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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
56 lines
2 KiB
C++
56 lines
2 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_RTCP_H_
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#define MODULES_AUDIO_CODING_NETEQ_RTCP_H_
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#include <stdint.h>
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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struct RtcpStatistics;
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struct RTPHeader;
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class Rtcp {
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public:
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Rtcp() { Init(0); }
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~Rtcp() {}
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// Resets the RTCP statistics, and sets the first received sequence number.
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void Init(uint16_t start_sequence_number);
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// Updates the RTCP statistics with a new received packet.
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void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
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// Returns the current RTCP statistics. If |no_reset| is true, the statistics
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// are not reset, otherwise they are.
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void GetStatistics(bool no_reset, RtcpStatistics* stats);
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private:
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uint16_t cycles_; // The number of wrap-arounds for the sequence number.
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uint16_t max_seq_no_; // The maximum sequence number received. Starts over
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// from 0 after wrap-around.
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uint16_t base_seq_no_; // The sequence number of the first received packet.
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uint32_t received_packets_; // The number of packets that have been received.
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uint32_t received_packets_prior_; // Number of packets received when last
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// report was generated.
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uint32_t expected_prior_; // Expected number of packets, at the time of the
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// last report.
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int64_t jitter_; // Current jitter value in Q4.
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int32_t transit_; // Clock difference for previous packet.
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RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_RTCP_H_
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