/libwebrtc/src/audio_source.rs
@@ -34,7 +34,6 @@ impl RtcAudioSource {
         [Native];
         fn set_audio_options(self: &Self, options: AudioSourceOptions) -> ();
         fn audio_options(self: &Self) -> AudioSourceOptions;
-        fn captured_frames(self: &Self) -> usize;
         fn sample_rate(self: &Self) -> u32;
         fn num_channels(self: &Self) -> u32;
     );
@@ -80,10 +79,6 @@ pub mod native {
             self.handle.audio_options()
         }
 
-        pub fn captured_frames(&self) -> usize {
-            self.handle.captured_frames()
-        }
-
         pub fn sample_rate(&self) -> u32 {
             self.handle.sample_rate()
         }

diff --git a/libwebrtc/src/native/audio_source.rs b/libwebrtc/src/native/audio_source.rs
@@ -79,7 +79,7 @@ impl NativeAudioSource {
                     // (We don't want to increase the captured_frames count  and no need to buffer)
                     interval.tick().await;
 
-                    let mut inner = source.inner.lock().await;
+                    let inner = source.inner.lock().await;
                     if inner.captured_frames > 0 {
                         break; // User captured something, stop injecting silence
                     }
@@ -89,7 +89,7 @@ impl NativeAudioSource {
                         &data,
                         sample_rate,
                         num_channels,
-                        sample_rate / 100,
+                        sample_rate as usize / 100,
                     );
                 }
             }

diff --git a/webrtc-sys/include/livekit/audio_track.h b/webrtc-sys/include/livekit/audio_track.h
@@ -112,8 +112,8 @@ class AudioTrackSource {
   void set_audio_options(const AudioSourceOptions& options) const;
 
   void on_captured_frame(rust::Slice<const int16_t> audio_data,
-                         int sample_rate,
-                         size_t number_of_channels,
+                         uint32_t sample_rate,
+                         uint32_t number_of_channels,
                          size_t number_of_frames) const;
 
   rtc::scoped_refptr<InternalSource> get() const;

diff --git a/webrtc-sys/src/audio_track.cpp b/webrtc-sys/src/audio_track.cpp
@@ -156,8 +156,8 @@ void AudioTrackSource::set_audio_options(
 }
 
 void AudioTrackSource::on_captured_frame(rust::Slice<const int16_t> audio_data,
-                                         int sample_rate,
-                                         size_t number_of_channels,
+                                         uint32_t sample_rate,
+                                         uint32_t number_of_channels,
                                          size_t number_of_frames) const {
   source_->on_captured_frame(audio_data, sample_rate, number_of_channels,
                              number_of_frames);