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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
85 lines
2.9 KiB
C++
85 lines
2.9 KiB
C++
/*
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* Copyright (c) 2019 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_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
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#include <cstddef>
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#include <cstdint>
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#include <deque>
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#include "absl/types/optional.h"
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namespace webrtc {
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// Records the association of RTP sequence numbers to timestamps and to whether
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// the packet was first and/or last in the frame.
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//
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// 1. Limits number of entries. Whenever |max_entries| is about to be exceeded,
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// the size is reduced by approximately 25%.
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// 2. RTP sequence numbers wrap around relatively infrequently.
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// This class therefore only remembers at most the last 2^15 RTP packets,
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// so that the newest packet's sequence number is still AheadOf the oldest
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// packet's sequence number.
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// 3. Media frames are sometimes split into several RTP packets.
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// In such a case, Insert() is expected to be called once for each packet.
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// The timestamp is not expected to change between those calls.
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class RtpSequenceNumberMap final {
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public:
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struct Info final {
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Info(uint32_t timestamp, bool is_first, bool is_last)
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: timestamp(timestamp), is_first(is_first), is_last(is_last) {}
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friend bool operator==(const Info& lhs, const Info& rhs) {
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return lhs.timestamp == rhs.timestamp && lhs.is_first == rhs.is_first &&
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lhs.is_last == rhs.is_last;
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}
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uint32_t timestamp;
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bool is_first;
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bool is_last;
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};
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explicit RtpSequenceNumberMap(size_t max_entries);
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RtpSequenceNumberMap(const RtpSequenceNumberMap& other) = delete;
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RtpSequenceNumberMap& operator=(const RtpSequenceNumberMap& other) = delete;
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~RtpSequenceNumberMap();
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void InsertPacket(uint16_t sequence_number, Info info);
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void InsertFrame(uint16_t first_sequence_number,
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size_t packet_count,
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uint32_t timestamp);
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absl::optional<Info> Get(uint16_t sequence_number) const;
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size_t AssociationCountForTesting() const;
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private:
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struct Association {
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explicit Association(uint16_t sequence_number)
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: Association(sequence_number, Info(0, false, false)) {}
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Association(uint16_t sequence_number, Info info)
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: sequence_number(sequence_number), info(info) {}
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uint16_t sequence_number;
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Info info;
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};
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const size_t max_entries_;
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// The non-transitivity of AheadOf() would be problematic with a map,
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// so we use a deque instead.
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std::deque<Association> associations_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_
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