Add histogram to track down errors when initializing the audio client.

media/audio/win/audio_low_latency_input_win.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 "media/audio/win/audio_low_latency_input_win.h"
 
 #include <cmath>
 #include <memory>
 
 #include "base/logging.h"
@@ skipping 697 matching lines @@
   // This setting should lead to lowest possible latency.
   HRESULT hr = audio_client_->Initialize(
       AUDCLNT_SHAREMODE_SHARED, flags,
       0,  // hnsBufferDuration
       0, &format_, device_id_ == AudioDeviceDescription::kCommunicationsDeviceId
                        ? &kCommunicationsSessionId
                        : nullptr);
 
   if (FAILED(hr)) {
     open_result_ = OPEN_RESULT_AUDIO_CLIENT_INIT_FAILED;
+    UMA_HISTOGRAM_SPARSE_SLOWLY("Media.Audio.Capture.Win.InitError", hr);

[henrika (OOO until Aug 14), 2017/02/23 11:20:52]

Just FYI. Mac specific stats are callled e.g.
Media.Audio.InputBufferSizeWasChangedMac. Like your style better.

     return hr;
   }
 
   // Retrieve the length of the endpoint buffer shared between the client
   // and the audio engine. The buffer length determines the maximum amount
   // of capture data that the audio engine can read from the endpoint buffer
   // during a single processing pass.
   // A typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
   hr = audio_client_->GetBufferSize(&endpoint_buffer_size_frames_);
   if (FAILED(hr)) {
@@ skipping 99 matching lines @@
                             OPEN_RESULT_MAX + 1);
 }
 
 double WASAPIAudioInputStream::ProvideInput(AudioBus* audio_bus,
                                             uint32_t frames_delayed) {
   fifo_->Consume()->CopyTo(audio_bus);
   return 1.0;
 }
 
 }  // namespace media