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- /*
- * Copyright (c) 2019 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_SOURCE_TRACKER_H_
- #define MODULES_RTP_RTCP_SOURCE_SOURCE_TRACKER_H_
- #include <cstdint>
- #include <list>
- #include <unordered_map>
- #include <utility>
- #include <vector>
- #include "absl/types/optional.h"
- #include "api/rtp_packet_infos.h"
- #include "api/transport/rtp/rtp_source.h"
- #include "rtc_base/synchronization/mutex.h"
- #include "rtc_base/time_utils.h"
- #include "system_wrappers/include/clock.h"
- namespace webrtc {
- //
- // Tracker for `RTCRtpContributingSource` and `RTCRtpSynchronizationSource`:
- // - https://w3c.github.io/webrtc-pc/#dom-rtcrtpcontributingsource
- // - https://w3c.github.io/webrtc-pc/#dom-rtcrtpsynchronizationsource
- //
- class SourceTracker {
- public:
- // Amount of time before the entry associated with an update is removed. See:
- // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
- static constexpr int64_t kTimeoutMs = 10000; // 10 seconds
- explicit SourceTracker(Clock* clock);
- SourceTracker(const SourceTracker& other) = delete;
- SourceTracker(SourceTracker&& other) = delete;
- SourceTracker& operator=(const SourceTracker& other) = delete;
- SourceTracker& operator=(SourceTracker&& other) = delete;
- // Updates the source entries when a frame is delivered to the
- // RTCRtpReceiver's MediaStreamTrack.
- void OnFrameDelivered(const RtpPacketInfos& packet_infos);
- // Returns an |RtpSource| for each unique SSRC and CSRC identifier updated in
- // the last |kTimeoutMs| milliseconds. Entries appear in reverse chronological
- // order (i.e. with the most recently updated entries appearing first).
- std::vector<RtpSource> GetSources() const;
- private:
- struct SourceKey {
- SourceKey(RtpSourceType source_type, uint32_t source)
- : source_type(source_type), source(source) {}
- // Type of |source|.
- RtpSourceType source_type;
- // CSRC or SSRC identifier of the contributing or synchronization source.
- uint32_t source;
- };
- struct SourceKeyComparator {
- bool operator()(const SourceKey& lhs, const SourceKey& rhs) const {
- return (lhs.source_type == rhs.source_type) && (lhs.source == rhs.source);
- }
- };
- struct SourceKeyHasher {
- size_t operator()(const SourceKey& value) const {
- return static_cast<size_t>(value.source_type) +
- static_cast<size_t>(value.source) * 11076425802534262905ULL;
- }
- };
- struct SourceEntry {
- // Timestamp indicating the most recent time a frame from an RTP packet,
- // originating from this source, was delivered to the RTCRtpReceiver's
- // MediaStreamTrack. Its reference clock is the outer class's |clock_|.
- int64_t timestamp_ms;
- // Audio level from an RFC 6464 or RFC 6465 header extension received with
- // the most recent packet used to assemble the frame associated with
- // |timestamp_ms|. May be absent. Only relevant for audio receivers. See the
- // specs for `RTCRtpContributingSource` for more info.
- absl::optional<uint8_t> audio_level;
- // Absolute capture time header extension received or interpolated from the
- // most recent packet used to assemble the frame. For more info see
- // https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
- absl::optional<AbsoluteCaptureTime> absolute_capture_time;
- // RTP timestamp of the most recent packet used to assemble the frame
- // associated with |timestamp_ms|.
- uint32_t rtp_timestamp;
- };
- using SourceList = std::list<std::pair<const SourceKey, SourceEntry>>;
- using SourceMap = std::unordered_map<SourceKey,
- SourceList::iterator,
- SourceKeyHasher,
- SourceKeyComparator>;
- // Updates an entry by creating it (if it didn't previously exist) and moving
- // it to the front of the list. Returns a reference to the entry.
- SourceEntry& UpdateEntry(const SourceKey& key)
- RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
- // Removes entries that have timed out. Marked as "const" so that we can do
- // pruning in getters.
- void PruneEntries(int64_t now_ms) const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
- Clock* const clock_;
- mutable Mutex lock_;
- // Entries are stored in reverse chronological order (i.e. with the most
- // recently updated entries appearing first). Mutability is needed for timeout
- // pruning in const functions.
- mutable SourceList list_ RTC_GUARDED_BY(lock_);
- mutable SourceMap map_ RTC_GUARDED_BY(lock_);
- };
- } // namespace webrtc
- #endif // MODULES_RTP_RTCP_SOURCE_SOURCE_TRACKER_H_
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