Style cleanups in RtpSender.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 4236e1f37d428f833fc6d1bbed4b62c5e8b292f4..e49ff69e1a4b33f98ea12eb35cc7ac9f5a27be76 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -22,27 +22,27 @@ namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
-RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
-    : _clock(clock),
-      _rtpSender(rtpSender),
-      _packetSizeSamples(160),
-      _dtmfEventIsOn(false),
-      _dtmfEventFirstPacketSent(false),
-      _dtmfPayloadType(-1),
-      _dtmfTimestamp(0),
-      _dtmfKey(0),
-      _dtmfLengthSamples(0),
-      _dtmfLevel(0),
-      _dtmfTimeLastSent(0),
-      _dtmfTimestampLastSent(0),
-      _REDPayloadType(-1),
-      _inbandVADactive(false),
-      _cngNBPayloadType(-1),
-      _cngWBPayloadType(-1),
-      _cngSWBPayloadType(-1),
-      _cngFBPayloadType(-1),
-      _lastPayloadType(-1),
-      _audioLevel_dBov(0) {}
+RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
+    : clock_(clock),
+      rtp_sender_(rtp_sender),
+      packet_size_samples_(160),
+      dtmf_event_is_on_(false),
+      dtmf_event_first_packet_sent_(false),
+      dtmf_payload_type_(-1),
+      dtmf_timestamp_(0),
+      dtmf_key_(0),
+      dtmf_length_samples_(0),
+      dtmf_level_(0),
+      dtmf_time_last_sent_(0),
+      dtmf_timestamp_last_sent_(0),
+      red_payload_type_(-1),
+      inband_vad_active_(false),
+      cngnb_payload_type_(-1),
+      cngwb_payload_type_(-1),
+      cngswb_payload_type_(-1),
+      cngfb_payload_type_(-1),
+      last_payload_type_(-1),
+      audio_level_dbov_(0) {}
 
 RTPSenderAudio::~RTPSenderAudio() {}
 
@@ -52,44 +52,43 @@ int RTPSenderAudio::AudioFrequency() const {
 
 // set audio packet size, used to determine when it's time to send a DTMF packet
 // in silence (CNG)
-int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packetSizeSamples) {
-  rtc::CritScope cs(&_sendAudioCritsect);
-
-  _packetSizeSamples = packetSizeSamples;
+int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
+  rtc::CritScope cs(&send_audio_critsect_);
+  packet_size_samples_ = packet_size_samples;
   return 0;
 }
 
 int32_t RTPSenderAudio::RegisterAudioPayload(
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-    const int8_t payloadType,
+    const int8_t payload_type,
     const uint32_t frequency,
     const size_t channels,
     const uint32_t rate,
     RtpUtility::Payload** payload) {
   if (RtpUtility::StringCompare(payloadName, "cn", 2)) {
-    rtc::CritScope cs(&_sendAudioCritsect);
+    rtc::CritScope cs(&send_audio_critsect_);
     //  we can have multiple CNG payload types
     switch (frequency) {
       case 8000:
-        _cngNBPayloadType = payloadType;
+        cngnb_payload_type_ = payload_type;
         break;
       case 16000:
-        _cngWBPayloadType = payloadType;
+        cngwb_payload_type_ = payload_type;
         break;
       case 32000:
-        _cngSWBPayloadType = payloadType;
+        cngswb_payload_type_ = payload_type;
         break;
       case 48000:
-        _cngFBPayloadType = payloadType;
+        cngfb_payload_type_ = payload_type;
         break;
       default:
         return -1;
     }
   } else if (RtpUtility::StringCompare(payloadName, "telephone-event", 15)) {
-    rtc::CritScope cs(&_sendAudioCritsect);
+    rtc::CritScope cs(&send_audio_critsect_);
     // Don't add it to the list
     // we dont want to allow send with a DTMF payloadtype
-    _dtmfPayloadType = payloadType;
+    dtmf_payload_type_ = payload_type;
     return 0;
     // The default timestamp rate is 8000 Hz, but other rates may be defined.
   }
@@ -103,27 +102,27 @@ int32_t RTPSenderAudio::RegisterAudioPayload(
   return 0;
 }
 
-bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
-  rtc::CritScope cs(&_sendAudioCritsect);
+bool RTPSenderAudio::MarkerBit(FrameType frame_type, int8_t payload_type) {
+  rtc::CritScope cs(&send_audio_critsect_);
   // for audio true for first packet in a speech burst
   bool markerBit = false;
-  if (_lastPayloadType != payload_type) {
-    if (payload_type != -1 && (_cngNBPayloadType == payload_type ||
-                               _cngWBPayloadType == payload_type ||
-                               _cngSWBPayloadType == payload_type ||
-                               _cngFBPayloadType == payload_type)) {
+  if (last_payload_type_ != payload_type) {
+    if (payload_type != -1 && (cngnb_payload_type_ == payload_type ||
+                               cngwb_payload_type_ == payload_type ||
+                               cngswb_payload_type_ == payload_type ||
+                               cngfb_payload_type_ == payload_type)) {
       // Only set a marker bit when we change payload type to a non CNG
       return false;
     }
 
     // payload_type differ
-    if (_lastPayloadType == -1) {
-      if (frameType != kAudioFrameCN) {
+    if (last_payload_type_ == -1) {
+      if (frame_type != kAudioFrameCN) {
         // first packet and NOT CNG
         return true;
       } else {
         // first packet and CNG
-        _inbandVADactive = true;
+        inband_vad_active_ = true;
         return false;
       }
     }
@@ -137,110 +136,111 @@ bool RTPSenderAudio::MarkerBit(FrameType frameType, int8_t payload_type) {
   }
 
   // For G.723 G.729, AMR etc we can have inband VAD
-  if (frameType == kAudioFrameCN) {
-    _inbandVADactive = true;
-  } else if (_inbandVADactive) {
-    _inbandVADactive = false;
+  if (frame_type == kAudioFrameCN) {
+    inband_vad_active_ = true;
+  } else if (inband_vad_active_) {
+    inband_vad_active_ = false;
     markerBit = true;
   }
   return markerBit;
 }
 
-int32_t RTPSenderAudio::SendAudio(FrameType frameType,
-                                  int8_t payloadType,
-                                  uint32_t captureTimeStamp,
-                                  const uint8_t* payloadData,
-                                  size_t dataSize,
+int32_t RTPSenderAudio::SendAudio(FrameType frame_type,
+                                  int8_t payload_type,
+                                  uint32_t capture_timestamp,
+                                  const uint8_t* payload_data,
+                                  size_t data_size,
                                   const RTPFragmentationHeader* fragmentation) {
   // TODO(pwestin) Breakup function in smaller functions.
-  size_t payloadSize = dataSize;
-  size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
-  uint16_t dtmfLengthMS = 0;
+  size_t payload_size = data_size;
+  size_t max_payload_length = rtp_sender_->MaxPayloadLength();
+  uint16_t dtmf_length_ms = 0;
   uint8_t key = 0;
   int red_payload_type;
   uint8_t audio_level_dbov;
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
-    rtc::CritScope cs(&_sendAudioCritsect);
-    red_payload_type = _REDPayloadType;
-    audio_level_dbov = _audioLevel_dBov;
-    dtmf_payload_type = _dtmfPayloadType;
-    packet_size_samples = _packetSizeSamples;
+    rtc::CritScope cs(&send_audio_critsect_);
+    red_payload_type = red_payload_type_;
+    audio_level_dbov = audio_level_dbov_;
+    dtmf_payload_type = dtmf_payload_type_;
+    packet_size_samples = packet_size_samples_;
   }
 
   // Check if we have pending DTMFs to send
-  if (!_dtmfEventIsOn && PendingDTMF()) {
+  if (!dtmf_event_is_on_ && PendingDTMF()) {
     int64_t delaySinceLastDTMF =
-        _clock->TimeInMilliseconds() - _dtmfTimeLastSent;
+        clock_->TimeInMilliseconds() - dtmf_time_last_sent_;
 
