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Migrate leftovers in media/ and modules/ to webrtc::Mutex.

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Bug: webrtc:11567
Change-Id: Id40a53fcec6cba1cd5af70422291ba46b0a6da8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178905
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31694}
This commit is contained in:
Markus Handell
2020-07-09 01:34:42 +02:00
committed by Commit Bot
parent 85585f4a52
commit f7303e6486
8 changed files with 143 additions and 164 deletions
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4
modules/rtp_rtcp/source/rtp_rtcp_impl.h
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@ -33,8 +33,8 @@
#include "modules/rtp_rtcp/source/rtp_packet_history.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
@ -352,7 +352,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
RtcpRttStats* const rtt_stats_;
// The processed RTT from RtcpRttStats.
rtc::CriticalSection critical_section_rtt_;
mutable Mutex mutex_rtt_;
int64_t rtt_ms_;
};