Adds more logging for audio input issues.

content/renderer/media/webrtc_audio_capturer.cc

Index: content/renderer/media/webrtc_audio_capturer.cc
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 7318cba1ba5e32b5c89f6aacfc41fa94f2106f9e..419f15b83f696488f309a7b0b5c58ccab0b6255b 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -38,6 +38,13 @@ const int kValidInputRates[] = {48000, 44100};
 const int kValidInputRates[] = {44100};
 #endif
 
+// Time constant for AudioPowerMonitor.  See AudioPowerMonitor ctor comments
+// for semantics.  This value was arbitrarily chosen, but seems to work well.
+const int kPowerMonitorTimeConstantMs = 10;
+
+// The time between two audio power level samples.
+const int kPowerMonitorLogIntervalSeconds = 10;
+
 }  // namespace
 
 // Reference counted container of WebRtcLocalAudioTrack delegate.
@@ -201,7 +208,6 @@ bool WebRtcAudioCapturer::Initialize() {
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
-
   return true;
 }
 
@@ -222,7 +228,10 @@ WebRtcAudioCapturer::WebRtcAudioCapturer(
       peer_connection_mode_(false),
       key_pressed_(false),
       need_audio_processing_(false),
-      audio_device_(audio_device) {
+      audio_device_(audio_device),
+      audio_power_monitor_(
+          device_info_.device.input.sample_rate,
+          base::TimeDelta::FromMilliseconds(kPowerMonitorTimeConstantMs)) {
   DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()";
 }
 
@@ -483,6 +492,20 @@ void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
     (*it)->SetAudioProcessor(audio_processor_);
   }
 
+  if ((base::TimeTicks::Now() - last_audio_level_log_time_).InSeconds() >
+          kPowerMonitorLogIntervalSeconds) {
+    audio_power_monitor_.Scan(*audio_source, audio_source->frames());
+
+    last_audio_level_log_time_ = base::TimeTicks::Now();
+
+    std::pair<float, bool> result =
+        audio_power_monitor_.ReadCurrentPowerAndClip();

[tommi (sloooow) - chröme, 2014/04/10 09:07:22]

nit: I don't suppose there's a single threaded AudioPowerMonitor that we could
use only on the capture thread and avoid grabbing locks?

[jiayl, 2014/04/10 16:46:34]

No, there isn't. 

+    WebRtcLogMessage(base::StringPrintf(
+      "WAC::Capture: current_audio_power=%.2fdBFS.", result.first));

[tommi (sloooow) - chröme, 2014/04/10 09:07:22]

indent

[jiayl, 2014/04/10 16:46:34]

Done.

+
+    audio_power_monitor_.Reset();
+  }
+
   // Push the data to the processor for processing.
   audio_processor_->PushCaptureData(audio_source);