1 files changed, 7 insertions, 6 deletions
diff --git a/audio/voip/audio_egress.h b/audio/voip/audio_egress.h
index e5632cde32..8ec048f915 100644
--- a/audio/voip/audio_egress.h
+++ b/audio/voip/audio_egress.h
@@ -20,8 +20,9 @@
 #include "call/audio_sender.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/rtp_rtcp/include/report_block_data.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
 #include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
 #include "rtc_base/time_utils.h"
@@ -43,7 +44,7 @@ namespace webrtc {
 // smaller footprint.
 class AudioEgress : public AudioSender, public AudioPacketizationCallback {
  public:
-  AudioEgress(RtpRtcp* rtp_rtcp,
+  AudioEgress(RtpRtcpInterface* rtp_rtcp,
               Clock* clock,
               TaskQueueFactory* task_queue_factory);
   ~AudioEgress() override;
@@ -72,7 +73,7 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback {
   // Retrieve current encoder format info. This returns encoder format set
   // by SetEncoder() and if encoder is not set, this will return nullopt.
   absl::optional<SdpAudioFormat> GetEncoderFormat() const {
-    rtc::CritScope lock(&lock_);
+    MutexLock lock(&lock_);
     return encoder_format_;
   }
 
@@ -99,17 +100,17 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback {
 
  private:
   void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
-    rtc::CritScope lock(&lock_);
+    MutexLock lock(&lock_);
     encoder_format_ = encoder_format;
   }
 
-  rtc::CriticalSection lock_;
+  mutable Mutex lock_;
 
   // Current encoder format selected by caller.
   absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
 
   // Synchronization is handled internally by RtpRtcp.
-  RtpRtcp* const rtp_rtcp_;
+  RtpRtcpInterface* const rtp_rtcp_;
 
   // Synchronization is handled internally by RTPSenderAudio.
   RTPSenderAudio rtp_sender_audio_;