Rename APM Config ReportedDelay to DelayAgnostic

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 48 matching lines @@
   explicit ExtendedFilter(bool enabled) : enabled(enabled) {}
   bool enabled;
 };
 
 // Use to disable the reported system delays. By disabling the reported system
 // delays the echo cancellation algorithm assumes the process and reverse
 // streams to be aligned. This configuration only applies to EchoCancellation
 // and not EchoControlMobile and is set with AudioProcessing::SetExtraOptions().
 // Note that by disabling reported system delays the EchoCancellation may
 // regress in performance.
+// TODO(henrik.lundin): Remove ReportedDelay once DelayAgnostic has
+// propagated through to all channels
+// (https://code.google.com/p/webrtc/issues/detail?id=4651).
 struct ReportedDelay {
   ReportedDelay() : enabled(true) {}
   explicit ReportedDelay(bool enabled) : enabled(enabled) {}
   bool enabled;
 };
+struct DelayAgnostic {
+  DelayAgnostic() : enabled(false) {}
+  explicit DelayAgnostic(bool enabled) : enabled(enabled) {}
+  bool enabled;
+};
 
 // Use to enable experimental gain control (AGC). At startup the experimental
 // AGC moves the microphone volume up to |startup_min_volume| if the current
 // microphone volume is set too low. The value is clamped to its operating range
 // [12, 255]. Here, 255 maps to 100%.
 //
 // Must be provided through AudioProcessing::Create(Confg&).
 #if defined(WEBRTC_CHROMIUM_BUILD)
 static const int kAgcStartupMinVolume = 85;
 #else
@@ skipping 698 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_