webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.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_oldapi.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_oldapi.cc

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

index 0af6af800287d1425cedd6c0d84c261440a5c8db..e5371d0b23b5b0aefefdd019e7f3b1eea35f5c59 100644

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

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

@@ -314,7 +314,8 @@ TEST_F(AudioCodingModuleTestOldApi, VerifyOutputFrame) {

EXPECT_EQ(id_, audio_frame.id_);

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_);

}