Add UMA reporting for audio hardware parameters.

media/audio/audio_output_resampler.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/audio_output_resampler.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/message_loop.h"
+#include "base/metrics/histogram.h"
 #include "base/time.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_output_dispatcher_impl.h"
 #include "media/audio/audio_output_proxy.h"
 #include "media/audio/audio_util.h"
+#include "media/audio/sample_rates.h"
 #include "media/base/audio_pull_fifo.h"
+#include "media/base/limits.h"
 #include "media/base/multi_channel_resampler.h"
 
+#if defined(OS_WIN)
+#include "media/audio/win/audio_low_latency_output_win.h"
+#endif
+
 namespace media {
 
+static void RecordStats(const AudioParameters& output_params) {
+  UMA_HISTOGRAM_ENUMERATION(
+      "Media.HardwareAudioBitsPerChannel", output_params.bits_per_sample(),

[scherkus (not reviewing), 2012/09/12 13:34:50]

don't you have a WASAPI version of this?

[DaleCurtis, 2012/09/12 14:03:52]

Not sure I'm landing that. I don't think it will work w/o supporting 24-bit
interleave on ToInterleaved() first.

+      limits::kMaxBitsPerSample);
+#if defined(OS_WIN)
+  // TODO(dalecurtis): Since channel mixing is handle by the output device right

[scherkus (not reviewing), 2012/09/12 13:35:13]

s/handle/handled/

[DaleCurtis, 2012/09/12 14:03:52]

Done.

+  // now and not by AudioOutputResampler, we need to query for hardware channel
+  // information.  Remove once AOR handles this, http://crbug.com/138762
+  UMA_HISTOGRAM_ENUMERATION(
+      "Media.HardwareAudioChannelLayout",
+      WASAPIAudioOutputStream::HardwareChannelLayout(), CHANNEL_LAYOUT_MAX);
+  UMA_HISTOGRAM_ENUMERATION(
+      "Media.HardwareAudioChannelCount",
+      WASAPIAudioOutputStream::HardwareChannelCount(), limits::kMaxChannels);
+#else
+  UMA_HISTOGRAM_ENUMERATION(
+      "Media.HardwareAudioChannelLayout", output_params.channel_layout(),
+      CHANNEL_LAYOUT_MAX);
+  UMA_HISTOGRAM_ENUMERATION(
+      "Media.HardwareAudioChannelCount", output_params.channels(),
+      limits::kMaxChannels);
+#endif
+  AudioSampleRate asr = media::AsAudioSampleRate(output_params.sample_rate());
+  if (asr != kUnexpectedAudioSampleRate) {
+    UMA_HISTOGRAM_ENUMERATION(
+        "Media.HardwareAudioSamplesPerSecond", asr, kUnexpectedAudioSampleRate);
+  } else {
+    UMA_HISTOGRAM_COUNTS(
+        "Media.HardwareAudioSamplesPerSecondUnexpected",
+        output_params.sample_rate());
+  }
+}
+
 AudioOutputResampler::AudioOutputResampler(AudioManager* audio_manager,
                                            const AudioParameters& input_params,
                                            const AudioParameters& output_params,
                                            const base::TimeDelta& close_delay)
     : AudioOutputDispatcher(audio_manager, input_params),
       source_callback_(NULL),
       io_ratio_(1),
       input_bytes_per_frame_(input_params.GetBytesPerFrame()),
       output_bytes_per_frame_(output_params.GetBytesPerFrame()),
       outstanding_audio_bytes_(0) {
@@ skipping 39 matching lines @@
               base::Unretained(this))));
     }
 
     DVLOG(1) << "I/O ratio is " << io_ratio_;
   }
 
   // TODO(dalecurtis): All this code should be merged into AudioOutputMixer once
   // we've stabilized the issues there.
   dispatcher_ = new AudioOutputDispatcherImpl(
       audio_manager, output_params, close_delay);
+
+  // Record UMA statistics for the hardware configuration.
+  RecordStats(output_params);
 }
 
 AudioOutputResampler::~AudioOutputResampler() {}
 
 bool AudioOutputResampler::OpenStream() {
   // TODO(dalecurtis): Automatically revert to high latency path if OpenStream()
   // fails; use default high latency output values + rebuffering / resampling.
   return dispatcher_->OpenStream();
 }
 
@@ skipping 106 matching lines @@
     source_callback_->OnError(stream, code);
 }
 
 void AudioOutputResampler::WaitTillDataReady() {
   base::AutoLock auto_lock(source_lock_);
   if (source_callback_ && !outstanding_audio_bytes_)
     source_callback_->WaitTillDataReady();
 }
 
 }  // namespace media