webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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Unified Diff: webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 4 months ago

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Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

index eea51a3489e285df9592de02cc21d500f4d99a65..418ddd111f104eb67f71c6216fdeaf4764f9a20e 100644

--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc

@@ -272,7 +272,8 @@ TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {

EXPECT_TRUE(acm_->Get10MsAudio(&audio_frame));

EXPECT_EQ(0u, audio_frame.timestamp_);

EXPECT_GT(audio_frame.num_channels_, 0);

- EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);

+ EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100),

+ audio_frame.samples_per_channel_);

EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);

}