[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function

webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc

Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
new file mode 100644
index 0000000000000000000000000000000000000000..ab1863ed46bc973fa0037a75fdb142bd1e5139ac
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+//    Sender report (SR) (RFC 3550).
+//     0                   1                   2                   3
+//     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//    |V=2|P|    RC   |   PT=SR=200   |             length            |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  0 |                         SSRC of sender                        |
+//    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+//  4 |              NTP timestamp, most significant word             |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//  8 |             NTP timestamp, least significant word             |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 12 |                         RTP timestamp                         |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 16 |                     sender's packet count                     |
+//    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// 20 |                      sender's octet count                     |
+// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
+
+SenderReport::SenderReport()
+    : sender_ssrc_(0),
+      rtp_timestamp_(0),
+      sender_packet_count_(0),
+      sender_octet_count_(0) {}
+
+bool SenderReport::Parse(const RtcpCommonHeader& header,
+                         const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t report_block_count = header.count_or_format;
+  if (header.payload_size_bytes <
+      kSenderBaseLength + report_block_count * ReportBlock::kLength) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the data.";
+    return false;
+  }
+  // Read SenderReport header.
+  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
+  uint32_t secs = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
+  uint32_t frac = ByteReader<uint32_t>::ReadBigEndian(&payload[8]);
+  ntp_.Set(secs, frac);
+  rtp_timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&payload[12]);
+  sender_packet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[16]);
+  sender_octet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[20]);
+  report_blocks_.resize(report_block_count);
+  const uint8_t* next_block = payload + kSenderBaseLength;
+  for (ReportBlock& block : report_blocks_) {
+    bool block_parsed = block.Parse(next_block, ReportBlock::kLength);
+    RTC_DCHECK(block_parsed);
+    next_block += ReportBlock::kLength;
+  }
+  // Double check we didn't read beyond provided buffer.
+  RTC_DCHECK_LE(next_block, payload + header.payload_size_bytes);
+  return true;
+}
+
+bool SenderReport::Create(uint8_t* packet,
+                          size_t* index,
+                          size_t max_length,
+                          RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+
+  CreateHeader(report_blocks_.size(), kPacketType, HeaderLength(), packet,
+               index);
+  // Write SenderReport header.
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 0], sender_ssrc_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 4], ntp_.seconds());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 8], ntp_.fractions());
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 12], rtp_timestamp_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 16],
+                                       sender_packet_count_);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[*index + 20],
+                                       sender_octet_count_);
+  *index += kSenderBaseLength;
+  // Write report blocks.
+  for (const ReportBlock& block : report_blocks_) {
+    block.Create(packet + *index);
+    *index += ReportBlock::kLength;
+  }
+  // Ensure bytes written match expected.
+  RTC_DCHECK_EQ(*index, index_end);
+  return true;
+}
+
+bool SenderReport::WithReportBlock(const ReportBlock& block) {
+  if (report_blocks_.size() >= kMaxNumberOfReportBlocks) {
+    LOG(LS_WARNING) << "Max report blocks reached.";
+    return false;
+  }
+  report_blocks_.push_back(block);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc