From 921d366aed151de049f86f7509f9be601746f8ed Mon Sep 17 00:00:00 2001 From: Mirko Bonadei Date: Mon, 14 Jan 2019 20:45:42 +0100 Subject: [PATCH] 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 Commit-Queue: Mirko Bonadei Cr-Commit-Position: refs/heads/master@{#26264} --- call/audio_state.h | 1 - call/call_config.h | 2 -- pc/dtmf_sender.cc | 2 -- 3 files changed, 5 deletions(-) diff --git a/call/audio_state.h b/call/audio_state.h index 1332c053f7..967314b067 100644 --- a/call/audio_state.h +++ b/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 Create( const AudioState::Config& config); diff --git a/call/call_config.h b/call/call_config.h index 67ccd51f19..260a3accdd 100644 --- a/call/call_config.h +++ b/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 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 f4c3617272..7c1d3c156a 100644 --- a/pc/dtmf_sender.cc +++ b/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(