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: Created 5 years, 6 months ago

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« webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c ('K') | « webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc ('k') | webrtc/modules/audio_coding/neteq/decision_logic_normal.h » ('j') | webrtc/modules/audio_coding/neteq/expand.cc » ('J')
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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 ad7e2f9be8a002dd4506d2b5b9f8a73148ee77f9..c61d25ad19acd026b11bca7dd528f87a3a19d4be 100644

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

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

@@ -276,7 +276,7 @@ void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,

int bitstream_len_byte_int = WebRtcOpus_Encode(

(channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,

&audio[read_samples], frame_length, kMaxBytes, bitstream);

- ASSERT_GT(bitstream_len_byte_int, -1);

+ ASSERT_GE(bitstream_len_byte_int, 0);

bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);

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