Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  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 266 matching lines @@
   const int max_frame_length_ms =
       GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
 
   FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
                             &config.supported_frame_lengths_ms);
   return config;
 }
 
 class AudioEncoderOpus::PacketLossFractionSmoother {
  public:
-  explicit PacketLossFractionSmoother(const Clock* clock)
-      : clock_(clock),
-        last_sample_time_ms_(clock_->TimeInMilliseconds()),
+  explicit PacketLossFractionSmoother()
+      : last_sample_time_ms_(rtc::TimeMillis()),
         smoother_(kAlphaForPacketLossFractionSmoother) {}
 
   // Gets the smoothed packet loss fraction.
   float GetAverage() const {
     float value = smoother_.filtered();
     return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
   }
 
   // Add new observation to the packet loss fraction smoother.
   void AddSample(float packet_loss_fraction) {
-    int64_t now_ms = clock_->TimeInMilliseconds();
+    int64_t now_ms = rtc::TimeMillis();
     smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
                     packet_loss_fraction);
     last_sample_time_ms_ = now_ms;
   }
 
  private:
-  const Clock* const clock_;
   int64_t last_sample_time_ms_;
 
   // An exponential filter is used to smooth the packet loss fraction.
   rtc::ExpFilter smoother_;
 };
 
 AudioEncoderOpus::Config::Config() {
 #if WEBRTC_OPUS_VARIABLE_COMPLEXITY
   low_rate_complexity = 9;
 #endif
@@ skipping 41 matching lines @@
 }
 
 AudioEncoderOpus::AudioEncoderOpus(
     const Config& config,
     AudioNetworkAdaptorCreator&& audio_network_adaptor_creator,
     std::unique_ptr<SmoothingFilter> bitrate_smoother)
     : send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
           "WebRTC-SendSideBwe-WithOverhead")),
       packet_loss_rate_(0.0),
       inst_(nullptr),
-      packet_loss_fraction_smoother_(new PacketLossFractionSmoother(
-          config.clock)),
+      packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
               : [this](const ProtoString& config_string,
-                       RtcEventLog* event_log,
-                       const Clock* clock) {
+                       RtcEventLog* event_log) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
-                                                           event_log, clock);
+                                                           event_log);
               }),
       bitrate_smoother_(bitrate_smoother
           ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
               // We choose 5sec as initial time constant due to empirical data.
-              new SmoothingFilterImpl(5000, config.clock))) {
+              new SmoothingFilterImpl(5000))) {
   RTC_CHECK(RecreateEncoderInstance(config));
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const CodecInst& codec_inst)
     : AudioEncoderOpus(CreateConfig(codec_inst), nullptr) {}
 
 AudioEncoderOpus::AudioEncoderOpus(int payload_type,
                                    const SdpAudioFormat& format)
     : AudioEncoderOpus(CreateConfig(payload_type, format), nullptr) {}
 
@@ skipping 63 matching lines @@
 }
 
 void AudioEncoderOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config_;
   conf.max_playback_rate_hz = frequency_hz;
   RTC_CHECK(RecreateEncoderInstance(conf));
 }
 
 bool AudioEncoderOpus::EnableAudioNetworkAdaptor(
     const std::string& config_string,
-    RtcEventLog* event_log,
-    const Clock* clock) {
+    RtcEventLog* event_log) {
   audio_network_adaptor_ =
-      audio_network_adaptor_creator_(config_string, event_log, clock);
+      audio_network_adaptor_creator_(config_string, event_log);
   return audio_network_adaptor_.get() != nullptr;
 }
 
 void AudioEncoderOpus::DisableAudioNetworkAdaptor() {
   audio_network_adaptor_.reset(nullptr);
 }
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   if (!audio_network_adaptor_) {
@@ skipping 235 matching lines @@
     SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
   if (config.enable_dtx)
     SetDtx(*config.enable_dtx);
   if (config.num_channels)
     SetNumChannelsToEncode(*config.num_channels);
 }
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
     const ProtoString& config_string,
-    RtcEventLog* event_log,
-    const Clock* clock) const {
+    RtcEventLog* event_log) const {
   AudioNetworkAdaptorImpl::Config config;
-  config.clock = clock;
   config.event_log = event_log;
   return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
       config,
       ControllerManagerImpl::Create(
           config_string, NumChannels(), supported_frame_lengths_ms(),
           kOpusMinBitrateBps, num_channels_to_encode_, next_frame_length_ms_,
-          GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
+          GetTargetBitrate(), config_.fec_enabled, GetDtx())));
 }
 
 void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
   if (audio_network_adaptor_) {
     int64_t now_ms = rtc::TimeMillis();
     if (!bitrate_smoother_last_update_time_ ||
         now_ms - *bitrate_smoother_last_update_time_ >=
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc