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

trunk/src/content/renderer/media/webrtc_audio_device_unittest.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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 203 matching lines @@
   virtual ~MockWebRtcAudioCapturerSink() {}
 
   // WebRtcAudioCapturerSink implementation.
   virtual int CaptureData(const std::vector<int>& channels,
                           const int16* audio_data,
                           int sample_rate,
                           int number_of_channels,
                           int number_of_frames,
                           int audio_delay_milliseconds,
                           int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE {
+                          bool need_audio_processing) OVERRIDE {
     // Signal that a callback has been received.
     event_->Signal();
     return 0;
   }
 
   // Set the format for the capture audio parameters.
   virtual void SetCaptureFormat(
       const media::AudioParameters& params) OVERRIDE {}
 
  private:
@@ skipping 139 matching lines @@
   scoped_ptr<uint8[]> buffer(new uint8[output_packet_size]);
   base::Time start_time = base::Time::Now();
   int delay = 0;
   std::vector<int> voe_channels;
   voe_channels.push_back(channel);
   for (int j = 0; j < kNumberOfPacketsForLoopbackTest; ++j) {
     // Sending fake capture data to WebRtc.
     capturer_sink->CaptureData(
         voe_channels,
         reinterpret_cast<int16*>(capture_data.get() + input_packet_size * j),
-        params.sample_rate(),
-        params.channels(),
-        params.frames_per_buffer(),
-        kHardwareLatencyInMs,
-        1.0,
-        enable_apm,
-        false);
+        params.sample_rate(), params.channels(), params.frames_per_buffer(),
+        kHardwareLatencyInMs, 1.0, enable_apm);
 
     // Receiving data from WebRtc.
     renderer_source->RenderData(
         reinterpret_cast<uint8*>(buffer.get()),
         num_output_channels, webrtc_audio_device->output_buffer_size(),
         kHardwareLatencyInMs + delay);
     delay = (base::Time::Now() - start_time).InMilliseconds();
   }
 
   int latency = (base::Time::Now() - start_time).InMilliseconds();
@@ skipping 563 matching lines @@
 #endif
 
 TEST_F(MAYBE_WebRTCAudioDeviceTest,
        MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content