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Simple lint fixes

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BUG=webrtc:5583

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

Cr-Commit-Position: refs/heads/master@{#12506}
This commit is contained in:
terelius
2016-04-26 05:28:11 -07:00
committed by Commit bot
parent b67aba81e5
commit 8c011e5ae6
15 changed files with 44 additions and 40 deletions
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2
webrtc/pc/audiomonitor.cc
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@ -101,4 +101,4 @@ rtc::Thread *AudioMonitor::monitor_thread() {
return monitoring_thread_; return monitoring_thread_;
} }
} } // namespace cricket

12
webrtc/pc/audiomonitor.h
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@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_AUDIOMONITOR_H_ #ifndef WEBRTC_PC_AUDIOMONITOR_H_
#define TALK_SESSION_MEDIA_AUDIOMONITOR_H_ #define WEBRTC_PC_AUDIOMONITOR_H_
#include <vector> #include <vector>
#include <utility>
#include "webrtc/base/sigslot.h" #include "webrtc/base/sigslot.h"
#include "webrtc/base/thread.h" #include "webrtc/base/thread.h"
#include "webrtc/p2p/base/port.h" #include "webrtc/p2p/base/port.h"
@ -24,7 +26,7 @@ struct AudioInfo {
int input_level; int input_level;
int output_level; int output_level;
typedef std::vector<std::pair<uint32_t, int> > StreamList; 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, class AudioMonitor : public rtc::MessageHandler,
@ -53,6 +55,6 @@ class AudioMonitor : public rtc::MessageHandler,
bool monitoring_; bool monitoring_;
}; };
} } // namespace cricket
#endif // TALK_SESSION_MEDIA_AUDIOMONITOR_H_ #endif // WEBRTC_PC_AUDIOMONITOR_H_

6
webrtc/pc/bundlefilter.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_BUNDLEFILTER_H_ #ifndef WEBRTC_PC_BUNDLEFILTER_H_
#define TALK_SESSION_MEDIA_BUNDLEFILTER_H_ #define WEBRTC_PC_BUNDLEFILTER_H_
#include <stdint.h> #include <stdint.h>
@ -51,4 +51,4 @@ class BundleFilter {
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_BUNDLEFILTER_H_ #endif // WEBRTC_PC_BUNDLEFILTER_H_

2
webrtc/pc/currentspeakermonitor.cc
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@ -10,6 +10,8 @@
#include "webrtc/pc/currentspeakermonitor.h" #include "webrtc/pc/currentspeakermonitor.h"
#include <vector>
#include "webrtc/base/logging.h" #include "webrtc/base/logging.h"
#include "webrtc/media/base/streamparams.h" #include "webrtc/media/base/streamparams.h"
#include "webrtc/pc/audiomonitor.h" #include "webrtc/pc/audiomonitor.h"

10
webrtc/pc/currentspeakermonitor.h
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@ -11,8 +11,8 @@
// CurrentSpeakerMonitor monitors the audio levels for a session and determines // CurrentSpeakerMonitor monitors the audio levels for a session and determines
// which participant is currently speaking. // which participant is currently speaking.
#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_ #ifndef WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_ #define WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
#include <map> #include <map>
@ -45,7 +45,7 @@ class AudioSourceContext {
// It's recommended that the audio monitor be started with a 100 ms period. // It's recommended that the audio monitor be started with a 100 ms period.
class CurrentSpeakerMonitor : public sigslot::has_slots<> { class CurrentSpeakerMonitor : public sigslot::has_slots<> {
public: public:
CurrentSpeakerMonitor(AudioSourceContext* audio_source_context); explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
~CurrentSpeakerMonitor(); ~CurrentSpeakerMonitor();
void Start(); void Start();
@ -90,6 +90,6 @@ class CurrentSpeakerMonitor : public sigslot::has_slots<> {
uint32_t min_time_between_switches_; uint32_t min_time_between_switches_;
}; };
} } // namespace cricket
#endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_ #endif // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_

6
webrtc/pc/externalhmac.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_ #ifndef WEBRTC_PC_EXTERNALHMAC_H_
#define TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_ #define WEBRTC_PC_EXTERNALHMAC_H_
// External libsrtp HMAC auth module which implements methods defined in // External libsrtp HMAC auth module which implements methods defined in
// auth_type_t. // auth_type_t.
@ -72,4 +72,4 @@ err_status_t external_hmac_compute(ExternalHmacContext* state,
err_status_t external_crypto_init(); err_status_t external_crypto_init();
#endif // defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH) #endif // defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH)
#endif // TALK_SESSION_MEDIA_EXTERNAL_HMAC_H_ #endif // WEBRTC_PC_EXTERNALHMAC_H_

2
webrtc/pc/mediamonitor.cc
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@ -88,4 +88,4 @@ void MediaMonitor::PollMediaChannel() {
worker_thread_->PostDelayed(rate_, this, MSG_MONITOR_POLL); worker_thread_->PostDelayed(rate_, this, MSG_MONITOR_POLL);
} }
} } // namespace cricket

