Remove the last piece of deprecated synchronous IO code.

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 addf9637a5ff68964a0cdfc471433d028de5981e..76a0df5b81b17b18cbf88d6e7d6cf7db13111c0d 100644
--- a/content/renderer/media/webrtc_local_audio_renderer.cc
+++ b/content/renderer/media/webrtc_local_audio_renderer.cc
@@ -283,8 +283,7 @@ void WebRtcLocalAudioRenderer::ReconfigureSink(
 
   sink_params_ = media::AudioParameters(source_params_.format(),
       source_params_.channel_layout(), source_params_.channels(),
-      source_params_.input_channels(), source_params_.sample_rate(),
-      source_params_.bits_per_sample(),
+      source_params_.sample_rate(), source_params_.bits_per_sample(),
 #if defined(OS_ANDROID)
       // On Android, input and output use the same sample rate. In order to
       // use the low latency mode, we need to use the buffer size suggested by