Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 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 146 matching lines @@
   // returns a pointer to an array where they are stored (which is required to
   // live as long as this encoder). Unless the returned array is empty, you may
   // not call any methods on this encoder afterwards, except for the
   // destructor. The default implementation just returns an empty array.
   // NOTE: This method is subject to change. Do not call or override it.
   virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
   ReclaimContainedEncoders();
 
   // Enables audio network adaptor. Returns true if successful.
   virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
-                                         RtcEventLog* event_log,
-                                         const Clock* clock);
+                                         RtcEventLog* event_log);
 
   // Disables audio network adaptor.
   virtual void DisableAudioNetworkAdaptor();
 
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
   // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
   // Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
@@ skipping 25 matching lines @@
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_