commit c8ee882753f5240018445a1ce5761fd1c3196b4e
author nisse <nisse@webrtc.org> Wed Jan 18 07:20:55 2017 -0800
committer Commit bot <commit-bot@chromium.org> Wed Jan 18 15:20:55 2017 +0000
tree d52a1ecd895d0d5fe817e3c8963cf00f0f2e0f15
parent f20dd0014d1cfc8a2e859a9e177e7fe2b21274ca

Replace use of ASSERT in test code.

In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.

In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using

  git grep -l ' ASSERT(' | grep -v common.h | \
    xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}

webrtc/api/rtcstatscollector_unittest.cc

diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc
index f8efb24..0ba9abf 100644
--- a/webrtc/api/rtcstatscollector_unittest.cc
+++ b/webrtc/api/rtcstatscollector_unittest.cc
@@ -790,22 +790,22 @@
   expected_outbound_video_codec.codec = "video/VP8";
   expected_outbound_video_codec.clock_rate = 1340;
 
-  ASSERT(report->Get(expected_inbound_audio_codec.id()));
+  ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id()));
   EXPECT_EQ(expected_inbound_audio_codec,
             report->Get(expected_inbound_audio_codec.id())->cast_to<
                   RTCCodecStats>());
 
-  ASSERT(report->Get(expected_outbound_audio_codec.id()));
+  ASSERT_TRUE(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()));
+  ASSERT_TRUE(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()));
+  ASSERT_TRUE(report->Get(expected_outbound_video_codec.id()));
   EXPECT_EQ(expected_outbound_video_codec,
             report->Get(expected_outbound_video_codec.id())->cast_to<
                   RTCCodecStats>());
@@ -1618,7 +1618,7 @@
   expected_audio.jitter = 4.5;
   expected_audio.fraction_lost = 5.5;
 
-  ASSERT(report->Get(expected_audio.id()));
+  ASSERT_TRUE(report->Get(expected_audio.id()));
   const RTCInboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
@@ -1703,7 +1703,7 @@
   expected_video.fraction_lost = 4.5;
   expected_video.frames_decoded = 8;
 
-  ASSERT(report->Get(expected_video.id()));
+  ASSERT_TRUE(report->Get(expected_video.id()));
   const RTCInboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCInboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
@@ -1776,7 +1776,7 @@
   expected_audio.bytes_sent = 3;
   expected_audio.round_trip_time = 4.5;
 
-  ASSERT(report->Get(expected_audio.id()));
+  ASSERT_TRUE(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
@@ -1859,7 +1859,7 @@
   expected_video.frames_encoded = 8;
   expected_video.qp_sum = 16;
 
-  ASSERT(report->Get(expected_video.id()));
+  ASSERT_TRUE(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);
@@ -1943,7 +1943,7 @@
   expected_audio.bytes_sent = 3;
   // |expected_audio.round_trip_time| should be undefined.
 
-  ASSERT(report->Get(expected_audio.id()));
+  ASSERT_TRUE(report->Get(expected_audio.id()));
   const RTCOutboundRTPStreamStats& audio = report->Get(
       expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(audio, expected_audio);
@@ -1965,7 +1965,7 @@
   // |expected_video.round_trip_time| should be undefined.
   // |expected_video.qp_sum| should be undefined.
 
-  ASSERT(report->Get(expected_video.id()));
+  ASSERT_TRUE(report->Get(expected_video.id()));
   const RTCOutboundRTPStreamStats& video = report->Get(
       expected_video.id())->cast_to<RTCOutboundRTPStreamStats>();
   EXPECT_EQ(video, expected_video);