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b5d918324c
RTP timestamp was recently added to contributing sources in the WebRTC specification. This CL implements that change in WebRTC. Bug: webrtc:10650 Change-Id: Ic0ccfbea7049a5b66063fa6cf60d01d5bd713132 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137515 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28020}
64 lines
1.8 KiB
C++
64 lines
1.8 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/rtp_receiver_interface.h" // For RtpSource
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#include "rtc_base/time_utils.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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void Update(int64_t now_ms,
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rtc::ArrayView<const uint32_t> csrcs,
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absl::optional<uint8_t> audio_level,
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uint32_t rtp_timestamp);
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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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struct Entry {
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Entry();
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Entry(int64_t timestamp_ms,
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absl::optional<uint8_t> audio_level,
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uint32_t rtp_timestamp);
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int64_t last_seen_ms;
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absl::optional<uint8_t> audio_level;
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uint32_t rtp_timestamp;
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};
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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, Entry> active_csrcs_;
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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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