move the APM to chrome.

content/renderer/media/webrtc_audio_device_impl.cc

Index: content/renderer/media/webrtc_audio_device_impl.cc
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index aadcda9b67232434a7177501215c4b78215d0416..b38423a9e0b365bbc1a6e64df81eaf162fa1a1a5 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -51,59 +51,23 @@ int32_t WebRtcAudioDeviceImpl::Release() {
   }
   return ret;
 }
-int WebRtcAudioDeviceImpl::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) {
-  int total_delay_ms = 0;
+void WebRtcAudioDeviceImpl::CaptureData(const std::vector<int>& channels,
+                                        const int16* audio_data,
+                                        int sample_rate,
+                                        int number_of_channels,
+                                        int number_of_frames) {
   {
     base::AutoLock auto_lock(lock_);
     // Return immediately when not recording or |channels| is empty.
     // See crbug.com/274017: renderer crash dereferencing invalid channels[0].
     if (!recording_ || channels.empty())
-      return 0;
-
-    // Store the reported audio delay locally.
-    input_delay_ms_ = audio_delay_milliseconds;
-    total_delay_ms = input_delay_ms_ + output_delay_ms_;
-    DVLOG(2) << "total delay: " << input_delay_ms_ + output_delay_ms_;
-  }
-
-  // Write audio samples in blocks of 10 milliseconds to the registered
-  // webrtc::AudioTransport sink. Keep writing until our internal byte
-  // buffer is empty.
-  const int16* audio_buffer = audio_data;
-  const int samples_per_10_msec = (sample_rate / 100);
-  int accumulated_audio_samples = 0;
-  uint32_t new_volume = 0;
-  while (accumulated_audio_samples < number_of_frames) {
-    // Deliver 10ms of recorded 16-bit linear PCM audio.
-    int new_mic_level = audio_transport_callback_->OnDataAvailable(
-        &channels[0],
-        channels.size(),
-        audio_buffer,
-        sample_rate,
-        number_of_channels,
-        samples_per_10_msec,
-        total_delay_ms,
-        current_volume,
-        key_pressed,
-        need_audio_processing);
-
-    accumulated_audio_samples += samples_per_10_msec;
-    audio_buffer += samples_per_10_msec * number_of_channels;
-
-    // The latest non-zero new microphone level will be returned.
-    if (new_mic_level)
-      new_volume = new_mic_level;
+      return;
   }
 
-  return new_volume;
+  // Deliver 10ms of recorded 16-bit linear PCM audio.
+  audio_transport_callback_->OnDataAvailable(
+      &channels[0], channels.size(), audio_data, sample_rate,
+      number_of_channels, number_of_frames, 0, 0, false, false);
 }
 
 void WebRtcAudioDeviceImpl::SetCaptureFormat(