commit 6287280d64ece74c22a0d8dde9e20bd23e68374e
author Markus Handell <handellm@webrtc.org> Mon Jul 06 13:15:07 2020
committer Commit Bot <commit-bot@chromium.org> Mon Jul 06 14:21:38 2020
tree e8f7721d86e4cbe029375bcd07d1d2a363bc5509
parent afd1dcbde0407f4b104cad73ea82dead822d7e70

Migrate audio/ to use webrtc::Mutex

Bug: webrtc:11567
Change-Id: Ic6a753f09aafb508690f4b8dadd4c99433fcfeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176741
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31635}

audio/audio_transport_impl.h

diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index 2d9b4cf..1643a29 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -20,7 +20,7 @@
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/typing_detection.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
+#include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace webrtc {
@@ -71,7 +71,7 @@
   AudioProcessing* audio_processing_ = nullptr;
 
   // Capture side.
-  rtc::CriticalSection capture_lock_;
+  mutable Mutex capture_lock_;
   std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
   int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
   size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;