Index: content/renderer/media/webrtc_local_audio_source_provider.cc |
diff --git a/content/renderer/media/webrtc_local_audio_source_provider.cc b/content/renderer/media/webrtc_local_audio_source_provider.cc |
index d77df9ac552ae06b1909eb256821a9e44e5c99d1..af465017c68505506889e35b02d40c8e642af44d 100644 |
--- a/content/renderer/media/webrtc_local_audio_source_provider.cc |
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc |
@@ -130,8 +130,8 @@ void WebRtcLocalAudioSourceProvider::provideInput( |
audio_converter_->Convert(output_wrapper_.get()); |
} |
-double WebRtcLocalAudioSourceProvider::ProvideInput( |
- media::AudioBus* audio_bus, base::TimeDelta buffer_delay) { |
+double WebRtcLocalAudioSourceProvider::ProvideInput(media::AudioBus* audio_bus, |
+ uint32_t frames_delayed) { |
if (fifo_->frames() >= audio_bus->frames()) { |
fifo_->Consume(audio_bus, 0, audio_bus->frames()); |
} else { |