Remove comments about using std::shared_ptr.

There are no plans to start using std::shared_ptr in WebRTC.

Bug: webrtc:10198
No-Try: True
Change-Id: I87a6c32b33b30d1b6b98eccda3400ce755a0ae95
Reviewed-on: https://webrtc-review.googlesource.com/c/117362
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26264}

call/audio_state.h

@@ -65,7 +65,6 @@ class AudioState : public rtc::RefCountInterface {
   virtual Stats GetAudioInputStats() const = 0;
   virtual void SetStereoChannelSwapping(bool enable) = 0;
 
-  // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
   static rtc::scoped_refptr<AudioState> Create(
       const AudioState::Config& config);
 

call/call_config.h

@@ -33,11 +33,9 @@ struct CallConfig {
   BitrateConstraints bitrate_config;
 
   // AudioState which is possibly shared between multiple calls.
-  // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
   rtc::scoped_refptr<AudioState> audio_state;
 
   // Audio Processing Module to be used in this call.
-  // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
   AudioProcessing* audio_processing = nullptr;
 
   // RtcEventLog to use for this call. Required.

pc/dtmf_sender.cc

@@ -77,8 +77,6 @@ DtmfSender::DtmfSender(rtc::Thread* signaling_thread,
       duration_(kDtmfDefaultDurationMs),
       inter_tone_gap_(kDtmfDefaultGapMs) {
   RTC_DCHECK(signaling_thread_);
-  // TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
-  // do that instead of relying on a "destroyed" signal.
   if (provider_) {
     RTC_DCHECK(provider_->GetOnDestroyedSignal());
     provider_->GetOnDestroyedSignal()->connect(