[rtp_rtcp] lint whitespace warning removed from most source/ files

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/dtmf_queue.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
-class RTPSenderAudio: public DTMFqueue
-{
-public:
- RTPSenderAudio(Clock* clock,
-                RTPSender* rtpSender,
-                RtpAudioFeedback* audio_feedback);
-    virtual ~RTPSenderAudio();
+class RTPSenderAudio : public DTMFqueue {
+ public:
+  RTPSenderAudio(Clock* clock,
+                 RTPSender* rtpSender,
+                 RtpAudioFeedback* audio_feedback);
+  virtual ~RTPSenderAudio();
 
-    int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                                 const int8_t payloadType,
-                                 const uint32_t frequency,
-                                 const uint8_t channels,
-                                 const uint32_t rate,
-                                 RtpUtility::Payload*& payload);
+  int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+                               int8_t payloadType,
+                               uint32_t frequency,
+                               uint8_t channels,
+                               uint32_t rate,
+                               RtpUtility::Payload*& payload);
 
-    int32_t SendAudio(const FrameType frameType,
-                      const int8_t payloadType,
-                      const uint32_t captureTimeStamp,
-                      const uint8_t* payloadData,
-                      const size_t payloadSize,
-                      const RTPFragmentationHeader* fragmentation);
+  int32_t SendAudio(FrameType frameType,
+                    int8_t payloadType,
+                    uint32_t captureTimeStamp,
+                    const uint8_t* payloadData,
+                    size_t payloadSize,
+                    const RTPFragmentationHeader* fragmentation);
 
-    // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-    int32_t SetAudioPacketSize(const uint16_t packetSizeSamples);
+  // set audio packet size, used to determine when it's time to send a DTMF
+  // packet in silence (CNG)
+  int32_t SetAudioPacketSize(uint16_t packetSizeSamples);
 
-    // Store the audio level in dBov for header-extension-for-audio-level-indication.
-    // Valid range is [0,100]. Actual value is negative.
-    int32_t SetAudioLevel(const uint8_t level_dBov);
+  // Store the audio level in dBov for
+  // header-extension-for-audio-level-indication.
+  // Valid range is [0,100]. Actual value is negative.
+  int32_t SetAudioLevel(uint8_t level_dBov);
 
-    // Send a DTMF tone using RFC 2833 (4733)
-    int32_t SendTelephoneEvent(const uint8_t key,
-                               const uint16_t time_ms,
-                               const uint8_t level);
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
-    int AudioFrequency() const;
+  int AudioFrequency() const;
 
-    // Set payload type for Redundant Audio Data RFC 2198
-    int32_t SetRED(const int8_t payloadType);
+  // Set payload type for Redundant Audio Data RFC 2198
+  int32_t SetRED(int8_t payloadType);
 
-    // Get payload type for Redundant Audio Data RFC 2198
-    int32_t RED(int8_t& payloadType) const;
+  // Get payload type for Redundant Audio Data RFC 2198
+  int32_t RED(int8_t& payloadType) const;
 
-protected:
-    int32_t SendTelephoneEventPacket(bool ended,
-                                     int8_t dtmf_payload_type,
-                                     uint32_t dtmfTimeStamp,
-                                     uint16_t duration,
-                                     bool markerBit); // set on first packet in talk burst
+ protected:
+  int32_t SendTelephoneEventPacket(
+      bool ended,
+      int8_t dtmf_payload_type,
+      uint32_t dtmfTimeStamp,
+      uint16_t duration,
+      bool markerBit);  // set on first packet in talk burst
 
-    bool MarkerBit(const FrameType frameType,
-                   const int8_t payloadType);
+  bool MarkerBit(const FrameType frameType, const int8_t payloadType);
 
-private:
- Clock* const _clock;
- RTPSender* const _rtpSender;
- RtpAudioFeedback* const _audioFeedback;
+ private:
+  Clock* const _clock;
+  RTPSender* const _rtpSender;
+  RtpAudioFeedback* const _audioFeedback;
 
- rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
+  rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;
 
- uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
+  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
 
- // DTMF
- bool _dtmfEventIsOn;
- bool _dtmfEventFirstPacketSent;
- int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
- uint32_t _dtmfTimestamp;
- uint8_t _dtmfKey;
- uint32_t _dtmfLengthSamples;
- uint8_t _dtmfLevel;
- int64_t _dtmfTimeLastSent;
- uint32_t _dtmfTimestampLastSent;
+  // DTMF
+  bool _dtmfEventIsOn;
+  bool _dtmfEventFirstPacketSent;
+  int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect);
+  uint32_t _dtmfTimestamp;
+  uint8_t _dtmfKey;
+  uint32_t _dtmfLengthSamples;
+  uint8_t _dtmfLevel;
+  int64_t _dtmfTimeLastSent;
+  uint32_t _dtmfTimestampLastSent;
 
- int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
 
- // VAD detection, used for markerbit
- bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
- int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
- int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
+  // VAD detection, used for markerbit
+  bool _inbandVADactive GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect);
 
- // Audio level indication
- // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
- uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+  // Audio level indication
+  // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
+  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
 };
 }  // namespace webrtc
 
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_