talksik/client-sdk-rust
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity

fix: datachannel deadlocks & dispose crashes (#87)

Browse Source 
- Fix datachannel deadlock on dispose
- Fix audio device being disposed on the wrong thread
- Allow multiple RtcRuntime to be created (Useful for unit tests where we use multiple PeerConnectionFactory)
This commit is contained in:
Théo Monnom
2023-06-10 18:49:00 +02:00
committed by GitHub
parent ebbd93b2d8
commit 33dfcb27b0
9 changed files with 90 additions and 39 deletions
Download Patch File Download Diff File Expand all files Collapse all files
webrtc-sys/src/webrtc.cpp
+44 -8
View File
@@ -16,6 +16,9 @@
#include "livekit/webrtc.h"
#include <algorithm>
#include <atomic>
#include <iostream>
#include <memory>
#include "livekit/audio_track.h"
@@ -27,10 +30,34 @@
#include "rtc_base/logging.h"
#include "rtc_base/synchronization/mutex.h"
#ifdef WEBRTC_WIN
#include "rtc_base/win32.h"
#endif
namespace livekit {
static webrtc::Mutex g_mutex{};
// Can't be atomic, we're using a Mutex because we need to wait for the
// execution of the first init
static uint32_t g_release_counter(0);
RtcRuntime::RtcRuntime() {
RTC_LOG(LS_INFO) << "RtcRuntime()";
RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
rtc::LogMessage::LogToDebug(rtc::LS_INFO);
RTC_LOG(LS_VERBOSE) << "RtcRuntime()";
{
// Not the best way to do it...
webrtc::MutexLock lock(&g_mutex);
if (g_release_counter == 0) {
RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
#ifdef WEBRTC_WIN
WSADATA data;
WSAStartup(MAKEWORD(1, 0), &data);
#endif
}
g_release_counter++;
}
network_thread_ = rtc::Thread::CreateWithSocketServer();
network_thread_->SetName("network_thread", &network_thread_);
@@ -44,14 +71,23 @@ RtcRuntime::RtcRuntime() {
}
RtcRuntime::~RtcRuntime() {
RTC_LOG(LS_INFO) << "~RtcRuntime()";
RTC_LOG(LS_VERBOSE) << "~RtcRuntime()";
rtc::ThreadManager::Instance()->SetCurrentThread(nullptr);
RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
worker_thread_->Stop();
signaling_thread_->Stop();
network_thread_->Stop();
worker_thread_->Quit();
signaling_thread_->Quit();
network_thread_->Quit();
{
webrtc::MutexLock lock(&g_mutex);
g_release_counter--;
if (g_release_counter == 0) {
RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
#ifdef WEBRTC_WIN
WSACleanup();
#endif
}
}
}
rtc::Thread* RtcRuntime::network_thread() const {