Index: webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc |
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc |
index de07ae20722a7240a20af3d2e9c00aa2ddc570aa..588e9e0f5a45c3ee426ff2f8020e23f2d07c5080 100644 |
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc |
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc |
@@ -133,11 +133,12 @@ class NetEqImplTest : public ::testing::Test { |
new TimestampScaler(*deps.decoder_database.get())); |
neteq_.reset(new NetEqImpl(config_, std::move(deps))); |
- ASSERT_TRUE(neteq_ != NULL); |
+ ASSERT_TRUE(neteq_ != nullptr); |
} |
void UseNoMocks() { |
- ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance"; |
+ ASSERT_TRUE(neteq_ == nullptr) |
+ << "Must call UseNoMocks before CreateInstance"; |
use_mock_buffer_level_filter_ = false; |
use_mock_decoder_database_ = false; |
use_mock_delay_peak_detector_ = false; |
@@ -530,7 +531,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) { |
// Check the timestamp for the last value in the sync buffer. This should |
// be one full frame length ahead of the RTP timestamp. |
const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test(); |
- ASSERT_TRUE(sync_buffer != NULL); |
+ ASSERT_TRUE(sync_buffer != nullptr); |
EXPECT_EQ(rtp_header.header.timestamp + kPayloadLengthSamples, |
sync_buffer->end_timestamp()); |