     if (delaySinceLastDTMF > 100) {
       // New tone to play
-      _dtmfTimestamp = captureTimeStamp;
-      if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0) {
-        _dtmfEventFirstPacketSent = false;
-        _dtmfKey = key;
-        _dtmfLengthSamples = (kDtmfFrequencyHz / 1000) * dtmfLengthMS;
-        _dtmfEventIsOn = true;
+      dtmf_timestamp_ = capture_timestamp;
+      if (NextDTMF(&key, &dtmf_length_ms, &dtmf_level_) >= 0) {
+        dtmf_event_first_packet_sent_ = false;
+        dtmf_key_ = key;
+        dtmf_length_samples_ = (kDtmfFrequencyHz / 1000) * dtmf_length_ms;
+        dtmf_event_is_on_ = true;
       }
     }
   }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it
-  if (_dtmfEventIsOn) {
-    if (frameType == kEmptyFrame) {
+  if (dtmf_event_is_on_) {
+    if (frame_type == kEmptyFrame) {
       // kEmptyFrame is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
-      if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
+      if (packet_size_samples >
+          (capture_timestamp - dtmf_timestamp_last_sent_)) {
         // not time to send yet
         return 0;
       }
     }
-    _dtmfTimestampLastSent = captureTimeStamp;
-    uint32_t dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
+    dtmf_timestamp_last_sent_ = capture_timestamp;
+    uint32_t dtmf_duration_samples = capture_timestamp - dtmf_timestamp_;
     bool ended = false;
     bool send = true;
 
-    if (_dtmfLengthSamples > dtmfDurationSamples) {
-      if (dtmfDurationSamples <= 0) {
+    if (dtmf_length_samples_ > dtmf_duration_samples) {
+      if (dtmf_duration_samples <= 0) {
         // Skip send packet at start, since we shouldn't use duration 0
         send = false;
       }
     } else {
       ended = true;
-      _dtmfEventIsOn = false;
-      _dtmfTimeLastSent = _clock->TimeInMilliseconds();
+      dtmf_event_is_on_ = false;
+      dtmf_time_last_sent_ = clock_->TimeInMilliseconds();
     }
     if (send) {
-      if (dtmfDurationSamples > 0xffff) {
+      if (dtmf_duration_samples > 0xffff) {
         // RFC 4733 2.5.2.3 Long-Duration Events
-        SendTelephoneEventPacket(ended, dtmf_payload_type, _dtmfTimestamp,
+        SendTelephoneEventPacket(ended, dtmf_payload_type, dtmf_timestamp_,
                                  static_cast<uint16_t>(0xffff), false);
 
         // set new timestap for this segment
-        _dtmfTimestamp = captureTimeStamp;
-        dtmfDurationSamples -= 0xffff;
-        _dtmfLengthSamples -= 0xffff;
+        dtmf_timestamp_ = capture_timestamp;
+        dtmf_duration_samples -= 0xffff;
+        dtmf_length_samples_ -= 0xffff;
 
         return SendTelephoneEventPacket(
-            ended, dtmf_payload_type, _dtmfTimestamp,
-            static_cast<uint16_t>(dtmfDurationSamples), false);
+            ended, dtmf_payload_type, dtmf_timestamp_,
+            static_cast<uint16_t>(dtmf_duration_samples), false);
       } else {
-        if (SendTelephoneEventPacket(ended, dtmf_payload_type, _dtmfTimestamp,
-                                     static_cast<uint16_t>(dtmfDurationSamples),
-                                     !_dtmfEventFirstPacketSent) != 0) {
+        if (SendTelephoneEventPacket(ended, dtmf_payload_type, dtmf_timestamp_,
+                                     dtmf_duration_samples,
+                                     !dtmf_event_first_packet_sent_) != 0) {
           return -1;
         }
-        _dtmfEventFirstPacketSent = true;
+        dtmf_event_first_packet_sent_ = true;
         return 0;
       }
     }
     return 0;
   }
-  if (payloadSize == 0 || payloadData == NULL) {
-    if (frameType == kEmptyFrame) {
+  if (payload_size == 0 || payload_data == NULL) {
+    if (frame_type == kEmptyFrame) {
       // we don't send empty audio RTP packets
       // no error since we use it to drive DTMF when we use VAD
       return 0;
@@ -248,7 +248,7 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
     return -1;
   }
   uint8_t dataBuffer[IP_PACKET_SIZE];
-  bool markerBit = MarkerBit(frameType, payloadType);
+  bool markerBit = MarkerBit(frame_type, payload_type);
 
   int32_t rtpHeaderLength = 0;
   uint16_t timestampOffset = 0;
@@ -257,21 +257,21 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
       fragmentation->fragmentationVectorSize > 1) {
     // have we configured RED? use its payload type
     // we need to get the current timestamp to calc the diff
-    uint32_t oldTimeStamp = _rtpSender->Timestamp();
-    rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, red_payload_type,
-                                                 markerBit, captureTimeStamp,
-                                                 _clock->TimeInMilliseconds());
+    uint32_t oldTimeStamp = rtp_sender_->Timestamp();
+    rtpHeaderLength = rtp_sender_->BuildRtpHeader(
+        dataBuffer, red_payload_type, markerBit, capture_timestamp,
+        clock_->TimeInMilliseconds());
 
