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af17595879
Bug: webrtc:7135 Change-Id: I7ce9afe575241542e4e3f7e2e8459ee3257eec76 Reviewed-on: https://webrtc-review.googlesource.com/93466 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24271}
47 lines
1.3 KiB
C++
47 lines
1.3 KiB
C++
/*
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* Copyright (c) 2018 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_CONTRIBUTING_SOURCES_H_
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#define MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
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#include <stdint.h>
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#include <map>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/array_view.h"
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#include "api/rtpreceiverinterface.h"
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namespace webrtc {
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class ContributingSources {
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public:
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ContributingSources();
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~ContributingSources();
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// TODO(bugs.webrtc.org/3333): Needs to be extended with audio-level, to
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// support RFC6465.
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void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs);
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// Returns contributing sources seen the last 10 s.
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std::vector<RtpSource> GetSources(int64_t now_ms) const;
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private:
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void DeleteOldEntries(int64_t now_ms);
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// Indexed by csrc.
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std::map<uint32_t, int64_t> last_seen_ms_;
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absl::optional<int64_t> next_pruning_ms_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
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