Replace use of ASSERT in test code.

webrtc/api/test/mockpeerconnectionobservers.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 // This file contains mock implementations of observers used in PeerConnection.
 
 #ifndef WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
 #define WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
 
 #include <memory>
 #include <string>
 
 #include "webrtc/api/datachannelinterface.h"
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
 class MockCreateSessionDescriptionObserver
     : public webrtc::CreateSessionDescriptionObserver {
  public:
   MockCreateSessionDescriptionObserver()
       : called_(false),
         result_(false) {}
   virtual ~MockCreateSessionDescriptionObserver() {}
@@ skipping 72 matching lines @@
   DataChannelInterface::DataState state_;
   std::vector<std::string> messages_;
 };
 
 class MockStatsObserver : public webrtc::StatsObserver {
  public:
   MockStatsObserver() : called_(false), stats_() {}
   virtual ~MockStatsObserver() {}
 
   virtual void OnComplete(const StatsReports& reports) {
-    ASSERT(!called_);
+    RTC_CHECK(!called_);
     called_ = true;
     stats_.Clear();
     stats_.number_of_reports = reports.size();
     for (const auto* r : reports) {
       if (r->type() == StatsReport::kStatsReportTypeSsrc) {
         stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel,
             &stats_.audio_output_level);
         GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel,
             &stats_.audio_input_level);
@@ skipping 13 matching lines @@
             &stats_.srtp_cipher);
       }
     }
   }
 
   bool called() const { return called_; }
   size_t number_of_reports() const { return stats_.number_of_reports; }
   double timestamp() const { return stats_.timestamp; }
 
   int AudioOutputLevel() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.audio_output_level;
   }
 
   int AudioInputLevel() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.audio_input_level;
   }
 
   int BytesReceived() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.bytes_received;
   }
 
   int BytesSent() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.bytes_sent;
   }
 
   int AvailableReceiveBandwidth() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.available_receive_bandwidth;
   }
 
   std::string DtlsCipher() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.dtls_cipher;
   }
 
   std::string SrtpCipher() const {
-    ASSERT(called_);
+    RTC_CHECK(called_);
     return stats_.srtp_cipher;
   }
 
  private:
   bool GetIntValue(const StatsReport* report,
                    StatsReport::StatsValueName name,
                    int* value) {
     const StatsReport::Value* v = report->FindValue(name);
     if (v) {
       // TODO(tommi): We should really just be using an int here :-/
@@ skipping 33 matching lines @@
     int bytes_sent;
     int available_receive_bandwidth;
     std::string dtls_cipher;
     std::string srtp_cipher;
   } stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_