Replace use of ASSERT in test code.

webrtc/api/rtcstatscollector_unittest.cc

Index: webrtc/api/rtcstatscollector_unittest.cc
diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index a9426622fb391cddbe54edf32b9ef4a9724a235e..6e0f340b8a89d9b62ed0cb8ba6173255cfa74ec6 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -687,22 +687,22 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
   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>());
@@ -1505,7 +1505,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
   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);
@@ -1584,7 +1584,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
   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);
@@ -1652,7 +1652,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   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);
@@ -1730,7 +1730,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
   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);
@@ -1813,7 +1813,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
   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);
@@ -1835,7 +1835,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Default) {
   // |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);