Adds more logging for audio input issues.

-Adds the audio input device id and name to the stream generation message.
-Re-enables the no_data_timer in AudioInputController for logging. The NO_DATA_ERROR is ignored and will not stop capturing.
-Adds a AudioPowerMonitor to WebRtcAudioCapturer to log the audio power level.

R=vrk@chromium.org,xians@chromium.org,tommi@chromium.org

BUG=360756

Review URL: https://codereview.chromium.org/229573003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@263127 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 32 insertions, 1 deletions

diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 7318cba..482d25f 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.
@@ -222,7 +229,10 @@ 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 +493,20 @@ 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();
+    WebRtcLogMessage(base::StringPrintf(
+        "WAC::Capture: current_audio_power=%.2fdBFS.", result.first));
+
+    audio_power_monitor_.Reset();
+  }
+
   // Push the data to the processor for processing.
   audio_processor_->PushCaptureData(audio_source);
 
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index 78a21811..63c846f 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -17,6 +17,7 @@
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/tagged_list.h"
 #include "media/audio/audio_input_device.h"
+#include "media/audio/audio_power_monitor.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 
@@ -207,6 +208,12 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // of RenderThread.
   WebRtcAudioDeviceImpl* audio_device_;
 
+  // Audio power monitor for logging audio power level.
+  media::AudioPowerMonitor audio_power_monitor_;
+
+  // Records when the last time audio power level is logged.
+  base::TimeTicks last_audio_level_log_time_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };