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}diff --git a/call/audio_state.h b/call/audio_state.h
index 1332c05..967314b 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -65,7 +65,6 @@
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);
diff --git a/call/call_config.h b/call/call_config.h
index 67ccd51..260a3ac 100644
--- a/call/call_config.h
+++ b/call/call_config.h
@@ -33,11 +33,9 @@
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.
diff --git a/pc/dtmf_sender.cc b/pc/dtmf_sender.cc
index f4c3617..7c1d3c1 100644
--- a/pc/dtmf_sender.cc
+++ b/pc/dtmf_sender.cc
@@ -77,8 +77,6 @@
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(