mirror of
https://github.com/mollyim/webrtc.git
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96a0f61917
This reverts commit be8b5348c7.
Reason for revert: Breaks downstream project
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}
TBR=phoglund@google.com,phoglund@webrtc.org,yvesg@webrtc.org
Change-Id: I7a6e1cdfef685173b76f234ad598083043dcd9a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8311
Reviewed-on: https://webrtc-review.googlesource.com/c/104022
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25015}
51 lines
1.5 KiB
C++
51 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"
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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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