| Index: content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| index ded95d6e001c8c24211d0cfa64569e1ca421b8ea..9b2741041d248e8e012e1dace9017cc6630178c8 100644
|
| --- a/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| +++ b/content/renderer/media/webrtc_audio_capturer_unittest.cc
|
| @@ -45,7 +45,7 @@ class MockPeerConnectionAudioSink : public PeerConnectionAudioSink {
|
| int number_of_channels, int number_of_frames,
|
| const std::vector<int>& channels,
|
| int audio_delay_milliseconds, int current_volume,
|
| - bool need_audio_processing, bool key_pressed) OVERRIDE {
|
| + bool need_audio_processing, bool key_pressed) override {
|
| EXPECT_EQ(sample_rate, params_.sample_rate());
|
| EXPECT_EQ(number_of_channels, params_.channels());
|
| EXPECT_EQ(number_of_frames, params_.frames_per_buffer());
|
| @@ -59,7 +59,7 @@ class MockPeerConnectionAudioSink : public PeerConnectionAudioSink {
|
| int current_volume,
|
| bool need_audio_processing,
|
| bool key_pressed));
|
| - virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE {
|
| + virtual void OnSetFormat(const media::AudioParameters& params) override {
|
| params_ = params;
|
| FormatIsSet();
|
| }
|
|
|