-    timestampOffset = uint16_t(_rtpSender->Timestamp() - oldTimeStamp);
+    timestampOffset = uint16_t(rtp_sender_->Timestamp() - oldTimeStamp);
   } else {
-    rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType,
-                                                 markerBit, captureTimeStamp,
-                                                 _clock->TimeInMilliseconds());
+    rtpHeaderLength = rtp_sender_->BuildRtpHeader(dataBuffer, payload_type,
+                                                 markerBit, capture_timestamp,
+                                                 clock_->TimeInMilliseconds());
   }
   if (rtpHeaderLength <= 0) {
     return -1;
   }
-  if (maxPayloadLength < (rtpHeaderLength + payloadSize)) {
+  if (max_payload_length < (rtpHeaderLength + payload_size)) {
     // Too large payload buffer.
     return -1;
   }
@@ -300,57 +300,57 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
       dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
       // copy the RED data
       memcpy(dataBuffer + rtpHeaderLength,
-             payloadData + fragmentation->fragmentationOffset[1],
+             payload_data + fragmentation->fragmentationOffset[1],
              fragmentation->fragmentationLength[1]);
 
       // copy the normal data
       memcpy(
           dataBuffer + rtpHeaderLength + fragmentation->fragmentationLength[1],
-          payloadData + fragmentation->fragmentationOffset[0],
+          payload_data + fragmentation->fragmentationOffset[0],
           fragmentation->fragmentationLength[0]);
 
-      payloadSize = fragmentation->fragmentationLength[0] +
+      payload_size = fragmentation->fragmentationLength[0] +
                     fragmentation->fragmentationLength[1];
     } else {
       // silence for too long send only new data
       dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
       memcpy(dataBuffer + rtpHeaderLength,
-             payloadData + fragmentation->fragmentationOffset[0],
+             payload_data + fragmentation->fragmentationOffset[0],
              fragmentation->fragmentationLength[0]);
 
-      payloadSize = fragmentation->fragmentationLength[0];
+      payload_size = fragmentation->fragmentationLength[0];
     }
   } else {
     if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
       // use the fragment info if we have one
       dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
       memcpy(dataBuffer + rtpHeaderLength,
-             payloadData + fragmentation->fragmentationOffset[0],
+             payload_data + fragmentation->fragmentationOffset[0],
              fragmentation->fragmentationLength[0]);
 
-      payloadSize = fragmentation->fragmentationLength[0];
+      payload_size = fragmentation->fragmentationLength[0];
     } else {
-      memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize);
+      memcpy(dataBuffer + rtpHeaderLength, payload_data, payload_size);
     }
   }
 
   {
-    rtc::CritScope cs(&_sendAudioCritsect);
-    _lastPayloadType = payloadType;
+    rtc::CritScope cs(&send_audio_critsect_);
+    last_payload_type_ = payload_type;
   }
   // Update audio level extension, if included.
-  size_t packetSize = payloadSize + rtpHeaderLength;
+  size_t packetSize = payload_size + rtpHeaderLength;
   RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
   RTPHeader rtp_header;
   rtp_parser.Parse(&rtp_header);
-  _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
-                               (frameType == kAudioFrameSpeech),
+  rtp_sender_->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
+                               (frame_type == kAudioFrameSpeech),
                                audio_level_dbov);
-  TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
-                         _rtpSender->Timestamp(), "seqnum",
-                         _rtpSender->SequenceNumber());
-  int32_t send_result = _rtpSender->SendToNetwork(
-      dataBuffer, payloadSize, rtpHeaderLength,
+  TRACE_EVENT_ASYNC_END2("webrtc", "Audio", capture_timestamp, "timestamp",
+                         rtp_sender_->Timestamp(), "seqnum",
+                         rtp_sender_->SequenceNumber());
+  int32_t send_result = rtp_sender_->SendToNetwork(
+      dataBuffer, payload_size, rtpHeaderLength,
       rtc::TimeMillis(), kAllowRetransmission,
       RtpPacketSender::kHighPriority);
   if (first_packet_sent_()) {
@@ -360,33 +360,33 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
-int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dBov) {
-  if (level_dBov > 127) {
+int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dbov) {
+  if (level_dbov > 127) {
     return -1;
   }
-  rtc::CritScope cs(&_sendAudioCritsect);
-  _audioLevel_dBov = level_dBov;
+  rtc::CritScope cs(&send_audio_critsect_);
+  audio_level_dbov_ = level_dbov;
   return 0;
 }
 
