webrtc/modules/rtp_rtcp/source/rtp_receiver_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_receiver_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_receiver_audio.h

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

index f10613bc887748566dc030c53ff937c0a9ae7a30..bec1578e798ad6fe4375b3a188c1f4b7de58e95b 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

@@ -14,7 +14,6 @@

#include <set>

#include "webrtc/base/onetimeevent.h"

-#include "webrtc/base/scoped_ptr.h"

#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"

#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"

#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"

@@ -23,8 +22,6 @@

namespace webrtc {

-class CriticalSectionWrapper;

// Handles audio RTP packets. This class is thread-safe.

class RTPReceiverAudio : public RTPReceiverStrategy,

public TelephoneEventHandler {

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