Adds more logging for audio input issues.

content/renderer/media/webrtc_audio_capturer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
@@ skipping 20 matching lines @@
 const int kValidInputRates[] =
     {192000, 96000, 48000, 44100, 32000, 16000, 8000};
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
 const int kValidInputRates[] = {48000, 44100};
 #elif defined(OS_ANDROID)
 const int kValidInputRates[] = {48000, 44100};
 #else
 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.
 // TODO(xians): Switch to MediaStreamAudioSinkOwner.
 class WebRtcAudioCapturer::TrackOwner
     : public base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner> {
  public:
   explicit TrackOwner(WebRtcLocalAudioTrack* track)
       : delegate_(track) {}
 
@@ skipping 143 matching lines @@
 
   // Create and configure the default audio capturing source.
   SetCapturerSource(AudioDeviceFactory::NewInputDevice(render_view_id_),
                     channel_layout,
                     static_cast<float>(device_info_.device.input.sample_rate));
 
   // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware
   // information from the capturer.
   if (audio_device_)
     audio_device_->AddAudioCapturer(this);
-
   return true;
 }
 
 WebRtcAudioCapturer::WebRtcAudioCapturer(
     int render_view_id,
     const StreamDeviceInfo& device_info,
     const blink::WebMediaConstraints& constraints,
     WebRtcAudioDeviceImpl* audio_device)
     : constraints_(constraints),
       audio_processor_(
           new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
               constraints, device_info.device.input.effects,
               device_info.device.type, audio_device)),
       running_(false),
       render_view_id_(render_view_id),
       device_info_(device_info),
       volume_(0),
       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()";
 }
 
 WebRtcAudioCapturer::~WebRtcAudioCapturer() {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(tracks_.IsEmpty());
   DCHECK(!running_);
   DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()";
 }
 
@@ skipping 240 matching lines @@
 
   // Notify the tracks on when the format changes. This will do nothing if
   // |tracks_to_notify_format| is empty.
   media::AudioParameters output_params = audio_processor_->OutputFormat();
   for (TrackList::ItemList::const_iterator it = tracks_to_notify_format.begin();
        it != tracks_to_notify_format.end(); ++it) {
     (*it)->OnSetFormat(output_params);
     (*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);
 
   // Process and consume the data in the processor until there is not enough
   // data in the processor.
   int16* output = NULL;
   int new_volume = 0;
   while (audio_processor_->ProcessAndConsumeData(
       audio_delay, current_volume, key_pressed, &new_volume, &output)) {
     // Feed the post-processed data to the tracks.
@@ skipping 85 matching lines @@
   DCHECK_NE(aec_dump_file, base::kInvalidPlatformFileValue);
   audio_processor_->StartAecDump(aec_dump_file);
 }
 
 void WebRtcAudioCapturer::StopAecDump() {
   DCHECK(thread_checker_.CalledOnValidThread());
   audio_processor_->StopAecDump();
 }
 
 }  // namespace content