mirror of
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70206d6608
Instead of modifying the API, we'll add a new function to return
the true value, and have a shim that returns what other code expects.
> This reverts commit 4c34f435db.
>
> Reason for revert: Broke internal projects. Type mismatch.
>
> Original change's description:
> > Make RTCP cumulative_lost be a signed value
> >
> > This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> > See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> >
> > Noticed on discuss-webrtc mailing list.
> >
> > BUG=webrtc:8626
> >
> > Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> > Reviewed-on: https://webrtc-review.googlesource.com/30901
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21142}
>
> TBR=stefan@webrtc.org,hta@webrtc.org
>
> Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8626
> Reviewed-on: https://webrtc-review.googlesource.com/31040
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21144}
Change-Id: I95c8c248f4f85c4d1aa2a47424d8c4d954d4ae7a
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21154}
71 lines
2.5 KiB
C++
71 lines
2.5 KiB
C++
/*
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* Copyright (c) 2015 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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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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#include "rtc_base/basictypes.h"
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namespace webrtc {
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namespace rtcp {
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// A ReportBlock represents the Sender Report packet from
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// RFC 3550 section 6.4.1.
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class ReportBlock {
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public:
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static const size_t kLength = 24;
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ReportBlock();
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~ReportBlock() {}
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bool Parse(const uint8_t* buffer, size_t length);
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// Fills buffer with the ReportBlock.
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// Consumes ReportBlock::kLength bytes.
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void Create(uint8_t* buffer) const;
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void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
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void SetFractionLost(uint8_t fraction_lost) {
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fraction_lost_ = fraction_lost;
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}
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bool SetCumulativeLost(int32_t cumulative_lost);
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void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) {
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extended_high_seq_num_ = ext_highest_seq_num;
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}
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void SetJitter(uint32_t jitter) { jitter_ = jitter; }
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void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
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void SetDelayLastSr(uint32_t delay_last_sr) {
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delay_since_last_sr_ = delay_last_sr;
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}
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uint32_t source_ssrc() const { return source_ssrc_; }
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uint8_t fraction_lost() const { return fraction_lost_; }
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int32_t cumulative_lost_signed() const { return cumulative_lost_; }
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// Deprecated - returns max(0, cumulative_lost_), not negative values.
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uint32_t cumulative_lost() const;
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uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
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uint32_t jitter() const { return jitter_; }
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uint32_t last_sr() const { return last_sr_; }
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uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
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private:
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uint32_t source_ssrc_; // 32 bits
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uint8_t fraction_lost_; // 8 bits representing a fixed point value 0..1
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int32_t cumulative_lost_; // Signed 24-bit value
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uint32_t extended_high_seq_num_; // 32 bits
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uint32_t jitter_; // 32 bits
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uint32_t last_sr_; // 32 bits
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uint32_t delay_since_last_sr_; // 32 bits, units of 1/65536 seconds
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};
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} // namespace rtcp
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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