Revert 217768 "Adding key press detection in the browser process."

trunk/src/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 84 matching lines @@
 class WebRtcAudioCapturer::TrackOwner
     : public base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner> {
  public:
   explicit TrackOwner(WebRtcLocalAudioTrack* track)
       : delegate_(track) {}
 
   void CaptureData(const int16* audio_data,
                    int number_of_channels,
                    int number_of_frames,
                    int audio_delay_milliseconds,
-                   int volume,
-                   bool key_pressed) {
+                   int volume) {
     base::AutoLock lock(lock_);
     if (delegate_) {
       delegate_->CaptureData(audio_data,
                              number_of_channels,
                              number_of_frames,
                              audio_delay_milliseconds,
-                             volume,
-                             key_pressed);
+                             volume);
     }
   }
 
   void SetCaptureFormat(const media::AudioParameters& params) {
     base::AutoLock lock(lock_);
     if (delegate_)
       delegate_->SetCaptureFormat(params);
   }
 
   void Reset() {
@@ skipping 294 matching lines @@
   // Store the setting since SetAutomaticGainControl() can be called before
   // Initialize(), in this case stored setting will be applied in Start().
   agc_is_enabled_ = enable;
 
   if (source_.get())
     source_->SetAutomaticGainControl(enable);
 }
 
 void WebRtcAudioCapturer::Capture(media::AudioBus* audio_source,
                                   int audio_delay_milliseconds,
-                                  double volume,
-                                  bool key_pressed) {
-// This callback is driven by AudioInputDevice::AudioThreadCallback if
-// |source_| is AudioInputDevice, otherwise it is driven by client's
-// CaptureCallback.
+                                  double volume) {
+  // This callback is driven by AudioInputDevice::AudioThreadCallback if
+  // |source_| is AudioInputDevice, otherwise it is driven by client's
+  // CaptureCallback.
 #if defined(OS_WIN) || defined(OS_MACOSX)
   DCHECK_LE(volume, 1.0);
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
   // We have a special situation on Linux where the microphone volume can be
   // "higher than maximum". The input volume slider in the sound preference
   // allows the user to set a scaling that is higher than 100%. It means that
   // even if the reported maximum levels is N, the actual microphone level can
   // go up to 1.5x*N and that corresponds to a normalized |volume| of 1.5x.
   DCHECK_LE(volume, 1.6);
 #endif
@@ skipping 23 matching lines @@
   // Interleave, scale, and clip input to int and store result in
   // a local byte buffer.
   audio_source->ToInterleaved(audio_source->frames(), bytes_per_sample,
                               buffer_ref_while_calling->buffer());
 
   // Feed the data to the tracks.
   for (TrackList::const_iterator it = tracks.begin();
        it != tracks.end();
        ++it) {
     (*it)->CaptureData(buffer_ref_while_calling->buffer(),
-                       audio_source->channels(),
-                       audio_source->frames(),
-                       audio_delay_milliseconds,
-                       volume,
-                       key_pressed);
+                       audio_source->channels(), audio_source->frames(),
+                       audio_delay_milliseconds, volume_);
   }
 }
 
 void WebRtcAudioCapturer::OnCaptureError() {
   NOTIMPLEMENTED();
 }
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   // |buffer_| can be NULL when SetCapturerSource() or Initialize() has not
   // been called.
   return buffer_.get() ? buffer_->params() : media::AudioParameters();
 }
 
 }  // namespace content