Replace use of ASSERT in test code.

webrtc/api/statscollector_unittest.cc

 /*
  *  Copyright 2014 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.
  */
 
 #include <stdio.h>
 
 #include <algorithm>
 #include <memory>
 
 #include "webrtc/api/statscollector.h"
 
 #include "webrtc/api/mediastream.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/mediastreamtrack.h"
 #include "webrtc/api/peerconnection.h"
 #include "webrtc/api/peerconnectionfactory.h"
 #include "webrtc/api/test/fakedatachannelprovider.h"
 #include "webrtc/api/test/fakevideotracksource.h"
 #include "webrtc/api/test/mock_peerconnection.h"
 #include "webrtc/api/test/mock_webrtcsession.h"
 #include "webrtc/api/videotrack.h"
 #include "webrtc/base/base64.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/fakesslidentity.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/media/base/fakemediaengine.h"
 #include "webrtc/media/base/test/mock_mediachannel.h"
 #include "webrtc/p2p/base/faketransportcontroller.h"
 #include "webrtc/pc/channelmanager.h"
 #include "webrtc/test/gmock.h"
@@ skipping 563 matching lines @@
       webrtc::StatsCollector* stats,
       cricket::VoiceChannel* voice_channel,
       const std::string& vc_name,
       MockVoiceMediaChannel* media_channel,
       cricket::VoiceSenderInfo* voice_sender_info,
       cricket::VoiceReceiverInfo* voice_receiver_info,
       cricket::VoiceMediaInfo* stats_read,
       StatsReports* reports) {
     // A track can't have both sender report and recv report at the same time
     // for now, this might change in the future though.
-    ASSERT((voice_sender_info == NULL) ^ (voice_receiver_info == NULL));
+    RTC_CHECK((voice_sender_info == NULL) ^ (voice_receiver_info == NULL));

[kwiberg-webrtc, 2017/01/17 09:39:25]

You may not be able to use ASSERT_* here. Will EXPECT_* do? (CHECK is a bad
choice for reasons stated in an earlier comment.)

[nisse-webrtc, 2017/01/17 12:19:21]

EXPECT_TRUE seems to work.
 
But I think RTC_CHECK may be appropriate here. This is a helper function used in
4 places in this file, and the RTC_CHECK enforces that those callers use it
correctly. I.e., if this ever fails, that's a bug in the test code, not in the
code under test.

Your call.

[kwiberg-webrtc, 2017/01/17 13:52:47]

Sounds reasonable.

 
     // Instruct the session to return stats containing the transport channel.
     InitSessionStats(vc_name);
     EXPECT_CALL(session_, GetStats(_)).WillRepeatedly(Invoke(
         [this](const ChannelNamePairs&) {
           return std::unique_ptr<SessionStats>(
               new SessionStats(session_stats_));
         }));
 
     // Constructs an ssrc stats update.
@@ skipping 676 matching lines @@
   const char local_ip[] = "192.168.0.1";
   const int remote_port = 2001;
   const char remote_ip[] = "192.168.0.2";
 
   rtc::SocketAddress local_address(local_ip, local_port);
   rtc::SocketAddress remote_address(remote_ip, remote_port);
   rtc::AdapterType network_type = rtc::ADAPTER_TYPE_ETHERNET;
   uint32_t priority = 1000;
 
   cricket::Candidate c;
-  ASSERT(c.id().length() > 0);
+  RTC_CHECK(c.id().length() > 0);

[kwiberg-webrtc, 2017/01/17 09:39:25]

ASSERT_* or EXPECT_*

[nisse-webrtc, 2017/01/17 12:19:21]

Changed to EXPECT_GT.

   c.set_type(cricket::LOCAL_PORT_TYPE);
   c.set_protocol(cricket::UDP_PROTOCOL_NAME);
   c.set_address(local_address);
   c.set_priority(priority);
   c.set_network_type(network_type);
   std::string report_id = AddCandidateReport(&stats, c, true)->id()->ToString();
   EXPECT_EQ("Cand-" + c.id(), report_id);
 
   c = cricket::Candidate();
-  ASSERT(c.id().length() > 0);
+  RTC_CHECK(c.id().length() > 0);
   c.set_type(cricket::PRFLX_PORT_TYPE);
   c.set_protocol(cricket::UDP_PROTOCOL_NAME);
   c.set_address(remote_address);
   c.set_priority(priority);
   c.set_network_type(network_type);
   report_id = AddCandidateReport(&stats, c, false)->id()->ToString();
   EXPECT_EQ("Cand-" + c.id(), report_id);
 
   stats.GetStats(NULL, &reports);
 
@@ skipping 687 matching lines @@
   EXPECT_CALL(*media_channel, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
   EXPECT_EQ(rtc::ToString(video_receiver_info.frames_decoded),
             ExtractSsrcStatsValue(reports,
                                   StatsReport::kStatsValueNameFramesDecoded));
 }
 
 }  // namespace webrtc