Expose WebRTC's audio_mixer

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::{apm, audio_resampler, frame_cryptor, yuv_helper};
+    pub use crate::imp::{apm, audio_mixer, audio_resampler, frame_cryptor, yuv_helper};
 }
 
 #[cfg(target_os = "android")]

libwebrtc/src/native/audio_mixer.rs

@@ -0,0 +1,108 @@
+// 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 crate::audio_frame::AudioFrame;
+use cxx::UniquePtr;
+use std::sync::Arc;
+use webrtc_sys::audio_mixer as sys;
+use webrtc_sys::audio_mixer::ffi;
+
+pub struct AudioMixer {
+    sys_handle: UniquePtr<ffi::AudioMixer>,
+}
+
+pub use ffi::AudioFrameInfo;
+
+pub trait AudioMixerSource {
+    fn ssrc(&self) -> i32;
+    fn preferred_sample_rate(&self) -> u32;
+    fn get_audio_frame_with_info<'a>(&self, target_sample_rate: u32) -> Option<AudioFrame>;
+}
+
+struct AudioMixerSourceImpl<T> {
+    inner: T,
+}
+impl<T: AudioMixerSource> sys::AudioMixerSource for AudioMixerSourceImpl<T> {
+    fn ssrc(&self) -> i32 {
+        self.inner.ssrc()
+    }
+
+    fn preferred_sample_rate(&self) -> i32 {
+        self.inner.preferred_sample_rate() as i32
+    }
+
+    fn get_audio_frame_with_info<'a>(
+        &self,
+        target_sample_rate: i32,
+        native_frame: sys::NativeAudioFrame,
+    ) -> AudioFrameInfo {
+        if let Some(frame) = self.inner.get_audio_frame_with_info(target_sample_rate as u32) {
+            let samples_count = (frame.sample_rate as usize / 100) as usize;
+            assert_eq!(
+                frame.sample_rate, target_sample_rate as u32,
+                "sample rate must match target_sample_rate"
+            );
+            assert_eq!(
+                frame.samples_per_channel as usize, samples_count,
+                "frame must contain 10ms of samples"
+            );
+            assert_eq!(
+                frame.data.len(),
+                samples_count * frame.num_channels as usize,
+                "slice must contain 10ms of samples"
+            );
+
+            unsafe {
+                native_frame.update_frame(
+                    0,
+                    frame.data.as_ptr(),
+                    frame.samples_per_channel as usize,
+                    frame.sample_rate as i32,
+                    frame.num_channels as usize,
+                );
+            }
+            return ffi::AudioFrameInfo::Normal;
+        } else {
+            return ffi::AudioFrameInfo::Muted;
+        }
+    }
+}
+
+impl AudioMixer {
+    pub fn new() -> Self {
+        let sys_handle = ffi::create_audio_mixer();
+        Self { sys_handle }
+    }
+
+    pub fn add_source(&mut self, source: impl AudioMixerSource + 'static) {
+        let source_impl = AudioMixerSourceImpl { inner: source };
+        let wrapper = Box::new(sys::AudioMixerSourceWrapper::new(Arc::new(source_impl)));
+        unsafe {
+            self.sys_handle.pin_mut().add_source(wrapper);
+        }
+    }
+
+    pub fn remove_source(&mut self, ssrc: i32) {
+        unsafe {
+            self.sys_handle.pin_mut().remove_source(ssrc);
+        }
+    }
+
+    pub fn mix(&mut self, num_channels: usize) -> &[i16] {
+        unsafe {
+            let len = self.sys_handle.pin_mut().mix(num_channels);
+            std::slice::from_raw_parts(self.sys_handle.data(), len)
+        }
+    }
+}

libwebrtc/src/native/mod.rs

@@ -16,6 +16,7 @@
 pub mod android;
 pub mod apm;
 pub mod audio_resampler;
+pub mod audio_mixer;
 pub mod audio_source;
 pub mod audio_stream;
 pub mod audio_track;

webrtc-sys/build.rs

@@ -49,6 +49,7 @@ fn main() {
         "src/android.rs",
         "src/prohibit_libsrtp_initialization.rs",
         "src/apm.rs",
+        "src/audio_mixer.rs",
     ]);
 
     builder.files(&[
@@ -77,6 +78,7 @@ fn main() {
         "src/global_task_queue.cpp",
         "src/prohibit_libsrtp_initialization.cpp",
         "src/apm.cpp",
+        "src/audio_mixer.cpp",
     ]);
 
     let webrtc_dir = webrtc_sys_build::webrtc_dir();