6
webrtc/pc/mediamonitor.h
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@ -10,8 +10,8 @@
// Class to collect statistics from a media channel // Class to collect statistics from a media channel
#ifndef TALK_SESSION_MEDIA_MEDIAMONITOR_H_ #ifndef WEBRTC_PC_MEDIAMONITOR_H_
#define TALK_SESSION_MEDIA_MEDIAMONITOR_H_ #define WEBRTC_PC_MEDIAMONITOR_H_
#include "webrtc/base/criticalsection.h" #include "webrtc/base/criticalsection.h"
#include "webrtc/base/sigslot.h" #include "webrtc/base/sigslot.h"
@ -79,4 +79,4 @@ typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIAMONITOR_H_ #endif // WEBRTC_PC_MEDIAMONITOR_H_

2
webrtc/pc/mediasession.cc
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@ -46,7 +46,7 @@ void GetSupportedCryptoSuiteNames(void (*func)(std::vector<int>*),
} }
#endif #endif
} }
} } // namespace
namespace cricket { namespace cricket {

6
webrtc/pc/mediasession.h
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@ -10,8 +10,8 @@
// Types and classes used in media session descriptions. // Types and classes used in media session descriptions.
#ifndef TALK_SESSION_MEDIA_MEDIASESSION_H_ #ifndef WEBRTC_PC_MEDIASESSION_H_
#define TALK_SESSION_MEDIA_MEDIASESSION_H_ #define WEBRTC_PC_MEDIASESSION_H_
#include <algorithm> #include <algorithm>
#include <map> #include <map>
@ -544,4 +544,4 @@ void GetDefaultSrtpCryptoSuiteNames(
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIASESSION_H_ #endif // WEBRTC_PC_MEDIASESSION_H_

6
webrtc/pc/mediasession_unittest.cc
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@ -410,12 +410,12 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
opts.recv_video = true; opts.recv_video = true;
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL)); std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
ASSERT_TRUE(offer.get() != NULL); ASSERT_TRUE(offer.get() != NULL);
ContentInfo* ac_offer= offer->GetContentByName("audio"); ContentInfo* ac_offer = offer->GetContentByName("audio");
ASSERT_TRUE(ac_offer != NULL); ASSERT_TRUE(ac_offer != NULL);
AudioContentDescription* acd_offer = AudioContentDescription* acd_offer =
static_cast<AudioContentDescription*>(ac_offer->description); static_cast<AudioContentDescription*>(ac_offer->description);
acd_offer->set_direction(direction_in_offer); acd_offer->set_direction(direction_in_offer);
ContentInfo* vc_offer= offer->GetContentByName("video"); ContentInfo* vc_offer = offer->GetContentByName("video");
ASSERT_TRUE(vc_offer != NULL); ASSERT_TRUE(vc_offer != NULL);
VideoContentDescription* vcd_offer = VideoContentDescription* vcd_offer =
static_cast<VideoContentDescription*>(vc_offer->description); static_cast<VideoContentDescription*>(vc_offer->description);
@ -889,7 +889,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
f1_.set_secure(SEC_ENABLED); f1_.set_secure(SEC_ENABLED);
f2_.set_secure(SEC_ENABLED); f2_.set_secure(SEC_ENABLED);
std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL)); 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); ASSERT_TRUE(dc_offer != NULL);
DataContentDescription* dcd_offer = DataContentDescription* dcd_offer =
static_cast<DataContentDescription*>(dc_offer->description); static_cast<DataContentDescription*>(dc_offer->description);

6
webrtc/pc/mediasink.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_MEDIASINK_H_ #ifndef WEBRTC_PC_MEDIASINK_H_
#define TALK_SESSION_MEDIA_MEDIASINK_H_ #define WEBRTC_PC_MEDIASINK_H_
namespace cricket { namespace cricket {
@ -28,4 +28,4 @@ class MediaSinkInterface {
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIASINK_H_ #endif // WEBRTC_PC_MEDIASINK_H_

6
webrtc/pc/rtcpmuxfilter.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_RTCPMUXFILTER_H_ #ifndef WEBRTC_PC_RTCPMUXFILTER_H_
#define TALK_SESSION_MEDIA_RTCPMUXFILTER_H_ #define WEBRTC_PC_RTCPMUXFILTER_H_
#include "webrtc/base/basictypes.h" #include "webrtc/base/basictypes.h"
#include "webrtc/p2p/base/sessiondescription.h" #include "webrtc/p2p/base/sessiondescription.h"
@ -69,4 +69,4 @@ class RtcpMuxFilter {
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_RTCPMUXFILTER_H_ #endif // WEBRTC_PC_RTCPMUXFILTER_H_

6
webrtc/pc/srtpfilter.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef TALK_SESSION_MEDIA_SRTPFILTER_H_ #ifndef WEBRTC_PC_SRTPFILTER_H_
#define TALK_SESSION_MEDIA_SRTPFILTER_H_ #define WEBRTC_PC_SRTPFILTER_H_
#include <list> #include <list>
#include <map> #include <map>
@ -310,4 +310,4 @@ class SrtpStat {
} // namespace cricket } // namespace cricket
#endif // TALK_SESSION_MEDIA_SRTPFILTER_H_ #endif // WEBRTC_PC_SRTPFILTER_H_

6
webrtc/pc/voicechannel.h
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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef _VOICECHANNEL_H_ #ifndef WEBRTC_PC_VOICECHANNEL_H_
#define _VOICECHANNEL_H_ #define WEBRTC_PC_VOICECHANNEL_H_
#include "webrtc/pc/channel.h" #include "webrtc/pc/channel.h"
#endif // _VOICECHANNEL_H_ #endif // WEBRTC_PC_VOICECHANNEL_H_