added uma stats to check the usage of media stream audio track audio processing

content/renderer/media/media_stream_audio_processor.cc

 // Copyright 2013 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/media_stream_audio_processor.h"
 
 #include "base/command_line.h"
 #include "base/debug/trace_event.h"
 #include "base/metrics/field_trial.h"
+#include "base/metrics/histogram.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_converter.h"
 #include "media/base/audio_fifo.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/webrtc/modules/audio_processing/typing_detection.h"
 
 namespace content {
 
 namespace {
 
 using webrtc::AudioProcessing;
 using webrtc::MediaConstraintsInterface;
 
 #if defined(OS_ANDROID)
 const int kAudioProcessingSampleRate = 16000;
 #else
 const int kAudioProcessingSampleRate = 32000;
 #endif
 const int kAudioProcessingNumberOfChannel = 1;

[tommi (sloooow) - chröme, 2014/03/17 16:00:39]

nit: kAudioProcessingNumberOfChannels (plural)

I know that the number is currently 1, but "number of channel" doesn't mean the
same thing as "number of channels" (plural) which is what you want here.

[no longer working on chromium, 2014/03/19 12:50:11]

Done.

 
 const int kMaxNumberOfBuffersInFifo = 2;
 
+enum AudioTrackProcessingStates {

[Alexei Svitkine (slow), 2014/03/17 22:15:50]

Nit: Add a comment that this is used for an UMA histograms and entries shouldn't
re-ordered or removed.

[no longer working on chromium, 2014/03/19 12:50:11]

Done.

+  AUDIO_PROCESSING_ENABLED = 0,
+  AUDIO_PROCESSING_DISABLED,
+  AUDIO_PROCESSING_IN_WEBRTC,
+  AUDIO_PROCESSING_MAX
+};
+
 }  // namespace
 
 class MediaStreamAudioProcessor::MediaStreamAudioConverter
     : public media::AudioConverter::InputCallback {
  public:
   MediaStreamAudioConverter(const media::AudioParameters& source_params,
                             const media::AudioParameters& sink_params)
      : source_params_(source_params),
        sink_params_(sink_params),
        audio_converter_(source_params, sink_params_, false) {
@@ skipping 211 matching lines @@
 void MediaStreamAudioProcessor::GetStats(AudioProcessorStats* stats) {
   stats->typing_noise_detected =
       (base::subtle::Acquire_Load(&typing_detected_) != false);
   GetAecStats(audio_processing_.get(), stats);
 }
 
 void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
     const blink::WebMediaConstraints& constraints, int effects,
     MediaStreamType type) {
   DCHECK(!audio_processing_);
-  if (!IsAudioTrackProcessingEnabled())
+  if (!IsAudioTrackProcessingEnabled()) {
+    UMA_HISTOGRAM_ENUMERATION("Media.AudioTrackProcessingStates",
+                              AUDIO_PROCESSING_IN_WEBRTC, AUDIO_PROCESSING_MAX);
     return;
+  }
 
   RTCMediaConstraints native_constraints(constraints);
 
   // Only apply the fixed constraints for gUM of MEDIA_DEVICE_AUDIO_CAPTURE.
   DCHECK(IsAudioMediaType(type));
   if (type == MEDIA_DEVICE_AUDIO_CAPTURE)
     ApplyFixedAudioConstraints(&native_constraints);
 
   if (effects & media::AudioParameters::ECHO_CANCELLER) {
     // If platform echo canceller is enabled, disable the software AEC.
@@ skipping 32 matching lines @@
   const bool enable_high_pass_filter = GetPropertyFromConstraints(
       &native_constraints, MediaConstraintsInterface::kHighpassFilter);
 
   audio_mirroring_ = GetPropertyFromConstraints(
       &native_constraints, webrtc::MediaConstraintsInterface::kAudioMirroring);
 
   // Return immediately if no audio processing component is enabled.
   if (!enable_aec && !enable_experimental_aec && !enable_ns &&
       !enable_high_pass_filter && !enable_typing_detection && !enable_agc &&
       !enable_experimental_ns) {
+    UMA_HISTOGRAM_ENUMERATION("Media.AudioTrackProcessingStates",
+                              AUDIO_PROCESSING_DISABLED, AUDIO_PROCESSING_MAX);
     return;
   }
 
   // Create and configure the webrtc::AudioProcessing.
   audio_processing_.reset(webrtc::AudioProcessing::Create(0));
 
   // Enable the audio processing components.
   if (enable_aec) {
     EnableEchoCancellation(audio_processing_.get());
     if (enable_experimental_aec)
@@ skipping 22 matching lines @@
   if (enable_agc)
     EnableAutomaticGainControl(audio_processing_.get());
 
   // Configure the audio format the audio processing is running on. This
   // has to be done after all the needed components are enabled.
   CHECK_EQ(audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate),
            0);
   CHECK_EQ(audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel,
                                                kAudioProcessingNumberOfChannel),
            0);
+
+  UMA_HISTOGRAM_ENUMERATION("Media.AudioTrackProcessingStates",

[Alexei Svitkine (slow), 2014/03/17 22:15:50]

Nit: Can you make a function in the anon namespace that takes an enum value and
logs this? Each macro use bloats the binary a bit, so it's better to consolidate
ones for the same histogram.

[no longer working on chromium, 2014/03/19 12:50:11]

Done.

+                            AUDIO_PROCESSING_ENABLED, AUDIO_PROCESSING_MAX);
 }
 
 void MediaStreamAudioProcessor::InitializeCaptureConverter(
     const media::AudioParameters& source_params) {
   DCHECK(main_thread_checker_.CalledOnValidThread());
   DCHECK(source_params.IsValid());
 
   // Create and initialize audio converter for the source data.
   // When the webrtc AudioProcessing is enabled, the sink format of the
   // converter will be the same as the post-processed data format, which is
@@ skipping 115 matching lines @@
 }
 
 bool MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled() const {
   const std::string group_name =
       base::FieldTrialList::FindFullName("MediaStreamAudioTrackProcessing");
   return group_name == "Enabled" || CommandLine::ForCurrentProcess()->HasSwitch(
       switches::kEnableAudioTrackProcessing);
 }
 
 }  // namespace content