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/*
* 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 const 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_