Adding typing detection to the APM in chrome.

content/renderer/media/media_stream_audio_processor_options.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_options.h"
 
 #include "base/files/file_path.h"
 #include "base/logging.h"
 #include "base/path_service.h"
 #include "base/strings/utf_string_conversions.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "media/audio/audio_parameters.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/include/audio_processing.h"
+#include "third_party/webrtc/modules/audio_processing/typing_detection.h"
 
 namespace content {
 
 namespace {
 
 // Constant constraint keys which enables default audio constraints on
 // mediastreams with audio.
 struct {
   const char* key;
   const char* value;
@@ skipping 88 matching lines @@
   int err = audio_processing->noise_suppression()->set_level(
       webrtc::NoiseSuppression::kHigh);
   err |= audio_processing->noise_suppression()->Enable(true);
   CHECK_EQ(err, 0);
 }
 
 void EnableHighPassFilter(AudioProcessing* audio_processing) {
   CHECK_EQ(audio_processing->high_pass_filter()->Enable(true), 0);
 }
 
-void EnableTypingDetection(AudioProcessing* audio_processing) {
+void EnableTypingDetection(AudioProcessing* audio_processing,
+                           webrtc::TypingDetection* typing_detector) {
   int err = audio_processing->voice_detection()->Enable(true);
   err |= audio_processing->voice_detection()->set_likelihood(
       webrtc::VoiceDetection::kVeryLowLikelihood);
   CHECK_EQ(err, 0);
+
+  // Configure the update period to 100ms (10 * 10ms) in the typing detector.
+  typing_detector->SetParameters(0, 0, 0, 0, 0, 10);
 }
 
 void EnableExperimentalEchoCancellation(AudioProcessing* audio_processing) {
   webrtc::Config config;
   config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true));
   audio_processing->SetExtraOptions(config);
 }
 
 void StartAecDump(AudioProcessing* audio_processing) {
   // TODO(grunell): Figure out a more suitable directory for the audio dump
@@ skipping 27 matching lines @@
   const webrtc::GainControl::Mode mode = webrtc::GainControl::kFixedDigital;
 #else
   const webrtc::GainControl::Mode mode = webrtc::GainControl::kAdaptiveAnalog;
 #endif
   int err = audio_processing->gain_control()->set_mode(mode);
   err |= audio_processing->gain_control()->Enable(true);
   CHECK_EQ(err, 0);
 }
 
 }  // namespace content