webrtc/modules/audio_coding/codecs/isac/main/source/isac_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/codecs/isac/main/source/isac_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/codecs/isac/main/source/isac_unittest.cc

diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

index a751c2479215b3966d4a90233c3f5659ebe6d7ff..84c712ee6ee77c249219f6c6739d2df9ad726759 100644

--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

@@ -97,10 +97,12 @@ TEST_F(IsacTest, IsacUpdateBWE) {

encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);

EXPECT_EQ(0, encoded_bytes);

encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);

+ EXPECT_GT(encoded_bytes, 0);

// Call to update bandwidth estimator with real data.

EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,

- encoded_bytes, 1, 12345, 56789));

+ static_cast<size_t>(encoded_bytes),

+ 1, 12345, 56789));

// Free memory.

EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));