Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 51 matching lines @@
     bool cbr_enabled = false;
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
     // This value may change in the struct's constructor.
     int low_rate_complexity = kDefaultComplexity;
     // low_rate_complexity is used when the bitrate is below this threshold.
     int complexity_threshold_bps = 12500;
     int complexity_threshold_window_bps = 1500;
     bool dtx_enabled = false;
     std::vector<int> supported_frame_lengths_ms;
-    const Clock* clock = Clock::GetRealTimeClock();
     int uplink_bandwidth_update_interval_ms = 200;
 
    private:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
     // If we are on Android, iOS and/or ARM, use a lower complexity setting as
     // default, to save encoder complexity.
     static const int kDefaultComplexity = 5;
 #else
     static const int kDefaultComplexity = 9;
 #endif
   };
 
   static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
   static Config CreateConfig(const CodecInst& codec_inst);
 
   using AudioNetworkAdaptorCreator =
       std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
-                                                         RtcEventLog*,
-                                                         const Clock*)>;
+                                                         RtcEventLog*)>;
   AudioEncoderOpus(
       const Config& config,
       AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
       std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
 
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderOpus() override;
 
   // Static interface for use by BuiltinAudioEncoderFactory.
@@ skipping 12 matching lines @@
 
   // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
   // being inactive. During that, it still sends 2 packets (one for content, one
   // for signaling) about every 400 ms.
   bool SetDtx(bool enable) override;
   bool GetDtx() const override;
 
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
   bool EnableAudioNetworkAdaptor(const std::string& config_string,
-                                 RtcEventLog* event_log,
-                                 const Clock* clock) override;
+                                 RtcEventLog* event_log) override;
   void DisableAudioNetworkAdaptor() override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
   void OnReceivedUplinkRecoverablePacketLossFraction(
       float uplink_recoverable_packet_loss_fraction) override;
   void OnReceivedUplinkBandwidth(
       int target_audio_bitrate_bps,
       rtc::Optional<int64_t> probing_interval_ms) override;
   void OnReceivedRtt(int rtt_ms) override;
   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
@@ skipping 26 matching lines @@
   void SetNumChannelsToEncode(size_t num_channels_to_encode);
   void SetProjectedPacketLossRate(float fraction);
 
   // TODO(minyue): remove "override" when we can deprecate
   // |AudioEncoder::SetTargetBitrate|.
   void SetTargetBitrate(int target_bps) override;
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
       const ProtoString& config_string,
-      RtcEventLog* event_log,
-      const Clock* clock) const;
+      RtcEventLog* event_log) const;
 
   void MaybeUpdateUplinkBandwidth();
 
   Config config_;
   const bool send_side_bwe_with_overhead_;
   float packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   size_t num_channels_to_encode_;
   int next_frame_length_ms_;
   int complexity_;
   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
   AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
   rtc::Optional<size_t> overhead_bytes_per_packet_;
   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
   rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_