| /* |
| * 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); |
| |
| // 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); |
| |
| int64_t last_seen_ms; |
| absl::optional<uint8_t> audio_level; |
| }; |
| |
| 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_ |