feat: AudioResampler (#66)

webrtc-sys/.gitignore

@@ -1 +0,0 @@
-/cmake-build-debug

webrtc-sys/build.rs

@@ -149,6 +149,7 @@ fn main() {
         "src/video_frame_buffer.rs",
         "src/yuv_helper.rs",
         "src/helper.rs",
+        "src/audio_resampler.rs",
     ]);
 
     builder.files(&[
@@ -169,6 +170,7 @@ fn main() {
         "src/video_encoder_factory.cpp",
         "src/video_decoder_factory.cpp",
         "src/audio_device.cpp",
+        "src/audio_resampler.cpp",
     ]);
 
     for include in includes {

webrtc-sys/include/livekit/audio_resampler.h

@@ -0,0 +1,47 @@
+/*
+ * Copyright 2023 LiveKit
+ *
+ * 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.
+ */
+
+#pragma once
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "api/data_channel_interface.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "livekit/webrtc.h"
+#include "rust/cxx.h"
+
+namespace livekit {
+
+class AudioResampler {
+ public:
+  size_t remix_and_resample(const int16_t* src,
+                            size_t samples_per_channel,
+                            size_t num_channels,
+                            int sample_rate_hz,
+                            size_t dest_num_channels,
+                            int dest_sample_rate_hz);
+
+  const int16_t* data() const;
+
+ private:
+  webrtc::AudioFrame frame_;
+  webrtc::PushResampler<int16_t> resampler_;
+};
+
+std::unique_ptr<AudioResampler> create_audio_resampler();
+
+}  // namespace livekit

webrtc-sys/src/audio_resampler.cpp

@@ -0,0 +1,47 @@
+/*
+ * Copyright 2023 LiveKit
+ *
+ * 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.
+ */
+
+#include "livekit/audio_resampler.h"
+
+#include <memory>
+
+#include "audio/remix_resample.h"
+
+namespace livekit {
+
+size_t AudioResampler::remix_and_resample(const int16_t* src,
+                                          size_t samples_per_channel,
+                                          size_t num_channels,
+                                          int sample_rate,
+                                          size_t dest_num_channels,
+                                          int dest_sample_rate) {
+  frame_.num_channels_ = dest_num_channels;
+  frame_.sample_rate_hz_ = dest_sample_rate;
+  webrtc::voe::RemixAndResample(src, samples_per_channel, num_channels,
+                                sample_rate, &resampler_, &frame_);
+
+  return frame_.num_channels() * frame_.samples_per_channel();
+}
+
+const int16_t* AudioResampler::data() const {
+  return frame_.data();
+}
+
+std::unique_ptr<AudioResampler> create_audio_resampler() {
+  return std::make_unique<AudioResampler>();
+}
+
+}  // namespace livekit

webrtc-sys/src/audio_resampler.rs

@@ -0,0 +1,26 @@
+use crate::impl_thread_safety;
+
+#[cxx::bridge(namespace = "livekit")]
+pub mod ffi {
+    unsafe extern "C++" {
+        include!("livekit/audio_resampler.h");
+
+        type AudioResampler;
+
+        unsafe fn remix_and_resample(
+            self: Pin<&mut AudioResampler>,
+            src: *const i16,
+            samples_per_channel: usize,
+            num_channels: usize,
+            sample_rate: i32,
+            dst_num_channels: usize,
+            dst_sample_rate: i32,
+        ) -> usize;
+
+        unsafe fn data(self: &AudioResampler) -> *const i16;
+
+        fn create_audio_resampler() -> UniquePtr<AudioResampler>;
+    }
+}
+
+impl_thread_safety!(ffi::AudioResampler, Send + Sync);

webrtc-sys/src/lib.rs

@@ -1,3 +1,4 @@
+pub mod audio_resampler;
 pub mod candidate;
 pub mod data_channel;
 pub mod helper;