cleanup: webrtc-sys & fix RtcRuntime disposing crashes (#81)
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/*
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* Copyright 2023 LiveKit
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*
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* Licensed under the Apache License, Version 2.0 (the “License”);
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an “AS IS” BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#pragma once
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#include <memory>
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#include "livekit/helper.h"
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#include "livekit/media_stream_track.h"
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#include "livekit/webrtc.h"
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#include "pc/local_audio_source.h"
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#include "rtc_base/synchronization/mutex.h"
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#include "rust/cxx.h"
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namespace livekit {
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class AudioTrack;
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class NativeAudioSink;
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class AudioTrackSource;
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} // namespace livekit
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#include "webrtc-sys/src/audio_track.rs.h"
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namespace livekit {
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class AudioTrack : public MediaStreamTrack {
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private:
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friend RtcRuntime;
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AudioTrack(std::shared_ptr<RtcRuntime> rtc_runtime,
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rtc::scoped_refptr<webrtc::AudioTrackInterface> track);
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public:
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~AudioTrack();
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void add_sink(const std::shared_ptr<NativeAudioSink>& sink) const;
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void remove_sink(const std::shared_ptr<NativeAudioSink>& sink) const;
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private:
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webrtc::AudioTrackInterface* track() const {
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return static_cast<webrtc::AudioTrackInterface*>(track_.get());
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}
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mutable webrtc::Mutex mutex_;
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// Same for VideoTrack:
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// Keep a strong reference to the added sinks, so we don't need to
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// manage the lifetime safety on the Rust side
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mutable std::vector<std::shared_ptr<NativeAudioSink>> sinks_;
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};
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class NativeAudioSink : public webrtc::AudioTrackSinkInterface {
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public:
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explicit NativeAudioSink(rust::Box<AudioSinkWrapper> observer);
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void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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private:
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rust::Box<AudioSinkWrapper> observer_;
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};
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std::shared_ptr<NativeAudioSink> new_native_audio_sink(
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rust::Box<AudioSinkWrapper> observer);
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class AudioTrackSource {
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class InternalSource : public webrtc::LocalAudioSource {
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public:
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InternalSource();
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SourceState state() const override;
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bool remote() const override;
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const cricket::AudioOptions options() const override;
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void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
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void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
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// AudioFrame should always contain 10 ms worth of data (see index.md of
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// acm)
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void on_captured_frame(rust::Slice<const int16_t> audio_data,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames);
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private:
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webrtc::Mutex mutex_;
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std::vector<webrtc::AudioTrackSinkInterface*> sinks_;
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cricket::AudioOptions options_{};
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};
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public:
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AudioTrackSource();
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void on_captured_frame(rust::Slice<const int16_t> audio_data,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) const;
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rtc::scoped_refptr<InternalSource> get() const;
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private:
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rtc::scoped_refptr<InternalSource> source_;
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};
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std::shared_ptr<AudioTrackSource> new_audio_track_source();
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static std::shared_ptr<MediaStreamTrack> audio_to_media(
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std::shared_ptr<AudioTrack> track) {
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return track;
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}
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static std::shared_ptr<AudioTrack> media_to_audio(
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std::shared_ptr<MediaStreamTrack> track) {
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return std::static_pointer_cast<AudioTrack>(track);
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}
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static std::shared_ptr<AudioTrack> _shared_audio_track() {
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return nullptr; // Ignore
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}
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} // namespace livekit
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