[fix] fix qos module

src/qos/report_block.h

@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 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 MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+namespace rtcp {
+
+// A ReportBlock represents the Sender Report packet from
+// RFC 3550 section 6.4.1.
+class ReportBlock {
+ public:
+  static constexpr size_t kLength = 24;
+
+  ReportBlock();
+  ~ReportBlock() {}
+
+  bool Parse(const uint8_t* buffer, size_t length);
+
+  // Fills buffer with the ReportBlock.
+  // Consumes ReportBlock::kLength bytes.
+  void Create(uint8_t* buffer) const;
+
+  void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
+  void SetFractionLost(uint8_t fraction_lost) {
+    fraction_lost_ = fraction_lost;
+  }
+  bool SetCumulativeLost(int32_t cumulative_lost);
+  void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) {
+    extended_high_seq_num_ = ext_highest_seq_num;
+  }
+  void SetJitter(uint32_t jitter) { jitter_ = jitter; }
+  void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
+  void SetDelayLastSr(uint32_t delay_last_sr) {
+    delay_since_last_sr_ = delay_last_sr;
+  }
+
+  uint32_t source_ssrc() const { return source_ssrc_; }
+  uint8_t fraction_lost() const { return fraction_lost_; }
+  int32_t cumulative_lost() const { return cumulative_lost_; }
+  uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
+  uint32_t jitter() const { return jitter_; }
+  uint32_t last_sr() const { return last_sr_; }
+  uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
+
+ private:
+  uint32_t source_ssrc_;     // 32 bits
+  uint8_t fraction_lost_;    // 8 bits representing a fixed point value 0..1
+  int32_t cumulative_lost_;  // Signed 24-bit value
+  uint32_t extended_high_seq_num_;  // 32 bits
+  uint32_t jitter_;                 // 32 bits
+  uint32_t last_sr_;                // 32 bits
+  uint32_t delay_since_last_sr_;    // 32 bits, units of 1/65536 seconds
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_