Replace ASSERT by RTC_DCHECK in all non-test code.

webrtc/api/rtpsender.cc

Index: webrtc/api/rtpsender.cc
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc
index bb5526a497be68c77a4608643e0570178d3cfd9d..b2e246120f03e2a9a500b311522252079d6743de 100644
--- a/webrtc/api/rtpsender.cc
+++ b/webrtc/api/rtpsender.cc
@@ -12,6 +12,7 @@
 
 #include "webrtc/api/localaudiosource.h"
 #include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/trace_event.h"
 
@@ -39,7 +40,7 @@ void LocalAudioSinkAdapter::OnData(const void* audio_data,
 
 void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
   rtc::CritScope lock(&lock_);
-  ASSERT(!sink || !sink_);
+  RTC_DCHECK(!sink || !sink_);
   sink_ = sink;
 }