[rtp_rtcp] Lint errors cleaned from rtp_utility

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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.
  */
 
@@ skipping 561 matching lines @@
     size_t length = bytes_left;
     int64_t capture_time_ms;
     if (!packet_history_.GetBestFittingPacket(buffer, &length,
                                               &capture_time_ms)) {
       break;
     }
     if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false))
       break;
     RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
     bytes_left -= static_cast<int>(length - rtp_header.headerLength);
   }
   return bytes_to_send - bytes_left;
 }
 
 void RTPSender::BuildPaddingPacket(uint8_t* packet,
                                    size_t header_length,
                                    size_t padding_length) {
   packet[0] |= 0x20;  // Set padding bit.
-  int32_t *data =
-      reinterpret_cast<int32_t *>(&(packet[header_length]));
+  int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
 
   // Fill data buffer with random data.
   for (size_t j = 0; j < (padding_length >> 2); ++j) {
     data[j] = rand();  // NOLINT
   }
   // Set number of padding bytes in the last byte of the packet.
   packet[header_length + padding_length - 1] =
       static_cast<uint8_t>(padding_length);
 }
 
@@ skipping 60 matching lines @@
     uint8_t padding_packet[IP_PACKET_SIZE];
     size_t header_length =
         CreateRtpHeader(padding_packet, payload_type, ssrc, false, timestamp,
                         sequence_number, std::vector<uint32_t>());
     BuildPaddingPacket(padding_packet, header_length, padding_bytes_in_packet);
     size_t length = padding_bytes_in_packet + header_length;
     int64_t now_ms = clock_->TimeInMilliseconds();
 
     RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
     RTPHeader rtp_header;
-    rtp_parser.Parse(rtp_header);
+    rtp_parser.Parse(&rtp_header);
 
     if (capture_time_ms > 0) {
       UpdateTransmissionTimeOffset(
           padding_packet, length, rtp_header, now_ms - capture_time_ms);
     }
 
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
     if (using_transport_seq) {
@@ skipping 31 matching lines @@
   if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
                                                data_buffer, &length,
                                                &capture_time_ms)) {
     // Packet not found.
     return 0;
   }
 
   if (paced_sender_) {
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
-    if (!rtp_parser.Parse(header)) {
+    if (!rtp_parser.Parse(&header)) {
       assert(false);
       return -1;
     }
     // Convert from TickTime to Clock since capture_time_ms is based on
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(
         RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
         corrected_capture_tims_ms, length - header.headerLength, true);
 
@@ skipping 165 matching lines @@
                               capture_time_ms,
                               retransmission && (rtx & kRtxRetransmitted) > 0,
                               retransmission);
 }
 
 bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                      size_t length,
                                      int64_t capture_time_ms,
                                      bool send_over_rtx,
                                      bool is_retransmit) {
-  uint8_t *buffer_to_send_ptr = buffer;
+  uint8_t* buffer_to_send_ptr = buffer;
 
   RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
   if (!is_retransmit && rtp_header.markerBit) {
     TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
                            capture_time_ms);
   }
 
   TRACE_EVENT_INSTANT2(
       TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PrepareAndSendPacket",
       "timestamp", rtp_header.timestamp, "seqnum", rtp_header.sequenceNumber);
 
   uint8_t data_buffer_rtx[IP_PACKET_SIZE];
@@ skipping 98 matching lines @@
 // TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
 int32_t RTPSender::SendToNetwork(uint8_t* buffer,
                                  size_t payload_length,
                                  size_t rtp_header_length,
                                  int64_t capture_time_ms,
                                  StorageType storage,
                                  RtpPacketSender::Priority priority) {
   RtpUtility::RtpHeaderParser rtp_parser(buffer,
                                          payload_length + rtp_header_length);
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   int64_t now_ms = clock_->TimeInMilliseconds();
 
   // |capture_time_ms| <= 0 is considered invalid.
   // TODO(holmer): This should be changed all over Video Engine so that negative
   // time is consider invalid, while 0 is considered a valid time.
   if (capture_time_ms > 0) {
     UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
                                  rtp_header, now_ms - capture_time_ms);
   }
@@ skipping 122 matching lines @@
   header[1] = static_cast<uint8_t>(payload_type);
   if (marker_bit) {
     header[1] |= kRtpMarkerBitMask;  // Marker bit is set.
   }
   ByteWriter<uint16_t>::WriteBigEndian(header + 2, sequence_number);
   ByteWriter<uint32_t>::WriteBigEndian(header + 4, timestamp);
   ByteWriter<uint32_t>::WriteBigEndian(header + 8, ssrc);
   int32_t rtp_header_length = kRtpHeaderLength;
 
   if (csrcs.size() > 0) {
-    uint8_t *ptr = &header[rtp_header_length];
+    uint8_t* ptr = &header[rtp_header_length];
     for (size_t i = 0; i < csrcs.size(); ++i) {
       ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
       ptr += 4;
     }
     header[0] = (header[0] & 0xf0) | csrcs.size();
 
     // Update length of header.
     rtp_header_length += sizeof(uint32_t) * csrcs.size();
   }
 
@@ skipping 631 matching lines @@
 
 void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
                                uint8_t* buffer_rtx) {
   CriticalSectionScoped cs(send_critsect_.get());
   uint8_t* data_buffer_rtx = buffer_rtx;
   // Add RTX header.
   RtpUtility::RtpHeaderParser rtp_parser(
       reinterpret_cast<const uint8_t*>(buffer), *length);
 
   RTPHeader rtp_header;
-  rtp_parser.Parse(rtp_header);
+  rtp_parser.Parse(&rtp_header);
 
   // Add original RTP header.
   memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
 
   // Replace payload type, if a specific type is set for RTX.
   if (rtx_payload_type_ != -1) {
     data_buffer_rtx[1] = static_cast<uint8_t>(rtx_payload_type_);
     if (rtp_header.markerBit)
       data_buffer_rtx[1] |= kRtpMarkerBitMask;
   }
 
   // Replace sequence number.
-  uint8_t *ptr = data_buffer_rtx + 2;
+  uint8_t* ptr = data_buffer_rtx + 2;
   ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
 
   // Replace SSRC.
   ptr += 6;
   ByteWriter<uint32_t>::WriteBigEndian(ptr, ssrc_rtx_);
 
   // Add OSN (original sequence number).
   ptr = data_buffer_rtx + rtp_header.headerLength;
   ByteWriter<uint16_t>::WriteBigEndian(ptr, rtp_header.sequenceNumber);
   ptr += 2;
@@ skipping 53 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc