Possible solution to synchronization problems in webrtc audio capturer.

content/renderer/media/webrtc_local_audio_renderer.cc

Index: content/renderer/media/webrtc_local_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_local_audio_renderer.cc b/content/renderer/media/webrtc_local_audio_renderer.cc
index d6efff69e6c730bec60d6ce39fe65b8face8f39b..f24045ab759dab16fe0556cf9ca830528b64a452 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -163,7 +163,7 @@ void WebRtcLocalAudioRenderer::Start() {
   // cases where resampling is needed on the output side.
   // TODO(henrika): verify this scheme on as many different devices and
   // combinations of sample rates as possible
-  media::AudioParameters source_params = source_->audio_parameter();
+  media::AudioParameters source_params = source_->audio_parameters();
   media::AudioParameters sink_params(source_params.format(),
                                      source_params.channel_layout(),
                                      source_params.sample_rate(),