Simple lint fixes

BUG=webrtc:5583

Review URL: https://codereview.webrtc.org/1919133002

Cr-Commit-Position: refs/heads/master@{#12506}

webrtc/pc/audiomonitor.cc

@@ -101,4 +101,4 @@ rtc::Thread *AudioMonitor::monitor_thread() {
   return monitoring_thread_;
 }
 
-}
+}  // namespace cricket

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 @@ struct AudioInfo {
   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 @@ class AudioMonitor : public rtc::MessageHandler,
   bool monitoring_;
 };
 
-}
+}  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_AUDIOMONITOR_H_
+#endif  // WEBRTC_PC_AUDIOMONITOR_H_

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 @@ class BundleFilter {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#endif  // WEBRTC_PC_BUNDLEFILTER_H_

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"

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 @@ class AudioSourceContext {
 // 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 @@ class CurrentSpeakerMonitor : public sigslot::has_slots<> {
   uint32_t min_time_between_switches_;
 };
 
-}
+}  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
+#endif  // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_

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_hmac_compute(ExternalHmacContext* state,
 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_

webrtc/pc/mediamonitor.cc

@@ -88,4 +88,4 @@ void MediaMonitor::PollMediaChannel() {
   worker_thread_->PostDelayed(rate_, this, MSG_MONITOR_POLL);
 }
 
-}
+}  // namespace cricket

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 @@ typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIAMONITOR_H_
+#endif  // WEBRTC_PC_MEDIAMONITOR_H_

webrtc/pc/mediasession.cc

@@ -46,7 +46,7 @@ void GetSupportedCryptoSuiteNames(void (*func)(std::vector<int>*),
   }
 #endif
 }
-}
+}  // namespace
 
 namespace cricket {
 

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 @@ void GetDefaultSrtpCryptoSuiteNames(
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIASESSION_H_
+#endif  // WEBRTC_PC_MEDIASESSION_H_

webrtc/pc/mediasession_unittest.cc

@@ -410,12 +410,12 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
     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 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   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);

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 @@ class MediaSinkInterface {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_MEDIASINK_H_
+#endif  // WEBRTC_PC_MEDIASINK_H_

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 @@ class RtcpMuxFilter {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_RTCPMUXFILTER_H_
+#endif  // WEBRTC_PC_RTCPMUXFILTER_H_

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 @@ class SrtpStat {
 
 }  // namespace cricket
 
-#endif  // TALK_SESSION_MEDIA_SRTPFILTER_H_
+#endif  // WEBRTC_PC_SRTPFILTER_H_

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_