Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
index 2ba10d2b20759ded6c8efb91ae689981c7e72676..4bc0266b7d2c62287cc4b70035f548f0ed44d3e2 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
@@ -12,6 +12,7 @@ |
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |
#include "webrtc/common_types.h" |
+#include "webrtc/base/criticalsection.h" |
#include "webrtc/base/onetimeevent.h" |
#include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h" |
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
@@ -73,7 +74,7 @@ class RTPSenderAudio : public DTMFqueue { |
Clock* const _clock; |
RTPSender* const _rtpSender; |
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect; |
+ rtc::CriticalSection _sendAudioCritsect; |
uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect); |