Simple lint fixes
BUG=webrtc:5583
Review URL: https://codereview.webrtc.org/1919133002
Cr-Commit-Position: refs/heads/master@{#12506}
diff --git a/webrtc/pc/audiomonitor.cc b/webrtc/pc/audiomonitor.cc
index 952ee9f..085ab93 100644
--- a/webrtc/pc/audiomonitor.cc
+++ b/webrtc/pc/audiomonitor.cc
@@ -101,4 +101,4 @@
return monitoring_thread_;
}
-}
+} // namespace cricket
diff --git a/webrtc/pc/audiomonitor.h b/webrtc/pc/audiomonitor.h
index 741709b..885c60f 100644
--- a/webrtc/pc/audiomonitor.h
+++ b/webrtc/pc/audiomonitor.h
@@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_AUDIOMONITOR_H_
-#define TALK_SESSION_MEDIA_AUDIOMONITOR_H_
+#ifndef WEBRTC_PC_AUDIOMONITOR_H_
+#define WEBRTC_PC_AUDIOMONITOR_H_
#include <vector>
+#include <utility>
+
#include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h"
#include "webrtc/p2p/base/port.h"
@@ -24,7 +26,7 @@
int input_level;
int output_level;
typedef std::vector<std::pair<uint32_t, int> > StreamList;
- StreamList active_streams; // ssrcs contributing to output_level
+ StreamList active_streams; // ssrcs contributing to output_level
};
class AudioMonitor : public rtc::MessageHandler,
@@ -53,6 +55,6 @@
bool monitoring_;
};
-}
+} // namespace cricket
-#endif // TALK_SESSION_MEDIA_AUDIOMONITOR_H_
+#endif // WEBRTC_PC_AUDIOMONITOR_H_
diff --git a/webrtc/pc/bundlefilter.h b/webrtc/pc/bundlefilter.h
index a698681..44a45da 100644
--- a/webrtc/pc/bundlefilter.h
+++ b/webrtc/pc/bundlefilter.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_BUNDLEFILTER_H_
-#define TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#ifndef WEBRTC_PC_BUNDLEFILTER_H_
+#define WEBRTC_PC_BUNDLEFILTER_H_
#include <stdint.h>
@@ -51,4 +51,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#endif // WEBRTC_PC_BUNDLEFILTER_H_
diff --git a/webrtc/pc/currentspeakermonitor.cc b/webrtc/pc/currentspeakermonitor.cc
index bbb33e3..e84aa82 100644
--- a/webrtc/pc/currentspeakermonitor.cc
+++ b/webrtc/pc/currentspeakermonitor.cc
@@ -10,6 +10,8 @@
#include "webrtc/pc/currentspeakermonitor.h"
+#include <vector>
+
#include "webrtc/base/logging.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/pc/audiomonitor.h"
diff --git a/webrtc/pc/currentspeakermonitor.h b/webrtc/pc/currentspeakermonitor.h
index 5ca68ae..19a61f9 100644
--- a/webrtc/pc/currentspeakermonitor.h
+++ b/webrtc/pc/currentspeakermonitor.h
@@ -11,8 +11,8 @@
// CurrentSpeakerMonitor monitors the audio levels for a session and determines
// which participant is currently speaking.
-#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
-#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
+#ifndef WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
+#define WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
#include <map>
@@ -45,7 +45,7 @@
// It's recommended that the audio monitor be started with a 100 ms period.
class CurrentSpeakerMonitor : public sigslot::has_slots<> {
public:
- CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
+ explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
~CurrentSpeakerMonitor();
void Start();
@@ -90,6 +90,6 @@
uint32_t min_time_between_switches_;
};
-}
+} // namespace cricket
-#endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
+#endif // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
diff --git a/webrtc/pc/externalhmac.h b/webrtc/pc/externalhmac.h
index 74652b0..edca74d 100644
--- a/webrtc/pc/externalhmac.h
+++ b/webrtc/pc/externalhmac.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
-#define TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
+#ifndef WEBRTC_PC_EXTERNALHMAC_H_
+#define WEBRTC_PC_EXTERNALHMAC_H_
// External libsrtp HMAC auth module which implements methods defined in
// auth_type_t.
@@ -72,4 +72,4 @@
err_status_t external_crypto_init();
#endif // defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH)
-#endif // TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_
+#endif // WEBRTC_PC_EXTERNALHMAC_H_
diff --git a/webrtc/pc/mediamonitor.cc b/webrtc/pc/mediamonitor.cc
index c9ba2d3..066094d 100644
--- a/webrtc/pc/mediamonitor.cc
+++ b/webrtc/pc/mediamonitor.cc
@@ -88,4 +88,4 @@
worker_thread_->PostDelayed(rate_, this, MSG_MONITOR_POLL);
}
-}
+} // namespace cricket
diff --git a/webrtc/pc/mediamonitor.h b/webrtc/pc/mediamonitor.h
index c2846be..d294cf1 100644
--- a/webrtc/pc/mediamonitor.h
+++ b/webrtc/pc/mediamonitor.h
@@ -10,8 +10,8 @@
// Class to collect statistics from a media channel
-#ifndef TALK_SESSION_MEDIA_MEDIAMONITOR_H_
-#define TALK_SESSION_MEDIA_MEDIAMONITOR_H_
+#ifndef WEBRTC_PC_MEDIAMONITOR_H_
+#define WEBRTC_PC_MEDIAMONITOR_H_
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/sigslot.h"
@@ -79,4 +79,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_MEDIAMONITOR_H_
+#endif // WEBRTC_PC_MEDIAMONITOR_H_
diff --git a/webrtc/pc/mediasession.cc b/webrtc/pc/mediasession.cc
index 6b05736..ea0eaa2 100644
--- a/webrtc/pc/mediasession.cc
+++ b/webrtc/pc/mediasession.cc
@@ -46,7 +46,7 @@
}
#endif
}
-}
+} // namespace
namespace cricket {
diff --git a/webrtc/pc/mediasession.h b/webrtc/pc/mediasession.h
index 98a1f07..6ac74f2 100644
--- a/webrtc/pc/mediasession.h
+++ b/webrtc/pc/mediasession.h
@@ -10,8 +10,8 @@
// Types and classes used in media session descriptions.
