Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I03b78bd2e411e9bcca199f85e4457511826cd17e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176745
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31649}
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index 78129e5..7a8829e 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -15,7 +15,7 @@
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/include/module_common_types.h"
-#include "rtc_base/critical_section.h"
+#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
@@ -88,7 +88,7 @@
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
uint8_t _payloadData[60 * 32 * 2 * 2];
- rtc::CriticalSection _channelCritSect;
+ Mutex _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
int16_t _lastPayloadType;