webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h - Issue 1877253002: Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module

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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

Issue 1877253002: Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: git cl format dtmf_queue.cc Created 4 years, 8 months ago

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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);

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