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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}
81 lines
2.7 KiB
C++
81 lines
2.7 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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#include "modules/rtp_rtcp/source/contributing_sources.h"
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namespace webrtc {
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namespace {
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// Allow some stale records to accumulate before cleaning.
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constexpr int64_t kPruningIntervalMs = 15 * rtc::kNumMillisecsPerSec;
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} // namespace
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constexpr int64_t ContributingSources::kHistoryMs;
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ContributingSources::ContributingSources() = default;
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ContributingSources::~ContributingSources() = default;
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void ContributingSources::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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Entry entry = {now_ms, audio_level, rtp_timestamp};
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for (uint32_t csrc : csrcs) {
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active_csrcs_[csrc] = entry;
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}
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if (!next_pruning_ms_) {
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next_pruning_ms_ = now_ms + kPruningIntervalMs;
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} else if (now_ms > next_pruning_ms_) {
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// To prevent unlimited growth, prune it every 15 seconds.
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DeleteOldEntries(now_ms);
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}
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}
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// Return contributing sources seen the last 10 s.
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// TODO(nisse): It would be more efficient to delete any stale entries while
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// iterating over the mapping, but then we'd have to make the method
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// non-const.
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std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
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std::vector<RtpSource> sources;
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for (auto& record : active_csrcs_) {
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if (record.second.last_seen_ms >= now_ms - kHistoryMs) {
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sources.emplace_back(record.second.last_seen_ms, record.first,
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RtpSourceType::CSRC, record.second.audio_level,
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record.second.rtp_timestamp);
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}
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}
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return sources;
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}
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// Delete stale entries.
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void ContributingSources::DeleteOldEntries(int64_t now_ms) {
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for (auto it = active_csrcs_.begin(); it != active_csrcs_.end();) {
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if (it->second.last_seen_ms >= now_ms - kHistoryMs) {
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// Still relevant.
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++it;
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} else {
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it = active_csrcs_.erase(it);
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}
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}
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next_pruning_ms_ = now_ms + kPruningIntervalMs;
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}
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ContributingSources::Entry::Entry() = default;
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ContributingSources::Entry::Entry(int64_t timestamp_ms,
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absl::optional<uint8_t> audio_level_arg,
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uint32_t rtp_timestamp)
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: last_seen_ms(timestamp_ms),
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audio_level(audio_level_arg),
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rtp_timestamp(rtp_timestamp) {}
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} // namespace webrtc
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