blob: 67e8967922116dd3b10e45d6e79e7511216feb8c [file] [log] [blame]
/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
#define MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
#include <stdint.h>
#include <map>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/rtp_receiver_interface.h" // For RtpSource
#include "rtc_base/time_utils.h" // For kNumMillisecsPerSec
namespace webrtc {
class ContributingSources {
public:
// Set by the spec, see
// https://www.w3.org/TR/webrtc/#dom-rtcrtpreceiver-getcontributingsources
static constexpr int64_t kHistoryMs = 10 * rtc::kNumMillisecsPerSec;
ContributingSources();
~ContributingSources();
void Update(int64_t now_ms,
rtc::ArrayView<const uint32_t> csrcs,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp);
// Returns contributing sources seen the last 10 s.
std::vector<RtpSource> GetSources(int64_t now_ms) const;
private:
struct Entry {
Entry();
Entry(int64_t timestamp_ms,
absl::optional<uint8_t> audio_level,
uint32_t rtp_timestamp);
int64_t last_seen_ms;
absl::optional<uint8_t> audio_level;
uint32_t rtp_timestamp;
};
void DeleteOldEntries(int64_t now_ms);
// Indexed by csrc.
std::map<uint32_t, Entry> active_csrcs_;
absl::optional<int64_t> next_pruning_ms_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_