Cast: Refactoring RtpReceiver to Clang format

media/cast/rtp_receiver/rtp_parser/rtp_parser.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
 
 #include "base/logging.h"
 #include "media/cast/cast_defines.h"
 #include "media/cast/rtp_receiver/rtp_receiver.h"
 #include "net/base/big_endian.h"
 
 namespace media {
 namespace cast {
 
 static const size_t kRtpCommonHeaderLength = 12;
 static const size_t kRtpCastHeaderLength = 7;
 static const uint8 kCastKeyFrameBitMask = 0x80;
 static const uint8 kCastReferenceFrameIdBitMask = 0x40;
 
 RtpParser::RtpParser(RtpData* incoming_payload_callback,
                      const RtpParserConfig parser_config)
     : data_callback_(incoming_payload_callback),
       parser_config_(parser_config) {}
 
 RtpParser::~RtpParser() {}
 
-bool RtpParser::ParsePacket(const uint8* packet, size_t length,
+bool RtpParser::ParsePacket(const uint8* packet,
+                            size_t length,
                             RtpCastHeader* rtp_header) {
-  if (length == 0) return false;
+  if (length == 0)
+    return false;
   // Get RTP general header.
-  if (!ParseCommon(packet, length, rtp_header)) return false;
+  if (!ParseCommon(packet, length, rtp_header))
+    return false;
   if (rtp_header->webrtc.header.payloadType == parser_config_.payload_type &&
-    rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
+      rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
     return ParseCast(packet + kRtpCommonHeaderLength,
-        length - kRtpCommonHeaderLength, rtp_header);
+                     length - kRtpCommonHeaderLength,
+                     rtp_header);
   }
   // Not a valid payload type / ssrc combination.
   return false;
 }
 
 bool RtpParser::ParseCommon(const uint8* packet,
                             size_t length,
                             RtpCastHeader* rtp_header) {
-  if (length < kRtpCommonHeaderLength) return false;
+  if (length < kRtpCommonHeaderLength)
+    return false;
   uint8 version = packet[0] >> 6;
-  if (version != 2) return false;
+  if (version != 2)
+    return false;
   uint8 cc = packet[0] & 0x0f;
   bool marker = ((packet[1] & 0x80) != 0);
   int payload_type = packet[1] & 0x7f;
 
   uint16 sequence_number;
   uint32 rtp_timestamp, ssrc;
   net::BigEndianReader big_endian_reader(packet + 2, 10);
   big_endian_reader.ReadU16(&sequence_number);
   big_endian_reader.ReadU32(&rtp_timestamp);
   big_endian_reader.ReadU32(&ssrc);
 
-  if (ssrc != parser_config_.ssrc) return false;
+  if (ssrc != parser_config_.ssrc)
+    return false;
 
-  rtp_header->webrtc.header.markerBit      = marker;
-  rtp_header->webrtc.header.payloadType    = payload_type;
+  rtp_header->webrtc.header.markerBit = marker;
+  rtp_header->webrtc.header.payloadType = payload_type;
   rtp_header->webrtc.header.sequenceNumber = sequence_number;
-  rtp_header->webrtc.header.timestamp      = rtp_timestamp;
-  rtp_header->webrtc.header.ssrc           = ssrc;
-  rtp_header->webrtc.header.numCSRCs       = cc;
+  rtp_header->webrtc.header.timestamp = rtp_timestamp;
+  rtp_header->webrtc.header.ssrc = ssrc;
+  rtp_header->webrtc.header.numCSRCs = cc;
 
   uint8 csrc_octs = cc * 4;
   rtp_header->webrtc.type.Audio.numEnergy = rtp_header->webrtc.header.numCSRCs;
   rtp_header->webrtc.header.headerLength = kRtpCommonHeaderLength + csrc_octs;
   rtp_header->webrtc.type.Audio.isCNG = false;
   rtp_header->webrtc.type.Audio.channel = parser_config_.audio_channels;
   // TODO(pwestin): look at x bit and skip data.
   return true;
 }
 
 bool RtpParser::ParseCast(const uint8* packet,
                           size_t length,
                           RtpCastHeader* rtp_header) {
-  if (length < kRtpCastHeaderLength) return false;
+  if (length < kRtpCastHeaderLength)
+    return false;
 
   // Extract header.
   const uint8* data_ptr = packet;
   size_t data_length = length;
   rtp_header->is_key_frame = (data_ptr[0] & kCastKeyFrameBitMask);
   rtp_header->is_reference = (data_ptr[0] & kCastReferenceFrameIdBitMask);
   rtp_header->frame_id = frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[1]);
 
   net::BigEndianReader big_endian_reader(data_ptr + 2, 4);
   big_endian_reader.ReadU16(&rtp_header->packet_id);
   big_endian_reader.ReadU16(&rtp_header->max_packet_id);
 
   if (rtp_header->is_reference) {
     rtp_header->reference_frame_id =
         reference_frame_id_wrap_helper_.MapTo32bitsFrameId(data_ptr[6]);
     data_ptr += kRtpCastHeaderLength;
     data_length -= kRtpCastHeaderLength;
   } else {
     data_ptr += kRtpCastHeaderLength - 1;
     data_length -= kRtpCastHeaderLength - 1;
   }
 
-  if (rtp_header->max_packet_id < rtp_header->packet_id) return false;
+  if (rtp_header->max_packet_id < rtp_header->packet_id)
+    return false;
 
   data_callback_->OnReceivedPayloadData(data_ptr, data_length, rtp_header);
   return true;
 }
 
 }  // namespace cast
 }  // namespace media