From 983546767adc8f75fc971dd058eafcb61e2a1856 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Th=C3=A9o=20Monnom?= <theo.monnom@outlook.com>
Date: Mon, 19 Jun 2023 01:40:51 +0200
Subject: [PATCH] fix: use native tls roots on ffi (#91)

---
 livekit-ffi/Cargo.toml                    | 2 +-
 webrtc-sys/include/livekit/audio_device.h | 8 ++++----
 webrtc-sys/src/audio_device.cpp           | 2 ++
 webrtc-sys/src/data_channel.cpp           | 5 ++++-
 4 files changed, 11 insertions(+), 6 deletions(-)

diff --git a/livekit-ffi/Cargo.toml b/livekit-ffi/Cargo.toml
index 21f76d9..2ed4fbb 100644
--- a/livekit-ffi/Cargo.toml
+++ b/livekit-ffi/Cargo.toml
@@ -7,7 +7,7 @@ description = "LiveKit interface for foreign languages"
 repository = "https://github.com/livekit/client-sdk-rust" 
 
 [features]
-default = ["__rustls-tls"]
+default = ["rustls-tls-native-roots"]
 native-tls = ["livekit/native-tls"]
 native-tls-vendored = ["livekit/native-tls-vendored"]
 rustls-tls-native-roots = ["livekit/rustls-tls-native-roots"]
diff --git a/webrtc-sys/include/livekit/audio_device.h b/webrtc-sys/include/livekit/audio_device.h
index 135e178..788fb63 100644
--- a/webrtc-sys/include/livekit/audio_device.h
+++ b/webrtc-sys/include/livekit/audio_device.h
@@ -118,12 +118,12 @@ class AudioDevice : public webrtc::AudioDeviceModule {
 
  private:
   mutable webrtc::Mutex mutex_;
-  webrtc::TaskQueueFactory* task_queue_factory_;
+  std::vector<int16_t> data_;
   std::unique_ptr<rtc::TaskQueue> audio_queue_;
   webrtc::RepeatingTaskHandle audio_task_;
-  std::vector<int16_t> data_;
   webrtc::AudioTransport* audio_transport_;
-  std::atomic<bool> playing_{false};
-  std::atomic<bool> initialized_{false};
+  webrtc::TaskQueueFactory* task_queue_factory_;
+  bool playing_{false};
+  bool initialized_{false};
 };
 }  // namespace livekit
diff --git a/webrtc-sys/src/audio_device.cpp b/webrtc-sys/src/audio_device.cpp
index 392f73a..18e63d6 100644
--- a/webrtc-sys/src/audio_device.cpp
+++ b/webrtc-sys/src/audio_device.cpp
@@ -150,11 +150,13 @@ bool AudioDevice::RecordingIsInitialized() const {
 }
 
 int32_t AudioDevice::StartPlayout() {
+  webrtc::MutexLock lock(&mutex_);
   playing_ = true;
   return 0;
 }
 
 int32_t AudioDevice::StopPlayout() {
+  webrtc::MutexLock lock(&mutex_);
   playing_ = false;
   return 0;
 }
diff --git a/webrtc-sys/src/data_channel.cpp b/webrtc-sys/src/data_channel.cpp
index 0eb1bcb..047af94 100644
--- a/webrtc-sys/src/data_channel.cpp
+++ b/webrtc-sys/src/data_channel.cpp
@@ -45,9 +45,12 @@ webrtc::DataChannelInit to_native_data_channel_init(DataChannelInit init) {
 DataChannel::DataChannel(
     std::shared_ptr<RtcRuntime> rtc_runtime,
     rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel)
-    : rtc_runtime_(rtc_runtime), data_channel_(std::move(data_channel)) {}
+    : rtc_runtime_(rtc_runtime), data_channel_(std::move(data_channel)) {
+  RTC_LOG(LS_VERBOSE) << "DataChannel::DataChannel()";
+}
 
 DataChannel::~DataChannel() {
+  RTC_LOG(LS_VERBOSE) << "DataChannel::~DataChannel()";
   unregister_observer();
 }