webrtc-sys/include/livekit/audio_mixer.h

@@ -0,0 +1,81 @@
+/*
+ * 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/scoped_refptr.h"
+#include "api/audio/audio_mixer.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/audio_buffer.h"
+
+#include "rust/cxx.h"
+
+namespace livekit {
+class AudioMixer;
+class NativeAudioFrame;
+}  // namespace livekit
+
+#include "webrtc-sys/src/audio_mixer.rs.h"
+
+namespace livekit {
+
+class NativeAudioFrame {
+public:
+    NativeAudioFrame(webrtc::AudioFrame* frame) : frame_(frame) {}
+    void update_frame(uint32_t timestamp, const int16_t* data, size_t samples_per_channel, int sample_rate_hz, size_t num_channels);
+private:
+    webrtc::AudioFrame* frame_;
+};
+
+class AudioMixerSource: public webrtc::AudioMixer::Source {
+public:
+    AudioMixerSource(rust::Box<AudioMixerSourceWrapper> source);
+
+    AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, webrtc::AudioFrame* audio_frame) override;
+
+    int Ssrc() const override;
+
+    int PreferredSampleRate() const override;
+
+    ~AudioMixerSource() {}
+
+  private:
+    rust::Box<AudioMixerSourceWrapper> source_;
+};
+
+class AudioMixer {
+ public:
+  AudioMixer();
+
+  void add_source(rust::Box<AudioMixerSourceWrapper> source);
+
+  void remove_source(int ssrc);
+
+  size_t mix(size_t num_channels);
+  const int16_t* data() const;
+
+ private:
+  webrtc::AudioFrame frame_;
+  std::vector<std::shared_ptr<AudioMixerSource>> sources_;
+  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer_;
+};
+
+std::unique_ptr<AudioMixer> create_audio_mixer();
+
+
+}  // namespace livekit

webrtc-sys/src/audio_mixer.cpp

@@ -0,0 +1,78 @@
+#include "livekit/audio_mixer.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio/audio_mixer.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc-sys/src/audio_mixer.rs.h"
+
+#include <memory>
+#include <iostream>
+
+namespace livekit {
+
+AudioMixer::AudioMixer() {
+    audio_mixer_ = webrtc::AudioMixerImpl::Create();
+}
+
+void AudioMixer::add_source(rust::Box<AudioMixerSourceWrapper> source) {
+    auto native_source = std::make_shared<AudioMixerSource>(std::move(source));
+
+    audio_mixer_->AddSource(native_source.get());
+    sources_.push_back(native_source);
+}
+
+void AudioMixer::remove_source(int source_ssrc) {
+    auto it = std::find_if(sources_.begin(), sources_.end(),
+        [source_ssrc](const auto& s) { return s->Ssrc() == source_ssrc; });
+
+    if (it != sources_.end()) {
+        audio_mixer_->RemoveSource(it->get());
+        sources_.erase(it);
+    }
+}
+
+size_t AudioMixer::mix(size_t number_of_channels) {
+    audio_mixer_->Mix(number_of_channels, &frame_);
+    return frame_.num_channels() * frame_.samples_per_channel() * sizeof(int16_t);
+}
+
+const int16_t* AudioMixer::data() const {
+  return frame_.data();
+}
+
+std::unique_ptr<AudioMixer> create_audio_mixer() {
+  return std::make_unique<AudioMixer>();
+}
+
+AudioMixerSource::AudioMixerSource(rust::Box<AudioMixerSourceWrapper> source) : source_(std::move(source)) {
+}
+
+int AudioMixerSource::Ssrc() const {
+    return source_->ssrc();
+}
+
+int AudioMixerSource::PreferredSampleRate() const {
+    return source_->preferred_sample_rate();
+}
+
+webrtc::AudioMixer::Source::AudioFrameInfo AudioMixerSource::GetAudioFrameWithInfo(int sample_rate, webrtc::AudioFrame* audio_frame) {
+    NativeAudioFrame frame(audio_frame);
+
+    livekit::AudioFrameInfo result =  source_->get_audio_frame_with_info(sample_rate, frame);
+
+    if (result == livekit::AudioFrameInfo::Normal) {
+        return webrtc::AudioMixer::Source::AudioFrameInfo::kNormal;
+    } else if (result == livekit::AudioFrameInfo::Muted) {
+        return webrtc::AudioMixer::Source::AudioFrameInfo::kMuted;
+    } else {
+        return webrtc::AudioMixer::Source::AudioFrameInfo::kError;
+    }
+}
+
+void NativeAudioFrame::update_frame(uint32_t timestamp, const int16_t* data,
+    size_t samples_per_channel, int sample_rate_hz, size_t num_channels) {
+    frame_->UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
+        webrtc::AudioFrame::SpeechType::kNormalSpeech, webrtc::AudioFrame::VADActivity::kVadUnknown,
+        num_channels);
+}
+
+}  // namespace livekit