content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc - Issue 661693003: Avoid reporting 0 as input level when AudioProcessing zero out the audio data

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Unified Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

Issue 661693003: Avoid reporting 0 as input level when AudioProcessing zero out the audio data (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master

Patch Set: fixed the unittests Created 6 years, 2 months ago

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« content/renderer/media/media_stream_audio_level_calculator.h ('K') | « content/renderer/media/webaudio_capturer_source.cc ('k') | content/renderer/media/webrtc_audio_capturer.cc » ('j') | no next file with comments »
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Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

index f4a037ca6e97e7c3f22117c24bc3e0b7a9c55ea1..c7afce616effe21d7ce227b83b4493f73144d21f 100644

--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

@@ -74,14 +74,14 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {

EXPECT_CALL(*sink,

OnData(_, 16, params_.sample_rate(), params_.channels(),

params_.frames_per_buffer()));

- track_->Capture(data.get(), base::TimeDelta(), 255, false, false);

+ track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);

// Remove the sink from the webrtc track.

webrtc_track->RemoveSink(sink.get());

sink.reset();

// Verify that no more callback gets into the sink.

- track_->Capture(data.get(), base::TimeDelta(), 255, false, false);

+ track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);

}

TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {