Replace NULL with nullptr or null in webrtc/api/.
BUG=webrtc:7147
Review-Url: https://codereview.webrtc.org/2715103002
Cr-Commit-Position: refs/heads/master@{#16880}
diff --git a/webrtc/api/jsep.h b/webrtc/api/jsep.h
index 5ebf7c1..aa6e9a1 100644
--- a/webrtc/api/jsep.h
+++ b/webrtc/api/jsep.h
@@ -68,8 +68,8 @@
};
// Creates a IceCandidateInterface based on SDP string.
-// Returns NULL if the sdp string can't be parsed.
-// |error| may be NULL.
+// Returns null if the sdp string can't be parsed.
+// |error| may be null.
IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& sdp,
@@ -141,8 +141,8 @@
};
// Creates a SessionDescriptionInterface based on the SDP string and the type.
-// Returns NULL if the sdp string can't be parsed or the type is unsupported.
-// |error| may be NULL.
+// Returns null if the sdp string can't be parsed or the type is unsupported.
+// |error| may be null.
SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
const std::string& sdp,
SdpParseError* error);
diff --git a/webrtc/api/jsepicecandidate.h b/webrtc/api/jsepicecandidate.h
index a8919e0..3566013 100644
--- a/webrtc/api/jsepicecandidate.h
+++ b/webrtc/api/jsepicecandidate.h
@@ -31,7 +31,7 @@
JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
const cricket::Candidate& candidate);
~JsepIceCandidate();
- // |err| may be NULL.
+ // |err| may be null.
bool Initialize(const std::string& sdp, SdpParseError* err);
void SetCandidate(const cricket::Candidate& candidate) {
candidate_ = candidate;
diff --git a/webrtc/api/jsepsessiondescription.h b/webrtc/api/jsepsessiondescription.h
index b5e1b4f..e4855b9 100644
--- a/webrtc/api/jsepsessiondescription.h
+++ b/webrtc/api/jsepsessiondescription.h
@@ -35,7 +35,7 @@
explicit JsepSessionDescription(const std::string& type);
virtual ~JsepSessionDescription();
- // |error| may be NULL.
+ // |error| may be null.
bool Initialize(const std::string& sdp, SdpParseError* error);
// Takes ownership of |description|.
diff --git a/webrtc/api/mediastreaminterface.h b/webrtc/api/mediastreaminterface.h
index 59ca66c..2d16e52 100644
--- a/webrtc/api/mediastreaminterface.h
+++ b/webrtc/api/mediastreaminterface.h
@@ -260,7 +260,7 @@
// virtual after it's implemented in chromium.
virtual bool GetSignalLevel(int* level) { return false; }
- // Get the audio processor used by the audio track. Return NULL if the track
+ // Get the audio processor used by the audio track. Return null if the track
// does not have any processor.
// TODO(deadbeef): Make the interface pure virtual.
virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() {
diff --git a/webrtc/api/notifier.h b/webrtc/api/notifier.h
index 878d01c..d883051 100644
--- a/webrtc/api/notifier.h
+++ b/webrtc/api/notifier.h
@@ -27,7 +27,7 @@
}
virtual void RegisterObserver(ObserverInterface* observer) {
- RTC_DCHECK(observer != NULL);
+ RTC_DCHECK(observer != nullptr);
observers_.push_back(observer);
}
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index 69bae29..165f2d5 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -532,7 +532,7 @@
return false;
}
- // Returns pointer to a DtmfSender on success. Otherwise returns NULL.
+ // Returns pointer to a DtmfSender on success. Otherwise returns null.
//
// This API is no longer part of the standard; instead DtmfSenders are
// obtained from RtpSenders. Which is what the implementation does; it finds
@@ -922,8 +922,8 @@
}
// A video source creator that allows selection of resolution and frame rate.
- // |constraints| decides video resolution and frame rate but can be NULL.
- // In the NULL case, use the version above.
+ // |constraints| decides video resolution and frame rate but can be null.
+ // In the null case, use the version above.
//
// |constraints| is only used for the invocation of this method, and can
// safely be destroyed afterwards.
@@ -952,7 +952,7 @@
const std::string& label,
VideoTrackSourceInterface* source) = 0;
- // Creates an new AudioTrack. At the moment |source| can be NULL.
+ // Creates an new AudioTrack. At the moment |source| can be null.
virtual rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& label,
AudioSourceInterface* source) = 0;
diff --git a/webrtc/api/proxy.h b/webrtc/api/proxy.h
index 46c424d..862de87 100644
--- a/webrtc/api/proxy.h
+++ b/webrtc/api/proxy.h
@@ -128,7 +128,7 @@
void Invoke(const rtc::Location& posted_from, rtc::Thread* t) {
if (t->IsCurrent()) {
- proxy_->OnMessage(NULL);
+ proxy_->OnMessage(nullptr);
} else {
e_.reset(new rtc::Event(false, false));
t->Post(posted_from, this, 0);
@@ -137,7 +137,10 @@
}
private:
- void OnMessage(rtc::Message*) { proxy_->OnMessage(NULL); e_->Set(); }
+ void OnMessage(rtc::Message*) {
+ proxy_->OnMessage(nullptr);
+ e_->Set();
+ }
std::unique_ptr<rtc::Event> e_;
rtc::MessageHandler* proxy_;
};
diff --git a/webrtc/api/statstypes.cc b/webrtc/api/statstypes.cc
index 300bb4a..b7459fa 100644
--- a/webrtc/api/statstypes.cc
+++ b/webrtc/api/statstypes.cc
@@ -779,7 +779,7 @@
return InsertNew(id);
}
-// Looks for a report with the given |id|. If one is not found, NULL
+// Looks for a report with the given |id|. If one is not found, null
// will be returned.
StatsReport* StatsCollection::Find(const StatsReport::Id& id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
diff --git a/webrtc/api/statstypes.h b/webrtc/api/statstypes.h
index 40d26ab..8ef79d9 100644
--- a/webrtc/api/statstypes.h
+++ b/webrtc/api/statstypes.h
@@ -415,7 +415,7 @@
StatsReport* FindOrAddNew(const StatsReport::Id& id);
StatsReport* ReplaceOrAddNew(const StatsReport::Id& id);
- // Looks for a report with the given |id|. If one is not found, NULL
+ // Looks for a report with the given |id|. If one is not found, null
// will be returned.
StatsReport* Find(const StatsReport::Id& id);
