webrtc M86: critical_section.h replaced by mutex.h

CoSMoSoftware · Jan 17, 2021 · fca8655 · fca8655
1 parent a8ca4c6
commit fca8655
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Showing 4 changed files with 7 additions and 8 deletions.
diff --git a/plugins/obs-outputs/AudioDeviceModuleWrapper.cpp b/plugins/obs-outputs/AudioDeviceModuleWrapper.cpp
@@ -43,10 +43,10 @@ bool AudioDeviceModuleWrapper::Initialized() const
 void AudioDeviceModuleWrapper::onIncomingData(uint8_t *data,
 					      size_t samples_per_channel)
 {
-	_critSect.Enter();
+	_critSect.Lock();
 	if (!audioTransport)
 		return;
-	_critSect.Leave();
+	_critSect.Unlock();
 
 	// Get audio
 	audio_t *audio = obs_get_audio();

diff --git a/plugins/obs-outputs/AudioDeviceModuleWrapper.h b/plugins/obs-outputs/AudioDeviceModuleWrapper.h
@@ -6,7 +6,7 @@
 #include <api/scoped_refptr.h>
 #include <modules/audio_device/include/audio_device_default.h>
 #include <rtc_base/checks.h>
-#include <rtc_base/critical_section.h>
+#include <rtc_base/synchronization/mutex.h>
 #include <rtc_base/ref_counted_object.h>
 
 #include <stdio.h>
@@ -83,7 +83,7 @@ class AudioDeviceModuleWrapper
 
 public:
 	bool _initialized;
-	rtc::CriticalSection _critSect;
+	webrtc::Mutex _critSect;
 	AudioTransport *audioTransport;
 	uint8_t pending[640 * 2 * 2];
 	size_t pendingLength;

diff --git a/plugins/obs-outputs/WebRTCStream.cpp b/plugins/obs-outputs/WebRTCStream.cpp
@@ -13,7 +13,6 @@
 #include "common_video/libyuv/include/webrtc_libyuv.h"
 #include "pc/rtc_stats_collector.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 #include <libyuv.h>
 
 #include <algorithm>
@@ -887,7 +886,7 @@ void WebRTCStream::getStats()
 
 rtc::scoped_refptr<const webrtc::RTCStatsReport> WebRTCStream::NewGetStats()
 {
-	rtc::CritScope lock(&crit_);
+	webrtc::MutexLock lock(&crit_);
 
 	if (nullptr == pc) {
 		return nullptr;

diff --git a/plugins/obs-outputs/WebRTCStream.h b/plugins/obs-outputs/WebRTCStream.h
@@ -27,7 +27,7 @@
 #include "api/scoped_refptr.h"
 #include "api/set_remote_description_observer_interface.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/critical_section.h"
+#include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/ref_counted_object.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/timestamp_aligner.h"
@@ -176,7 +176,7 @@ class WebRTCStream : public rtc::RefCountedObject<WebRTCStreamInterface> {
 
 	std::thread thread_closeAsync;
 
-	rtc::CriticalSection crit_;
+	webrtc::Mutex crit_;
 
 	// Audio Wrapper
 	rtc::scoped_refptr<AudioDeviceModuleWrapper> adm;