Rename elad.alon to eladalon, to avoid confusion between repositories.

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 18 matching lines @@
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 namespace internal {
-// TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
+// TODO(eladalon): Subsequent CL will make these values experiment-dependent.
 constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
 constexpr size_t kPacketLossRateMinNumAckedPackets = 50;
 constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
 
 namespace {
 void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
                  rtc::FunctionView<void(AudioEncoder*)> lambda) {
   channel_proxy->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
     RTC_DCHECK(encoder_ptr);
     lambda(encoder_ptr->get());
@@ skipping 283 matching lines @@
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   return 0;
 }
 
 void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   // Only packets that belong to this stream are of interest.
   if (ssrc == config_.rtp.ssrc) {
     rtc::CritScope lock(&packet_loss_tracker_cs_);
-    // TODO(elad.alon): This function call could potentially reset the window,
+    // TODO(eladalon): This function call could potentially reset the window,
     // setting both PLR and RPLR to unknown. Consider (during upcoming
     // refactoring) passing an indication of such an event.
     packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis());
   }
 }
 
 void AudioSendStream::OnPacketFeedbackVector(
     const std::vector<PacketFeedback>& packet_feedback_vector) {
-  // TODO(elad.alon): This fails in UT; fix and uncomment.
+  // TODO(eladalon): This fails in UT; fix and uncomment.
   // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7405
   // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   rtc::Optional<float> plr;
   rtc::Optional<float> rplr;
   {
     rtc::CritScope lock(&packet_loss_tracker_cs_);
     packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
     plr = packet_loss_tracker_.GetPacketLossRate();
     rplr = packet_loss_tracker_.GetRecoverablePacketLossRate();
   }
-  // TODO(elad.alon): If R/PLR go back to unknown, no indication is given that
+  // TODO(eladalon): If R/PLR go back to unknown, no indication is given that
   // the previously sent value is no longer relevant. This will be taken care
   // of with some refactoring which is now being done.
   if (plr) {
     channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
   }
   if (rplr) {
     channel_proxy_->OnRecoverableUplinkPacketLossRate(*rplr);
   }
 }
 
@@ skipping 232 matching lines @@
     if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc