Replace use of ASSERT in test code.

webrtc/api/rtcstatscollector_unittest.cc

 /*
  *  Copyright 2016 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.
  */
 
@@ skipping 669 matching lines @@
   expected_inbound_video_codec.payload_type = 3;
   expected_inbound_video_codec.codec = "video/H264";
   expected_inbound_video_codec.clock_rate = 1339;
 
   RTCCodecStats expected_outbound_video_codec(
       "RTCCodec_OutboundVideo_4", report->timestamp_us());
   expected_outbound_video_codec.payload_type = 4;
   expected_outbound_video_codec.codec = "video/VP8";
   expected_outbound_video_codec.clock_rate = 1340;
 
-  ASSERT(report->Get(expected_inbound_audio_codec.id()));
+  RTC_CHECK(report->Get(expected_inbound_audio_codec.id()));

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

I'm pretty sure you should be using ASSERT_TRUE here.

(You *can* use ASSERT_* here because it's the top level function of a
test---ASSERT_* sets some flag and then does a return---and you *should* use
ASSERT_* rather than CHECK because the former will fail just the current test,
whereas the latter will kill the entire process, possibly making other tests
collateral damage.)

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

Done. Applied to all places in this file.

   EXPECT_EQ(expected_inbound_audio_codec,
             report->Get(expected_inbound_audio_codec.id())->cast_to<
                   RTCCodecStats>());
 
-  ASSERT(report->Get(expected_outbound_audio_codec.id()));
+  RTC_CHECK(report->Get(expected_outbound_audio_codec.id()));
   EXPECT_EQ(expected_outbound_audio_codec,
             report->Get(expected_outbound_audio_codec.id())->cast_to<
                   RTCCodecStats>());
 
-  ASSERT(report->Get(expected_inbound_video_codec.id()));
+  RTC_CHECK(report->Get(expected_inbound_video_codec.id()));
   EXPECT_EQ(expected_inbound_video_codec,
             report->Get(expected_inbound_video_codec.id())->cast_to<
                   RTCCodecStats>());
 
-  ASSERT(report->Get(expected_outbound_video_codec.id()));
+  RTC_CHECK(report->Get(expected_outbound_video_codec.id()));
   EXPECT_EQ(expected_outbound_video_codec,
             report->Get(expected_outbound_video_codec.id())->cast_to<
                   RTCCodecStats>());
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
   std::unique_ptr<CertificateInfo> audio_local_certinfo =
       CreateFakeCertificateAndInfoFromDers(
           std::vector<std::string>({ "(local) audio" }));
   audio_local_certinfo = CreateFakeCertificateAndInfoFromDers(
@@ skipping 782 matching lines @@
   expected_audio.media_type = "audio";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_InboundAudio_42";
   expected_audio.packets_received = 2;
   expected_audio.bytes_received = 3;
   expected_audio.packets_lost = 42;
   expected_audio.jitter = 4.5;
   expected_audio.fraction_lost = 5.5;
 
-  ASSERT(report->Get(expected_audio.id()));
+  RTC_CHECK(report->Get(expected_audio.id()));
   const RTCInboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   ASSERT_TRUE(report->Get(*expected_audio.transport_id));
   ASSERT_TRUE(report->Get(*expected_audio.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
@@ skipping 58 matching lines @@
   expected_video.codec_id = "RTCCodec_InboundVideo_42";
   expected_video.fir_count = 5;
   expected_video.pli_count = 6;
   expected_video.nack_count = 7;
   expected_video.packets_received = 2;
   expected_video.bytes_received = 3;
   expected_video.packets_lost = 42;
   expected_video.fraction_lost = 4.5;
   expected_video.frames_decoded = 8;
 
-  ASSERT(report->Get(expected_video.id()));
+  RTC_CHECK(report->Get(expected_video.id()));
   const RTCInboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*video.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
@@ skipping 47 matching lines @@
   expected_audio.ssrc = "1";
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
   expected_audio.packets_sent = 2;
   expected_audio.bytes_sent = 3;
   expected_audio.round_trip_time = 4.5;
 
-  ASSERT(report->Get(expected_audio.id()));
+  RTC_CHECK(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   ASSERT_TRUE(report->Get(*expected_audio.transport_id));
   ASSERT_TRUE(report->Get(*expected_audio.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel();
@@ skipping 57 matching lines @@
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
   expected_video.packets_sent = 5;
   expected_video.bytes_sent = 6;
   expected_video.round_trip_time = 7.5;
   expected_video.frames_encoded = 8;
   expected_video.qp_sum = 16;
 
-  ASSERT(report->Get(expected_video.id()));
+  RTC_CHECK(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 
   ASSERT_TRUE(report->Get(*expected_video.transport_id));
   ASSERT_TRUE(report->Get(*expected_video.codec_id));
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
   MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel();
@@ skipping 62 matching lines @@
   expected_audio.ssrc = "1";
   expected_audio.is_remote = false;
   expected_audio.media_type = "audio";
   expected_audio.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
   expected_audio.packets_sent = 2;
   expected_audio.bytes_sent = 3;
   // |expected_audio.round_trip_time| should be undefined.
 
-  ASSERT(report->Get(expected_audio.id()));
+  RTC_CHECK(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
 
   RTCOutboundRTPStreamStats expected_video(
       "RTCOutboundRTPVideoStream_1", report->timestamp_us());
   expected_video.ssrc = "1";
   expected_video.is_remote = false;
   expected_video.media_type = "video";
   expected_video.transport_id = "RTCTransport_TransportName_" +
       rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
   expected_video.codec_id = "RTCCodec_OutboundVideo_42";
   expected_video.fir_count = 2;
   expected_video.pli_count = 3;
   expected_video.nack_count = 4;
   expected_video.packets_sent = 5;
   expected_video.bytes_sent = 6;
   expected_video.frames_encoded = 7;
   // |expected_video.round_trip_time| should be undefined.
   // |expected_video.qp_sum| should be undefined.
 
-  ASSERT(report->Get(expected_video.id()));
+  RTC_CHECK(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
 }
 
 TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
   std::unique_ptr<cricket::Candidate> rtp_local_candidate = CreateFakeCandidate(
       "42.42.42.42", 42, "protocol", cricket::LOCAL_PORT_TYPE, 42);
   std::unique_ptr<cricket::Candidate> rtp_remote_candidate =
       CreateFakeCandidate("42.42.42.42", 42, "protocol",
@@ skipping 162 matching lines @@
   rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
 };
 
 TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
   collector_->VerifyThreadUsageAndResultsMerging();
 }
 
 }  // namespace
 
 }  // namespace webrtc