webrtc/modules/audio_coding/main/test/opus_test.cc - Issue 1172163004: Reformat existing code. There should be no functional effects.

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Unified Diff: webrtc/modules/audio_coding/main/test/opus_test.cc

Issue 1172163004: Reformat existing code. There should be no functional effects. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 6 months ago

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Index: webrtc/modules/audio_coding/main/test/opus_test.cc

diff --git a/webrtc/modules/audio_coding/main/test/opus_test.cc b/webrtc/modules/audio_coding/main/test/opus_test.cc

index 09301df51c3e6992985e717f035c82a4cab6d857..a407fc5d36f0222897f7792d2f125415a480bf5d 100644

--- a/webrtc/modules/audio_coding/main/test/opus_test.cc

+++ b/webrtc/modules/audio_coding/main/test/opus_test.cc

@@ -277,12 +277,12 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,

bitstream_len_byte = WebRtcOpus_Encode(

opus_mono_encoder_, &audio[read_samples],

frame_length, kMaxBytes, bitstream);

- ASSERT_GT(bitstream_len_byte, -1);

+ ASSERT_GE(bitstream_len_byte, 0);

} else {

bitstream_len_byte = WebRtcOpus_Encode(

opus_stereo_encoder_, &audio[read_samples],

frame_length, kMaxBytes, bitstream);

- ASSERT_GT(bitstream_len_byte, -1);

+ ASSERT_GE(bitstream_len_byte, 0);

}

// Simulate packet loss by setting |packet_loss_| to "true" in