Style cleanups in RtpSender.

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/base/criticalsection.h"
 #include "webrtc/base/onetimeevent.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);
   virtual ~RTPSenderAudio();
 
   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                               int8_t payloadType,
+                               int8_t payload_type,
                                uint32_t frequency,
                                size_t channels,
                                uint32_t rate,
                                RtpUtility::Payload** payload);
 
-  int32_t SendAudio(FrameType frameType,
-                    int8_t payloadType,
-                    uint32_t captureTimeStamp,
-                    const uint8_t* payloadData,
-                    size_t payloadSize,
+  int32_t SendAudio(FrameType frame_type,
+                    int8_t payload_type,
+                    uint32_t capture_timestamp,
+                    const uint8_t* payload_data,
+                    size_t payload_size,
                     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(uint16_t packetSizeSamples);
+  int32_t SetAudioPacketSize(uint16_t acket_size_samples_);

[danilchap, 2016/06/15 13:40:01]

p

[danilchap, 2016/06/15 13:41:15]

and remove _

[Sergey Ulanov, 2016/06/15 18:27:53]

Done.

 
   // 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);
+  int32_t SetAudioLevel(uint8_t level_dbov);
 
   // Send a DTMF tone using RFC 2833 (4733)
   int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
 
   int AudioFrequency() const;
 
   // Set payload type for Redundant Audio Data RFC 2198
-  int32_t SetRED(int8_t payloadType);
+  int32_t SetRED(int8_t payload_type);
 
   // Get payload type for Redundant Audio Data RFC 2198
-  int32_t RED(int8_t* payloadType) const;
+  int32_t RED(int8_t* payload_type) 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
 
-  bool MarkerBit(const FrameType frameType, const int8_t payloadType);
+  bool MarkerBit(const FrameType frame_type, const int8_t payload_type);
 
  private:
-  Clock* const _clock;
-  RTPSender* const _rtpSender;
+  Clock* const clock_;
+  RTPSender* const rtp_sender_;
 
-  rtc::CriticalSection _sendAudioCritsect;
+  rtc::CriticalSection send_audio_critsect_;
 
-  uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect);
+  uint16_t packet_size_samples_ GUARDED_BY(send_audio_critsect_);
 
   // 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;
+  bool dtmf_event_is_on_;
+  bool dtmf_event_first_packet_sent_;
+  int8_t dtmf_payload_type_ GUARDED_BY(send_audio_critsect_);
+  uint32_t dtmf_timestamp_;
+  uint8_t dtmf_key_;
+  uint32_t dtmf_length_samples_;
+  uint8_t dtmf_level_;
+  int64_t dtmf_time_last_sent_;
+  uint32_t dtmf_timestamp_last_sent_;
 
-  int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect);
+  int8_t red_payload_type_ GUARDED_BY(send_audio_critsect_);
 
-  // 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 inband_vad_active_ GUARDED_BY(send_audio_critsect_);
+  int8_t cngnb_payload_type_ GUARDED_BY(send_audio_critsect_);
+  int8_t cngwb_payload_type_ GUARDED_BY(send_audio_critsect_);
+  int8_t cngswb_payload_type_ GUARDED_BY(send_audio_critsect_);
+  int8_t cngfb_payload_type_ GUARDED_BY(send_audio_critsect_);
+  int8_t last_payload_type_ GUARDED_BY(send_audio_critsect_);
 
   // Audio level indication
   // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
-  uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect);
+  uint8_t audio_level_dbov_ GUARDED_BY(send_audio_critsect_);
   OneTimeEvent first_packet_sent_;
 };
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_