-#ifndef TALK_SESSION_MEDIA_MEDIASESSION_H_
-#define TALK_SESSION_MEDIA_MEDIASESSION_H_
+#ifndef WEBRTC_PC_MEDIASESSION_H_
+#define WEBRTC_PC_MEDIASESSION_H_
#include <algorithm>
#include <map>
@@ -544,4 +544,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_MEDIASESSION_H_
+#endif // WEBRTC_PC_MEDIASESSION_H_
diff --git a/webrtc/pc/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc
index 2747502..a6f6658 100644
--- a/webrtc/pc/mediasession_unittest.cc
+++ b/webrtc/pc/mediasession_unittest.cc
@@ -410,12 +410,12 @@
opts.recv_video = true;
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer.get() != NULL);
- ContentInfo* ac_offer= offer->GetContentByName("audio");
+ ContentInfo* ac_offer = offer->GetContentByName("audio");
ASSERT_TRUE(ac_offer != NULL);
AudioContentDescription* acd_offer =
static_cast<AudioContentDescription*>(ac_offer->description);
acd_offer->set_direction(direction_in_offer);
- ContentInfo* vc_offer= offer->GetContentByName("video");
+ ContentInfo* vc_offer = offer->GetContentByName("video");
ASSERT_TRUE(vc_offer != NULL);
VideoContentDescription* vcd_offer =
static_cast<VideoContentDescription*>(vc_offer->description);
@@ -889,7 +889,7 @@
f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
- ContentInfo* dc_offer= offer->GetContentByName("data");
+ ContentInfo* dc_offer = offer->GetContentByName("data");
ASSERT_TRUE(dc_offer != NULL);
DataContentDescription* dcd_offer =
static_cast<DataContentDescription*>(dc_offer->description);
diff --git a/webrtc/pc/mediasink.h b/webrtc/pc/mediasink.h
index e97a390..01bd3fa 100644
--- a/webrtc/pc/mediasink.h
+++ b/webrtc/pc/mediasink.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_MEDIASINK_H_
-#define TALK_SESSION_MEDIA_MEDIASINK_H_
+#ifndef WEBRTC_PC_MEDIASINK_H_
+#define WEBRTC_PC_MEDIASINK_H_
namespace cricket {
@@ -28,4 +28,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_MEDIASINK_H_
+#endif // WEBRTC_PC_MEDIASINK_H_
diff --git a/webrtc/pc/rtcpmuxfilter.h b/webrtc/pc/rtcpmuxfilter.h
index 272c804..94dc41d 100644
--- a/webrtc/pc/rtcpmuxfilter.h
+++ b/webrtc/pc/rtcpmuxfilter.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
-#define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
+#ifndef WEBRTC_PC_RTCPMUXFILTER_H_
+#define WEBRTC_PC_RTCPMUXFILTER_H_
#include "webrtc/base/basictypes.h"
#include "webrtc/p2p/base/sessiondescription.h"
@@ -69,4 +69,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
+#endif // WEBRTC_PC_RTCPMUXFILTER_H_
diff --git a/webrtc/pc/srtpfilter.h b/webrtc/pc/srtpfilter.h
index f4c1d33..a168710 100644
--- a/webrtc/pc/srtpfilter.h
+++ b/webrtc/pc/srtpfilter.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TALK_SESSION_MEDIA_SRTPFILTER_H_
-#define TALK_SESSION_MEDIA_SRTPFILTER_H_
+#ifndef WEBRTC_PC_SRTPFILTER_H_
+#define WEBRTC_PC_SRTPFILTER_H_
#include <list>
#include <map>
@@ -310,4 +310,4 @@
} // namespace cricket
-#endif // TALK_SESSION_MEDIA_SRTPFILTER_H_
+#endif // WEBRTC_PC_SRTPFILTER_H_
diff --git a/webrtc/pc/voicechannel.h b/webrtc/pc/voicechannel.h
index 9b6f16e..78524ab 100644
--- a/webrtc/pc/voicechannel.h
+++ b/webrtc/pc/voicechannel.h
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef _VOICECHANNEL_H_
-#define _VOICECHANNEL_H_
+#ifndef WEBRTC_PC_VOICECHANNEL_H_
+#define WEBRTC_PC_VOICECHANNEL_H_
#include "webrtc/pc/channel.h"
-#endif // _VOICECHANNEL_H_
+#endif // WEBRTC_PC_VOICECHANNEL_H_
