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https://github.com/mollyim/webrtc.git
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2e00abc98e
Reason for reland: Downstream project fixed. Original change's description: > [cleanup] Remove useless includes. > > Manual cleanup guided by include-what-you-use diagnostic. > > Bug: webrtc:8311 > Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68 > Reviewed-on: https://webrtc-review.googlesource.com/c/103320 > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25013} Bug: webrtc:8311 Change-Id: Id6ec4aeb798886a90ace640a190eaf16497ba31b Reviewed-on: https://webrtc-review.googlesource.com/c/104120 Commit-Queue: Yves Gerey <yvesg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25034}
52 lines
1.5 KiB
C++
52 lines
1.5 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" // For RtpSource
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#include "rtc_base/timeutils.h" // For kNumMillisecsPerSec
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namespace webrtc {
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class ContributingSources {
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public:
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// Set by the spec, see
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// https://www.w3.org/TR/webrtc/#dom-rtcrtpreceiver-getcontributingsources
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static constexpr int64_t kHistoryMs = 10 * rtc::kNumMillisecsPerSec;
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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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