add AudioProcessingModule

libwebrtc/src/lib.rs

@@ -66,7 +66,7 @@ pub mod video_track;
 pub mod native {
     pub use webrtc_sys::webrtc::ffi::create_random_uuid;
 
-    pub use crate::imp::{audio_resampler, frame_cryptor, yuv_helper};
+    pub use crate::imp::{aec, audio_resampler, frame_cryptor, yuv_helper};
 }
 
 #[cfg(target_os = "android")]

libwebrtc/src/native/aec.rs

@@ -0,0 +1,51 @@
+// Copyright 2023 LiveKit, Inc.
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+use cxx::UniquePtr;
+use webrtc_sys::aec::ffi as sys_aec;
+
+pub struct Aec {
+    sys_handle: UniquePtr<sys_aec::Aec>,
+    sample_rate: i32,
+    num_channels: i32,
+}
+
+impl Aec {
+    pub fn new(sample_rate: i32, num_channels: i32) -> Self {
+        Self {
+            sys_handle: sys_aec::create_aec(sample_rate, num_channels),
+            sample_rate,
+            num_channels,
+        }
+    }
+
+    pub fn cancel_echo(&mut self, capture: &mut [i16], render: &[i16]) {
+        let required_samples = (self.sample_rate as usize / 100) * self.num_channels as usize;
+        assert_eq!(
+            capture.len(),
+            required_samples,
+            "Capture slice must have 10ms worth of samples"
+        );
+        assert_eq!(render.len(), required_samples, "Render slice must have 10ms worth of samples");
+
+        unsafe {
+            self.sys_handle.pin_mut().cancel_echo(
+                capture.as_mut_ptr(),
+                capture.len(),
+                render.as_ptr(),
+                render.len(),
+            );
+        }
+    }
+}

libwebrtc/src/native/mod.rs

@@ -12,6 +12,7 @@
 // See the License for the specific language governing permissions and
 // limitations under the License.
 
+pub mod aec;
 #[cfg(target_os = "android")]
 pub mod android;
 pub mod audio_resampler;

livekit-ffi/protocol/audio_frame.proto

@@ -87,6 +87,28 @@ message RemixAndResampleResponse {
   required OwnedAudioFrameBuffer buffer = 1;
 }
 
+// AEC
+
+
+message NewAecRequest {
+  required uint32 sample_rate = 1;
+  required uint32 num_channels = 2;
+}
+message NewAecResponse {
+  required OwnedAec aec = 1;
+}
+
+message CancelEchoRequest {
+  required uint64 aec_handle = 1;
+  required uint64 cap_ptr = 2; // *mut i16
+  required uint32 cap_size = 3; // in bytes
+  required uint64 rend_ptr = 4; // *const i16
+  required uint32 rend_size = 5;
+}
+
+message CancelEchoResponse {
+  optional string error = 3;
+}
 
 // New resampler using SoX (much better quality)
 
@@ -237,6 +259,13 @@ message OwnedAudioResampler {
 }
 
 
+//
+// AEC
+//
+
+message OwnedAec {
+  required FfiOwnedHandle handle = 1;
+}
 
 //
 // Sox AudioResampler

livekit-ffi/protocol/ffi.proto

@@ -120,7 +120,10 @@ message FfiRequest {
     // Data Channel
     SetDataChannelBufferedAmountLowThresholdRequest set_data_channel_buffered_amount_low_threshold = 47;
 
-    // NEXT_ID: 49
+    NewAecRequest new_aec = 49;
+    CancelEchoRequest cancel_echo = 50;
+
+    // NEXT_ID: 51
   }
 }
 
@@ -188,7 +191,10 @@ message FfiResponse {
     // Data Channel
     SetDataChannelBufferedAmountLowThresholdResponse set_data_channel_buffered_amount_low_threshold = 46;
 
-    // NEXT_ID: 48
+    NewAecResponse new_aec = 48;
+    CancelEchoResponse cancel_echo = 49;
+
+    // NEXT_ID: 50
   }
 }