 // Set payload type for Redundant Audio Data RFC 2198
-int32_t RTPSenderAudio::SetRED(int8_t payloadType) {
-  if (payloadType < -1) {
+int32_t RTPSenderAudio::SetRED(int8_t payload_type) {
+  if (payload_type < -1) {
     return -1;
   }
-  rtc::CritScope cs(&_sendAudioCritsect);
-  _REDPayloadType = payloadType;
+  rtc::CritScope cs(&send_audio_critsect_);
+  red_payload_type_ = payload_type;
   return 0;
 }
 
 // Get payload type for Redundant Audio Data RFC 2198
-int32_t RTPSenderAudio::RED(int8_t* payloadType) const {
-  rtc::CritScope cs(&_sendAudioCritsect);
-  if (_REDPayloadType == -1) {
+int32_t RTPSenderAudio::RED(int8_t* payload_type) const {
+  rtc::CritScope cs(&send_audio_critsect_);
+  if (red_payload_type_ == -1) {
     // not configured
     return -1;
   }
-  *payloadType = _REDPayloadType;
+  *payload_type = red_payload_type_;
   return 0;
 }
 
@@ -395,8 +395,8 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
                                            uint16_t time_ms,
                                            uint8_t level) {
   {
-    rtc::CritScope lock(&_sendAudioCritsect);
-    if (_dtmfPayloadType < 0) {
+    rtc::CritScope lock(&send_audio_critsect_);
+    if (dtmf_payload_type_ < 0) {
       // TelephoneEvent payloadtype not configured
       return -1;
     }
@@ -406,7 +406,7 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
 
 int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                                                  int8_t dtmf_payload_type,
-                                                 uint32_t dtmfTimeStamp,
+                                                 uint32_t dtmf_timestamp,
                                                  uint16_t duration,
                                                  bool markerBit) {
   uint8_t dtmfbuffer[IP_PACKET_SIZE];
@@ -419,8 +419,8 @@ int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
   }
   do {
     // Send DTMF data
-    _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit,
-                               dtmfTimeStamp, _clock->TimeInMilliseconds());
+    rtp_sender_->BuildRtpHeader(dtmfbuffer, dtmf_payload_type, markerBit,
+                               dtmf_timestamp, clock_->TimeInMilliseconds());
 
     // reset CSRC and X bit
     dtmfbuffer[0] &= 0xe0;
@@ -436,20 +436,20 @@ int32_t RTPSenderAudio::SendTelephoneEventPacket(bool ended,
     */
     // R bit always cleared
     uint8_t R = 0x00;
-    uint8_t volume = _dtmfLevel;
+    uint8_t volume = dtmf_level_;
 
     // First packet un-ended
     uint8_t E = ended ? 0x80 : 0x00;
 
     // First byte is Event number, equals key number
-    dtmfbuffer[12] = _dtmfKey;
+    dtmfbuffer[12] = dtmf_key_;
     dtmfbuffer[13] = E | R | volume;
     ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
 
-    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
-                         "Audio::SendTelephoneEvent", "timestamp",
-                         dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
-    retVal = _rtpSender->SendToNetwork(
+    TRACE_EVENT_INSTANT2(
+        TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
+        "timestamp", dtmf_timestamp, "seqnum", rtp_sender_->SequenceNumber());
+    retVal = rtp_sender_->SendToNetwork(
         dtmfbuffer, 4, 12, rtc::TimeMillis(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;